From 6a40dc5ab5036722d8102ba7190dbd9d72982637 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 13 Oct 2014 11:37:18 +0530 Subject: ALSA: au88x0: added reference of vortex_t added a pointer of the vortex in the following functions : vortex_alsafmt_aspfmt vortex_Vort3D_InitializeSource a3dsrc_ZeroStateA3D so that we can have a reference of the vortex in the function. this reference of the vortex will actually be used in a later patch to convert the pr_* macro to dev_*. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.h | 4 ++-- sound/pci/au88x0/au88x0_a3d.c | 6 +++--- sound/pci/au88x0/au88x0_core.c | 5 +++-- sound/pci/au88x0/au88x0_pcm.c | 2 +- 4 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index 466a5c8..3a8fefe 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -243,7 +243,7 @@ static int vortex_core_init(vortex_t * card); static int vortex_core_shutdown(vortex_t * card); static void vortex_enable_int(vortex_t * card); static irqreturn_t vortex_interrupt(int irq, void *dev_id); -static int vortex_alsafmt_aspfmt(int alsafmt); +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); @@ -278,7 +278,7 @@ static void vortex_mix_setvolumebyte(vortex_t * vortex, unsigned char mix, static void vortex_Vort3D_enable(vortex_t * v); static void vortex_Vort3D_disable(vortex_t * v); static void vortex_Vort3D_connect(vortex_t * vortex, int en); -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en); +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v); #endif /* Driver stuff. */ diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 30f760e..bc9cda3a 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -484,7 +484,7 @@ static void a3dsrc_ZeroState(a3dsrc_t * a) } /* Reset entire A3D engine */ -static void a3dsrc_ZeroStateA3D(a3dsrc_t * a) +static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) { int i, var, var2; @@ -601,7 +601,7 @@ static void vortex_Vort3D_enable(vortex_t *v) Vort3DRend_Initialize(v, XT_HEADPHONE); for (i = 0; i < NR_A3D; i++) { vortex_A3dSourceHw_Initialize(v, i % 4, i >> 2); - a3dsrc_ZeroStateA3D(&(v->a3d[0])); + a3dsrc_ZeroStateA3D(&v->a3d[0], v); } /* Register ALSA controls */ vortex_a3d_register_controls(v); @@ -676,7 +676,7 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) } /* Initialize one single A3D source. */ -static void vortex_Vort3D_InitializeSource(a3dsrc_t * a, int en) +static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { pr_warn diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 72e8128..00e2096 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2177,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, return -EBUSY; } /* (De)Initialize A3D hardware source. */ - vortex_Vort3D_InitializeSource(&(vortex->a3d[a3d]), en); + vortex_Vort3D_InitializeSource(&vortex->a3d[a3d], en, + vortex); } /* Make SPDIF out exclusive to "spdif" device when in use. */ if ((stream->type == VORTEX_PCM_SPDIF) && (en)) { @@ -2765,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) /* Alsa support. */ -static int vortex_alsafmt_aspfmt(int alsafmt) +static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) { int fmt; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5adc6b9..bdde182 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -332,7 +332,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) dir = 1; else dir = 0; - fmt = vortex_alsafmt_aspfmt(runtime->format); + fmt = vortex_alsafmt_aspfmt(runtime->format, chip); spin_lock_irq(&chip->lock); if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { vortex_adbdma_setmode(chip, dma, 1, dir, fmt, -- cgit v1.1 From 70c84418bf74f582e29906f1eeb19f2e9da53ddd Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 13 Oct 2014 11:37:19 +0530 Subject: ALSA: au88x0: pr_* replaced with dev_* pr_* macros replaced with dev_* as they are more preffered over pr_*. each file which had pr_* was reviewed manually and replaced with dev_*. here we have actually used the reference of the vortex which was added to some functions in the previous patch of this series. The prefix of the CARD_NAME and prefix of "vortex:" was also removed as the dev_* will print the device name. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.c | 33 +++++++------- sound/pci/au88x0/au88x0_a3d.c | 15 ++++--- sound/pci/au88x0/au88x0_core.c | 97 +++++++++++++++++++++------------------- sound/pci/au88x0/au88x0_eq.c | 3 +- sound/pci/au88x0/au88x0_game.c | 3 +- sound/pci/au88x0/au88x0_mpu401.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +-- sound/pci/au88x0/au88x0_synth.c | 17 ++++--- 8 files changed, 94 insertions(+), 82 deletions(-) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 21ce31f..e9c3833 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -48,11 +48,10 @@ static void vortex_fix_latency(struct pci_dev *vortex) { int rc; if (!(rc = pci_write_config_byte(vortex, 0x40, 0xff))) { - pr_info( CARD_NAME - ": vortex latency is 0xff\n"); + dev_info(&vortex->dev, "vortex latency is 0xff\n"); } else { - pr_warn( CARD_NAME - ": could not set vortex latency: pci error 0x%x\n", rc); + dev_warn(&vortex->dev, + "could not set vortex latency: pci error 0x%x\n", rc); } } @@ -70,11 +69,10 @@ static void vortex_fix_agp_bridge(struct pci_dev *via) if (!(rc = pci_read_config_byte(via, 0x42, &value)) && ((value & 0x10) || !(rc = pci_write_config_byte(via, 0x42, value | 0x10)))) { - pr_info( CARD_NAME - ": bridge config is 0x%x\n", value | 0x10); + dev_info(&via->dev, "bridge config is 0x%x\n", value | 0x10); } else { - pr_warn( CARD_NAME - ": could not set vortex latency: pci error 0x%x\n", rc); + dev_warn(&via->dev, + "could not set vortex latency: pci error 0x%x\n", rc); } } @@ -97,7 +95,8 @@ static void snd_vortex_workaround(struct pci_dev *vortex, int fix) PCI_DEVICE_ID_AMD_FE_GATE_7007, NULL); } if (via) { - pr_info( CARD_NAME ": Activating latency workaround...\n"); + dev_info(&vortex->dev, + "Activating latency workaround...\n"); vortex_fix_latency(vortex); vortex_fix_agp_bridge(via); } @@ -153,7 +152,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) return err; if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { - pr_err( "error to set DMA mask\n"); + dev_err(card->dev, "error to set DMA mask\n"); pci_disable_device(pci); return -ENXIO; } @@ -182,7 +181,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { - pr_err( "MMIO area remap failed.\n"); + dev_err(card->dev, "MMIO area remap failed.\n"); err = -ENOMEM; goto ioremap_out; } @@ -191,14 +190,14 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) * This must be done before we do request_irq otherwise we can get spurious * interrupts that we do not handle properly and make a mess of things */ if ((err = vortex_core_init(chip)) != 0) { - pr_err( "hw core init failed\n"); + dev_err(card->dev, "hw core init failed\n"); goto core_out; } if ((err = request_irq(pci->irq, vortex_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) != 0) { - pr_err( "cannot grab irq\n"); + dev_err(card->dev, "cannot grab irq\n"); goto irq_out; } chip->irq = pci->irq; @@ -342,11 +341,11 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) chip->rev = pci->revision; #ifdef CHIP_AU8830 if ((chip->rev) != 0xfe && (chip->rev) != 0xfa) { - pr_alert( - "vortex: The revision (%x) of your card has not been seen before.\n", + dev_alert(card->dev, + "The revision (%x) of your card has not been seen before.\n", chip->rev); - pr_alert( - "vortex: Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n"); + dev_alert(card->dev, + "Please email the results of 'lspci -vv' to openvortex-dev@nongnu.org.\n"); snd_card_free(card); err = -ENODEV; return err; diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index bc9cda3a..ab0f8731 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -489,7 +489,8 @@ static void a3dsrc_ZeroStateA3D(a3dsrc_t *a, vortex_t *v) int i, var, var2; if ((a->vortex) == NULL) { - pr_err( "vortex: ZeroStateA3D: ERROR: a->vortex is NULL\n"); + dev_err(v->card->dev, + "ZeroStateA3D: ERROR: a->vortex is NULL\n"); return; } @@ -628,15 +629,15 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) v->mixxtlk[0] = vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN); if (v->mixxtlk[0] < 0) { - pr_warn - ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n"); + dev_warn(v->card->dev, + "vortex_Vort3D: ERROR: not enough free mixer resources.\n"); return; } v->mixxtlk[1] = vortex_adb_checkinout(v, v->fixed_res, en, VORTEX_RESOURCE_MIXIN); if (v->mixxtlk[1] < 0) { - pr_warn - ("vortex: vortex_Vort3D: ERROR: not enough free mixer resources.\n"); + dev_warn(v->card->dev, + "vortex_Vort3D: ERROR: not enough free mixer resources.\n"); return; } #endif @@ -679,8 +680,8 @@ static void vortex_Vort3D_connect(vortex_t * v, int en) static void vortex_Vort3D_InitializeSource(a3dsrc_t *a, int en, vortex_t *v) { if (a->vortex == NULL) { - pr_warn - ("vortex: Vort3D_InitializeSource: A3D source not initialized\n"); + dev_warn(v->card->dev, + "Vort3D_InitializeSource: A3D source not initialized\n"); return; } if (en) { diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 00e2096..4667c32 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -285,8 +285,8 @@ vortex_mixer_addWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) temp = hwread(vortex->mmio, prev); //printk(KERN_INFO "vortex: mixAddWTD: while addr=%x, val=%x\n", prev, temp); if ((++lifeboat) > 0xf) { - pr_err( - "vortex_mixer_addWTD: lifeboat overflow\n"); + dev_err(vortex->card->dev, + "vortex_mixer_addWTD: lifeboat overflow\n"); return 0; } } @@ -303,7 +303,7 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) eax = hwread(vortex->mmio, VORTEX_MIXER_SR); if (((1 << ch) & eax) == 0) { - pr_err( "mix ALARM %x\n", eax); + dev_err(vortex->card->dev, "mix ALARM %x\n", eax); return 0; } ebp = VORTEX_MIXER_CHNBASE + (ch << 2); @@ -324,8 +324,8 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) //printk(KERN_INFO "vortex: mixdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); while ((edx & 0xf) != mix) { if ((esi) > 0xf) { - pr_err( - "vortex: mixdelWTD: error lifeboat overflow\n"); + dev_err(vortex->card->dev, + "mixdelWTD: error lifeboat overflow\n"); return 0; } esp14 = ebx; @@ -492,7 +492,7 @@ vortex_src_persist_convratio(vortex_t * vortex, unsigned char src, int ratio) hwwrite(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2), ratio); temp = hwread(vortex->mmio, VORTEX_SRC_CONVRATIO + (src << 2)); if ((++lifeboat) > 0x9) { - pr_err( "Vortex: Src cvr fail\n"); + dev_err(vortex->card->dev, "Src cvr fail\n"); break; } } @@ -684,8 +684,8 @@ vortex_src_addWTD(vortex_t * vortex, unsigned char src, unsigned char ch) temp = hwread(vortex->mmio, prev); //printk(KERN_INFO "vortex: srcAddWTD: while addr=%x, val=%x\n", prev, temp); if ((++lifeboat) > 0xf) { - pr_err( - "vortex_src_addWTD: lifeboat overflow\n"); + dev_err(vortex->card->dev, + "vortex_src_addWTD: lifeboat overflow\n"); return 0; } } @@ -703,7 +703,7 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) eax = hwread(vortex->mmio, VORTEX_SRCBLOCK_SR); if (((1 << ch) & eax) == 0) { - pr_err( "src alarm\n"); + dev_err(vortex->card->dev, "src alarm\n"); return 0; } ebp = VORTEX_SRC_CHNBASE + (ch << 2); @@ -724,8 +724,8 @@ vortex_src_delWTD(vortex_t * vortex, unsigned char src, unsigned char ch) //printk(KERN_INFO "vortex: srcdelWTD: 1 addr=%x, val=%x, src=%x\n", ebx, edx, src); while ((edx & 0xf) != src) { if ((esi) > 0xf) { - pr_warn - ("vortex: srcdelWTD: error, lifeboat overflow\n"); + dev_warn(vortex->card->dev, + "srcdelWTD: error, lifeboat overflow\n"); return 0; } esp14 = ebx; @@ -819,8 +819,8 @@ vortex_fifo_setadbctrl(vortex_t * vortex, int fifo, int stereo, int priority, do { temp = hwread(vortex->mmio, VORTEX_FIFO_ADBCTRL + (fifo << 2)); if (lifeboat++ > 0xbb8) { - pr_err( - "Vortex: vortex_fifo_setadbctrl fail\n"); + dev_err(vortex->card->dev, + "vortex_fifo_setadbctrl fail\n"); break; } } @@ -915,7 +915,8 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, do { temp = hwread(vortex->mmio, VORTEX_FIFO_WTCTRL + (fifo << 2)); if (lifeboat++ > 0xbb8) { - pr_err( "Vortex: vortex_fifo_setwtctrl fail\n"); + dev_err(vortex->card->dev, + "vortex_fifo_setwtctrl fail\n"); break; } } @@ -1042,7 +1043,7 @@ static void vortex_fifo_init(vortex_t * vortex) for (x = NR_ADB - 1; x >= 0; x--) { hwwrite(vortex->mmio, addr, (FIFO_U0 | FIFO_U1)); if (hwread(vortex->mmio, addr) != (FIFO_U0 | FIFO_U1)) - pr_err( "bad adb fifo reset!"); + dev_err(vortex->card->dev, "bad adb fifo reset!"); vortex_fifo_clearadbdata(vortex, x, FIFO_SIZE); addr -= 4; } @@ -1053,9 +1054,9 @@ static void vortex_fifo_init(vortex_t * vortex) for (x = NR_WT - 1; x >= 0; x--) { hwwrite(vortex->mmio, addr, FIFO_U0); if (hwread(vortex->mmio, addr) != FIFO_U0) - pr_err( - "bad wt fifo reset (0x%08x, 0x%08x)!\n", - addr, hwread(vortex->mmio, addr)); + dev_err(vortex->card->dev, + "bad wt fifo reset (0x%08x, 0x%08x)!\n", + addr, hwread(vortex->mmio, addr)); vortex_fifo_clearwtdata(vortex, x, FIFO_SIZE); addr -= 4; } @@ -1213,8 +1214,9 @@ static int vortex_adbdma_bufshift(vortex_t * vortex, int adbdma) if (dma->period_virt >= dma->nr_periods) dma->period_virt -= dma->nr_periods; if (delta != 1) - pr_info( "vortex: %d virt=%d, real=%d, delta=%d\n", - adbdma, dma->period_virt, dma->period_real, delta); + dev_info(vortex->card->dev, + "%d virt=%d, real=%d, delta=%d\n", + adbdma, dma->period_virt, dma->period_real, delta); return delta; } @@ -1482,8 +1484,8 @@ static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma) dma->period_real = page; if (delta != 1) - pr_warn( "vortex: wt virt = %d, delta = %d\n", - dma->period_virt, delta); + dev_warn(vortex->card->dev, "wt virt = %d, delta = %d\n", + dma->period_virt, delta); return delta; } @@ -1667,9 +1669,9 @@ vortex_adb_addroutes(vortex_t * vortex, unsigned char channel, hwread(vortex->mmio, VORTEX_ADB_RTBASE + (temp << 2)) & ADB_MASK; if ((lifeboat++) > ADB_MASK) { - pr_err( - "vortex_adb_addroutes: unending route! 0x%x\n", - *route); + dev_err(vortex->card->dev, + "vortex_adb_addroutes: unending route! 0x%x\n", + *route); return; } } @@ -1703,9 +1705,9 @@ vortex_adb_delroutes(vortex_t * vortex, unsigned char channel, hwread(vortex->mmio, VORTEX_ADB_RTBASE + (prev << 2)) & ADB_MASK; if (((lifeboat++) > ADB_MASK) || (temp == ADB_MASK)) { - pr_err( - "vortex_adb_delroutes: route not found! 0x%x\n", - route0); + dev_err(vortex->card->dev, + "vortex_adb_delroutes: route not found! 0x%x\n", + route0); return; } } @@ -2045,7 +2047,9 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) } } } - pr_err( "vortex: FATAL: ResManager: resource type %d exhausted.\n", restype); + dev_err(vortex->card->dev, + "FATAL: ResManager: resource type %d exhausted.\n", + restype); return -ENOMEM; } @@ -2173,7 +2177,8 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, memset(stream->resources, 0, sizeof(unsigned char) * VORTEX_RESOURCE_LAST); - pr_err( "vortex: out of A3D sources. Sorry\n"); + dev_err(vortex->card->dev, + "out of A3D sources. Sorry\n"); return -EBUSY; } /* (De)Initialize A3D hardware source. */ @@ -2422,7 +2427,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) hwread(vortex->mmio, VORTEX_IRQ_SOURCE); // Is at least one IRQ flag set? if (source == 0) { - pr_err( "vortex: missing irq source\n"); + dev_err(vortex->card->dev, "missing irq source\n"); return IRQ_NONE; } @@ -2430,19 +2435,19 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) // Attend every interrupt source. if (unlikely(source & IRQ_ERR_MASK)) { if (source & IRQ_FATAL) { - pr_err( "vortex: IRQ fatal error\n"); + dev_err(vortex->card->dev, "IRQ fatal error\n"); } if (source & IRQ_PARITY) { - pr_err( "vortex: IRQ parity error\n"); + dev_err(vortex->card->dev, "IRQ parity error\n"); } if (source & IRQ_REG) { - pr_err( "vortex: IRQ reg error\n"); + dev_err(vortex->card->dev, "IRQ reg error\n"); } if (source & IRQ_FIFO) { - pr_err( "vortex: IRQ fifo error\n"); + dev_err(vortex->card->dev, "IRQ fifo error\n"); } if (source & IRQ_DMA) { - pr_err( "vortex: IRQ dma error\n"); + dev_err(vortex->card->dev, "IRQ dma error\n"); } handled = 1; } @@ -2490,7 +2495,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) } if (!handled) { - pr_err( "vortex: unknown irq source %x\n", source); + dev_err(vortex->card->dev, "unknown irq source %x\n", source); } return IRQ_RETVAL(handled); } @@ -2547,7 +2552,7 @@ vortex_codec_write(struct snd_ac97 * codec, unsigned short addr, unsigned short while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) { udelay(100); if (lifeboat++ > POLL_COUNT) { - pr_err( "vortex: ac97 codec stuck busy\n"); + dev_err(card->card->dev, "ac97 codec stuck busy\n"); return; } } @@ -2573,7 +2578,7 @@ static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short while (!(hwread(card->mmio, VORTEX_CODEC_CTRL) & 0x100)) { udelay(100); if (lifeboat++ > POLL_COUNT) { - pr_err( "vortex: ac97 codec stuck busy\n"); + dev_err(card->card->dev, "ac97 codec stuck busy\n"); return 0xffff; } } @@ -2587,7 +2592,8 @@ static unsigned short vortex_codec_read(struct snd_ac97 * codec, unsigned short udelay(100); data = hwread(card->mmio, VORTEX_CODEC_IO); if (lifeboat++ > POLL_COUNT) { - pr_err( "vortex: ac97 address never arrived\n"); + dev_err(card->card->dev, + "ac97 address never arrived\n"); return 0xffff; } } while ((data & VORTEX_CODEC_ADDMASK) != @@ -2684,7 +2690,7 @@ static void vortex_spdif_init(vortex_t * vortex, int spdif_sr, int spdif_mode) static int vortex_core_init(vortex_t *vortex) { - pr_info( "Vortex: init.... "); + dev_info(vortex->card->dev, "init started\n"); /* Hardware Init. */ hwwrite(vortex->mmio, VORTEX_CTRL, 0xffffffff); msleep(5); @@ -2729,7 +2735,7 @@ static int vortex_core_init(vortex_t *vortex) //vortex_enable_timer_int(vortex); //vortex_disable_timer_int(vortex); - pr_info( "done.\n"); + dev_info(vortex->card->dev, "init.... done.\n"); spin_lock_init(&vortex->lock); return 0; @@ -2738,7 +2744,7 @@ static int vortex_core_init(vortex_t *vortex) static int vortex_core_shutdown(vortex_t * vortex) { - pr_info( "Vortex: shutdown..."); + dev_info(vortex->card->dev, "shutdown started\n"); #ifndef CHIP_AU8820 vortex_eq_free(vortex); vortex_Vort3D_disable(vortex); @@ -2760,7 +2766,7 @@ static int vortex_core_shutdown(vortex_t * vortex) msleep(5); hwwrite(vortex->mmio, VORTEX_IRQ_SOURCE, 0xffff); - pr_info( "done.\n"); + dev_info(vortex->card->dev, "shutdown.... done.\n"); return 0; } @@ -2794,7 +2800,8 @@ static int vortex_alsafmt_aspfmt(int alsafmt, vortex_t *v) break; default: fmt = 0x8; - pr_err( "vortex: format unsupported %d\n", alsafmt); + dev_err(v->card->dev, + "format unsupported %d\n", alsafmt); break; } return fmt; diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 9404ba7..9585c5c 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -845,7 +845,8 @@ snd_vortex_peaks_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u vortex_Eqlzr_GetAllPeaks(vortex, peaks, &count); if (count != 20) { - pr_err( "vortex: peak count error 20 != %d \n", count); + dev_err(vortex->card->dev, + "peak count error 20 != %d\n", count); return -1; } for (i = 0; i < 20; i++) diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c index 72daf6c..151815b 100644 --- a/sound/pci/au88x0/au88x0_game.c +++ b/sound/pci/au88x0/au88x0_game.c @@ -98,7 +98,8 @@ static int vortex_gameport_register(vortex_t *vortex) vortex->gameport = gp = gameport_allocate_port(); if (!gp) { - pr_err( "vortex: cannot allocate memory for gameport\n"); + dev_err(vortex->card->dev, + "cannot allocate memory for gameport\n"); return -ENOMEM; } diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 328c194..1025e55 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -73,7 +73,7 @@ static int snd_vortex_midi(vortex_t *vortex) /* Check if anything is OK. */ temp = hwread(vortex->mmio, VORTEX_MIDI_DATA); if (temp != MPU401_ACK /*0xfe */ ) { - pr_err( "midi port doesn't acknowledge!\n"); + dev_err(vortex->card->dev, "midi port doesn't acknowledge!\n"); return -ENODEV; } /* Enable MPU401 interrupts. */ diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index bdde182..a6d6d8d 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -227,7 +227,7 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) { - pr_err( "Vortex: pcm page alloc failed!\n"); + dev_err(chip->card->dev, "Vortex: pcm page alloc failed!\n"); return err; } /* @@ -371,7 +371,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } #ifndef CHIP_AU8810 else { - pr_info( "vortex: wt start %d\n", dma); + dev_info(chip->card->dev, "wt start %d\n", dma); vortex_wtdma_startfifo(chip, dma); } #endif @@ -384,7 +384,7 @@ static int snd_vortex_pcm_trigger(struct snd_pcm_substream *substream, int cmd) vortex_adbdma_stopfifo(chip, dma); #ifndef CHIP_AU8810 else { - pr_info( "vortex: wt stop %d\n", dma); + dev_info(chip->card->dev, "wt stop %d\n", dma); vortex_wtdma_stopfifo(chip, dma); } #endif diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index f094bac..78e12f4 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -90,7 +90,7 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch) hwwrite(vortex->mmio, WT_PARM(wt, 2), 0); temp = hwread(vortex->mmio, WT_PARM(wt, 3)); - pr_debug( "vortex: WT PARM3: %x\n", temp); + dev_dbg(vortex->card->dev, "WT PARM3: %x\n", temp); //hwwrite(vortex->mmio, WT_PARM(wt, 3), temp); hwwrite(vortex->mmio, WT_DELAY(wt, 0), 0); @@ -98,7 +98,8 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch) hwwrite(vortex->mmio, WT_DELAY(wt, 2), 0); hwwrite(vortex->mmio, WT_DELAY(wt, 3), 0); - pr_debug( "vortex: WT GMODE: %x\n", hwread(vortex->mmio, WT_GMODE(wt))); + dev_dbg(vortex->card->dev, "WT GMODE: %x\n", + hwread(vortex->mmio, WT_GMODE(wt))); hwwrite(vortex->mmio, WT_PARM(wt, 2), 0xffffffff); hwwrite(vortex->mmio, WT_PARM(wt, 3), 0xcff1c810); @@ -106,7 +107,8 @@ static int vortex_wt_allocroute(vortex_t * vortex, int wt, int nr_ch) voice->parm0 = voice->parm1 = 0xcfb23e2f; hwwrite(vortex->mmio, WT_PARM(wt, 0), voice->parm0); hwwrite(vortex->mmio, WT_PARM(wt, 1), voice->parm1); - pr_debug( "vortex: WT GMODE 2 : %x\n", hwread(vortex->mmio, WT_GMODE(wt))); + dev_dbg(vortex->card->dev, "WT GMODE 2 : %x\n", + hwread(vortex->mmio, WT_GMODE(wt))); return 0; } @@ -196,14 +198,15 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, if ((reg == 5) || ((reg >= 7) && (reg <= 10)) || (reg == 0xc)) { if (wt >= (NR_WT / NR_WT_PB)) { - pr_warn - ("vortex: WT SetReg: bank out of range. reg=0x%x, wt=%d\n", - reg, wt); + dev_warn(vortex->card->dev, + "WT SetReg: bank out of range. reg=0x%x, wt=%d\n", + reg, wt); return 0; } } else { if (wt >= NR_WT) { - pr_err( "vortex: WT SetReg: voice out of range\n"); + dev_err(vortex->card->dev, + "WT SetReg: voice out of range\n"); return 0; } } -- cgit v1.1 From 54841a06c54eb55918948c12ab9b5f02cacb6ab3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Oct 2014 14:00:16 +0200 Subject: ALSA: seq: Use atomic ops for autoload refcount ... just to robustify for races. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 91a786a..775ea93 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -127,15 +127,15 @@ static void snd_seq_device_info(struct snd_info_entry *entry, #ifdef CONFIG_MODULES /* avoid auto-loading during module_init() */ -static int snd_seq_in_init; +static atomic_t snd_seq_in_init = ATOMIC_INIT(0); void snd_seq_autoload_lock(void) { - snd_seq_in_init++; + atomic_inc(&snd_seq_in_init); } void snd_seq_autoload_unlock(void) { - snd_seq_in_init--; + atomic_dec(&snd_seq_in_init); } #endif @@ -147,7 +147,7 @@ void snd_seq_device_load_drivers(void) /* Calling request_module during module_init() * may cause blocking. */ - if (snd_seq_in_init) + if (atomic_read(&snd_seq_in_init)) return; mutex_lock(&ops_mutex); -- cgit v1.1 From 68ab61084de3220e2fb0a698c890ba91decddc85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Oct 2014 14:06:25 +0200 Subject: ALSA: seq: bind seq driver automatically Currently the sequencer module binding is performed independently from the card module itself. The reason behind it is to keep the sequencer stuff optional and allow the system running without it (e.g. for using PCM or rawmidi only). This works in most cases, but a remaining problem is that the binding isn't done automatically when a new driver module is probed. Typically this becomes visible when a hotplug driver like usb audio is used. This patch tries to address this and other potential issues. First, the seq-binder (seq_device.c) tries to load a missing driver module at creating a new device object. This is done asynchronously in a workq for avoiding the deadlock (modprobe call in module init path). This action, however, should be enabled only when the sequencer stuff was already initialized, i.e. snd-seq module was already loaded. For that, a new function, snd_seq_autoload_init() is introduced here; this clears the blocking of autoloading, and also tries to load all pending driver modules. Reported-by: Adam Goode Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 4 ++ sound/core/seq/seq.c | 3 ++ sound/core/seq/seq_device.c | 92 +++++++++++++++++++++++++++++++++------------ 3 files changed, 76 insertions(+), 23 deletions(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 2398521..eea5400 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -108,9 +108,13 @@ int snd_seq_event_port_detach(int client, int port); #ifdef CONFIG_MODULES void snd_seq_autoload_lock(void); void snd_seq_autoload_unlock(void); +void snd_seq_autoload_init(void); +#define snd_seq_autoload_exit() snd_seq_autoload_lock() #else #define snd_seq_autoload_lock() #define snd_seq_autoload_unlock() +#define snd_seq_autoload_init() +#define snd_seq_autoload_exit() #endif #endif /* __SOUND_SEQ_KERNEL_H */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7121105..bebdd2e 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -110,6 +110,7 @@ static int __init alsa_seq_init(void) if ((err = snd_seq_system_client_init()) < 0) goto error; + snd_seq_autoload_init(); error: snd_seq_autoload_unlock(); return err; @@ -131,6 +132,8 @@ static void __exit alsa_seq_exit(void) /* release event memory */ snd_sequencer_memory_done(); + + snd_seq_autoload_exit(); } module_init(alsa_seq_init) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 775ea93..a8e2c60 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -56,6 +56,7 @@ MODULE_LICENSE("GPL"); #define DRIVER_LOADED (1<<0) #define DRIVER_REQUESTED (1<<1) #define DRIVER_LOCKED (1<<2) +#define DRIVER_REQUESTING (1<<3) struct ops_list { char id[ID_LEN]; /* driver id */ @@ -127,7 +128,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry, #ifdef CONFIG_MODULES /* avoid auto-loading during module_init() */ -static atomic_t snd_seq_in_init = ATOMIC_INIT(0); +static atomic_t snd_seq_in_init = ATOMIC_INIT(1); /* blocked as default */ void snd_seq_autoload_lock(void) { atomic_inc(&snd_seq_in_init); @@ -137,32 +138,72 @@ void snd_seq_autoload_unlock(void) { atomic_dec(&snd_seq_in_init); } -#endif -void snd_seq_device_load_drivers(void) +static void autoload_drivers(void) { -#ifdef CONFIG_MODULES - struct ops_list *ops; + /* avoid reentrance */ + if (atomic_inc_return(&snd_seq_in_init) == 1) { + struct ops_list *ops; + + mutex_lock(&ops_mutex); + list_for_each_entry(ops, &opslist, list) { + if ((ops->driver & DRIVER_REQUESTING) && + !(ops->driver & DRIVER_REQUESTED)) { + ops->used++; + mutex_unlock(&ops_mutex); + ops->driver |= DRIVER_REQUESTED; + request_module("snd-%s", ops->id); + mutex_lock(&ops_mutex); + ops->used--; + } + } + mutex_unlock(&ops_mutex); + } + atomic_dec(&snd_seq_in_init); +} - /* Calling request_module during module_init() - * may cause blocking. - */ - if (atomic_read(&snd_seq_in_init)) - return; +static void call_autoload(struct work_struct *work) +{ + autoload_drivers(); +} - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if (! (ops->driver & DRIVER_LOADED) && - ! (ops->driver & DRIVER_REQUESTED)) { - ops->used++; - mutex_unlock(&ops_mutex); - ops->driver |= DRIVER_REQUESTED; - request_module("snd-%s", ops->id); - mutex_lock(&ops_mutex); - ops->used--; - } +static DECLARE_WORK(autoload_work, call_autoload); + +static void try_autoload(struct ops_list *ops) +{ + if (!ops->driver) { + ops->driver |= DRIVER_REQUESTING; + schedule_work(&autoload_work); } +} + +static void queue_autoload_drivers(void) +{ + struct ops_list *ops; + + mutex_lock(&ops_mutex); + list_for_each_entry(ops, &opslist, list) + try_autoload(ops); mutex_unlock(&ops_mutex); +} + +void snd_seq_autoload_init(void) +{ + atomic_dec(&snd_seq_in_init); +#ifdef CONFIG_SND_SEQUENCER_MODULE + /* initial autoload only when snd-seq is a module */ + queue_autoload_drivers(); +#endif +} +#else +#define try_autoload(ops) /* NOP */ +#endif + +void snd_seq_device_load_drivers(void) +{ +#ifdef CONFIG_MODULES + queue_autoload_drivers(); + flush_work(&autoload_work); #endif } @@ -214,13 +255,14 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, ops->num_devices++; mutex_unlock(&ops->reg_mutex); - unlock_driver(ops); - if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) { snd_seq_device_free(dev); return err; } + try_autoload(ops); + unlock_driver(ops); + if (result) *result = dev; @@ -554,6 +596,9 @@ static int __init alsa_seq_device_init(void) static void __exit alsa_seq_device_exit(void) { +#ifdef CONFIG_MODULES + cancel_work_sync(&autoload_work); +#endif remove_drivers(); #ifdef CONFIG_PROC_FS snd_info_free_entry(info_entry); @@ -570,6 +615,7 @@ EXPORT_SYMBOL(snd_seq_device_new); EXPORT_SYMBOL(snd_seq_device_register_driver); EXPORT_SYMBOL(snd_seq_device_unregister_driver); #ifdef CONFIG_MODULES +EXPORT_SYMBOL(snd_seq_autoload_init); EXPORT_SYMBOL(snd_seq_autoload_lock); EXPORT_SYMBOL(snd_seq_autoload_unlock); #endif -- cgit v1.1 From d5129f33a0d155d69cb0652cfc87bbc4d132ca17 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Oct 2014 14:09:42 +0200 Subject: Subject: ALSA: seq: Remove autoload locks in driver registration Since we're calling request_module() asynchronously now, we can get rid of the autoload lock in snd_seq_device_register_driver(), as well as in the snd-seq driver registration itself. This enables the automatic loading of dependent sequencer modules, such as snd-seq-virmidi from snd-emu10k1-synth. Signed-off-by: Takashi Iwai --- sound/core/seq/seq.c | 2 -- sound/core/seq/seq_device.c | 7 +------ 2 files changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index bebdd2e..7e0aabb 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -86,7 +86,6 @@ static int __init alsa_seq_init(void) { int err; - snd_seq_autoload_lock(); if ((err = client_init_data()) < 0) goto error; @@ -112,7 +111,6 @@ static int __init alsa_seq_init(void) snd_seq_autoload_init(); error: - snd_seq_autoload_unlock(); return err; } diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index a8e2c60..0631bda 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -360,16 +360,12 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, entry->init_device == NULL || entry->free_device == NULL) return -EINVAL; - snd_seq_autoload_lock(); ops = find_driver(id, 1); - if (ops == NULL) { - snd_seq_autoload_unlock(); + if (ops == NULL) return -ENOMEM; - } if (ops->driver & DRIVER_LOADED) { pr_warn("ALSA: seq: driver_register: driver '%s' already exists\n", id); unlock_driver(ops); - snd_seq_autoload_unlock(); return -EBUSY; } @@ -386,7 +382,6 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, mutex_unlock(&ops->reg_mutex); unlock_driver(ops); - snd_seq_autoload_unlock(); return 0; } -- cgit v1.1 From 49fd46d2ffc92de3f4bd2e337f60b8b6052dc8b6 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 19 Oct 2014 09:11:25 +0200 Subject: ALSA: snd-usb: drop unused varible assigments Don't assign 'len' in cases where we don't make use of the returned value. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2e4a9db..63a8adb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1290,9 +1290,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kctl->id.name, sizeof(kctl->id.name), 1); if (!len) - len = snprintf(kctl->id.name, - sizeof(kctl->id.name), - "Feature %d", unitid); + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Feature %d", unitid); } if (!mapped_name) @@ -1305,9 +1304,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, */ if (!mapped_name && !(state->oterm.type >> 16)) { if ((state->oterm.type & 0xff00) == 0x0100) - len = append_ctl_name(kctl, " Capture"); + append_ctl_name(kctl, " Capture"); else - len = append_ctl_name(kctl, " Playback"); + append_ctl_name(kctl, " Playback"); } append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); -- cgit v1.1 From ae11601b80b984859deda1f2d430a23bfabd3bea Mon Sep 17 00:00:00 2001 From: Ramesh Babu Date: Wed, 15 Oct 2014 12:34:59 +0530 Subject: ASoC: core: Call mute for cpu dais as well We call mute for codec dai only, we should call this for cpu dai as well to allow cpu dais (FEs) in DSPs to be muted/unmuted on shutdown/startup Signed-off-by: Ramesh Babu Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..1cd027a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -654,6 +654,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->rate = 0; } + snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream); + if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); @@ -772,6 +774,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) for (i = 0; i < rtd->num_codecs; i++) snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, substream->stream); + snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); out: mutex_unlock(&rtd->pcm_mutex); -- cgit v1.1 From fd587e320041d42eb21d12bb62da9e8ac08fd6c2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 6 Oct 2014 23:14:23 +0800 Subject: ASoC: cs4265: Remove unused *dev field from struct cs4265_private Signed-off-by: Axel Lin Acked-by: Paul Handrigan Signed-off-by: Mark Brown --- sound/soc/codecs/cs4265.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 4fdd47d..ce60868 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -32,7 +32,6 @@ #include "cs4265.h" struct cs4265_private { - struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpio; u8 format; @@ -598,7 +597,6 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, GFP_KERNEL); if (cs4265 == NULL) return -ENOMEM; - cs4265->dev = &i2c_client->dev; cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap); if (IS_ERR(cs4265->regmap)) { -- cgit v1.1 From c973b8a7dc50ace86393f209b19aa7fd0bfaf66b Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 6 Oct 2014 23:09:47 +0800 Subject: ASoC: cs4271: Split SPI and I2C code into different modules Currently the cs4271 driver depends on SND_SOC_I2C_AND_SPI. So the driver cannot be built as built-in if CONFIG_I2C=m. Split SPI and I2C code into different modules to avoid this issue. Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/cirrus/Kconfig | 3 +- sound/soc/codecs/Kconfig | 18 ++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/cs4271-i2c.c | 62 +++++++++++++++++ sound/soc/codecs/cs4271-spi.c | 55 +++++++++++++++ sound/soc/codecs/cs4271.c | 155 +++++------------------------------------- sound/soc/codecs/cs4271.h | 11 +++ 7 files changed, 167 insertions(+), 141 deletions(-) create mode 100644 sound/soc/codecs/cs4271-i2c.c create mode 100644 sound/soc/codecs/cs4271-spi.c create mode 100644 sound/soc/codecs/cs4271.h diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 5477c54..7b7fbcd 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -36,7 +36,8 @@ config SND_EP93XX_SOC_EDB93XX tristate "SoC Audio support for Cirrus Logic EDB93xx boards" depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) select SND_EP93XX_SOC_I2S - select SND_SOC_CS4271 + select SND_SOC_CS4271_I2C if I2C + select SND_SOC_CS4271_SPI if SPI_MASTER help Say Y or M here if you want to add support for I2S audio on the Cirrus Logic EDB93xx boards. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..7650625 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -50,7 +50,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4265 if I2C select SND_SOC_CS4270 if I2C - select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS4271_I2C if I2C + select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C @@ -370,8 +371,19 @@ config SND_SOC_CS4270_VD33_ERRATA depends on SND_SOC_CS4270 config SND_SOC_CS4271 - tristate "Cirrus Logic CS4271 CODEC" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_CS4271_I2C + tristate "Cirrus Logic CS4271 CODEC (I2C)" + depends on I2C + select SND_SOC_CS4271 + select REGMAP_I2C + +config SND_SOC_CS4271_SPI + tristate "Cirrus Logic CS4271 CODEC (SPI)" + depends on SPI_MASTER + select SND_SOC_CS4271 + select REGMAP_SPI config SND_SOC_CS42XX8 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..ac7ec31 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,6 +41,8 @@ snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4265-objs := cs4265.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs4271-i2c-objs := cs4271-i2c.o +snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o @@ -217,6 +219,8 @@ obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS4271_I2C) += snd-soc-cs4271-i2c.o +obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c new file mode 100644 index 0000000..b264da0 --- /dev/null +++ b/sound/soc/codecs/cs4271-i2c.c @@ -0,0 +1,62 @@ +/* + * CS4271 I2C audio driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include "cs4271.h" + +static int cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = cs4271_regmap_config; + config.reg_bits = 8; + config.val_bits = 8; + + return cs4271_probe(&client->dev, + devm_regmap_init_i2c(client, &config)); +} + +static int cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id cs4271_i2c_id[] = { + { "cs4271", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), + }, + .probe = cs4271_i2c_probe, + .remove = cs4271_i2c_remove, + .id_table = cs4271_i2c_id, +}; +module_i2c_driver(cs4271_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS4271 I2C Driver"); +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4271-spi.c b/sound/soc/codecs/cs4271-spi.c new file mode 100644 index 0000000..acd49d8 --- /dev/null +++ b/sound/soc/codecs/cs4271-spi.c @@ -0,0 +1,55 @@ +/* + * CS4271 SPI audio driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include "cs4271.h" + +static int cs4271_spi_probe(struct spi_device *spi) +{ + struct regmap_config config; + + config = cs4271_regmap_config; + config.reg_bits = 16; + config.val_bits = 8; + config.read_flag_mask = 0x21; + config.write_flag_mask = 0x20; + + return cs4271_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); +} + +static int cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), + }, + .probe = cs4271_spi_probe, + .remove = cs4271_spi_remove, +}; +module_spi_driver(cs4271_spi_driver); + +MODULE_DESCRIPTION("ASoC CS4271 SPI Driver"); +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 93cec52..79a4efc 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -23,8 +23,6 @@ #include #include #include -#include -#include #include #include #include @@ -32,6 +30,7 @@ #include #include #include +#include "cs4271.h" #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ @@ -527,14 +526,15 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) #endif /* CONFIG_PM */ #ifdef CONFIG_OF -static const struct of_device_id cs4271_dt_ids[] = { +const struct of_device_id cs4271_dt_ids[] = { { .compatible = "cirrus,cs4271", }, { } }; MODULE_DEVICE_TABLE(of, cs4271_dt_ids); +EXPORT_SYMBOL_GPL(cs4271_dt_ids); #endif -static int cs4271_probe(struct snd_soc_codec *codec) +static int cs4271_codec_probe(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; @@ -587,7 +587,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) return 0; } -static int cs4271_remove(struct snd_soc_codec *codec) +static int cs4271_codec_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -599,8 +599,8 @@ static int cs4271_remove(struct snd_soc_codec *codec) }; static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { - .probe = cs4271_probe, - .remove = cs4271_remove, + .probe = cs4271_codec_probe, + .remove = cs4271_codec_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, @@ -642,14 +642,8 @@ static int cs4271_common_probe(struct device *dev, return 0; } -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, +const struct regmap_config cs4271_regmap_config = { .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, .reg_defaults = cs4271_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), @@ -657,140 +651,27 @@ static const struct regmap_config cs4271_spi_regmap = { .volatile_reg = cs4271_volatile_reg, }; +EXPORT_SYMBOL_GPL(cs4271_regmap_config); -static int cs4271_spi_probe(struct spi_device *spi) +int cs4271_probe(struct device *dev, struct regmap *regmap) { struct cs4271_private *cs4271; int ret; - ret = cs4271_common_probe(&spi->dev, &cs4271); - if (ret < 0) - return ret; - - spi_set_drvdata(spi, cs4271); - cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); - if (IS_ERR(cs4271->regmap)) - return PTR_ERR(cs4271->regmap); - - return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, - &cs4271_dai, 1); -} - -static int cs4271_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver cs4271_spi_driver = { - .driver = { - .name = "cs4271", - .owner = THIS_MODULE, - .of_match_table = of_match_ptr(cs4271_dt_ids), - }, - .probe = cs4271_spi_probe, - .remove = cs4271_spi_remove, -}; -#endif /* defined(CONFIG_SPI_MASTER) */ - -#if IS_ENABLED(CONFIG_I2C) -static const struct i2c_device_id cs4271_i2c_id[] = { - {"cs4271", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); -static const struct regmap_config cs4271_i2c_regmap = { - .reg_bits = 8, - .val_bits = 8, - .max_register = CS4271_LASTREG, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - -static int cs4271_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - struct cs4271_private *cs4271; - int ret; - - ret = cs4271_common_probe(&client->dev, &cs4271); + ret = cs4271_common_probe(dev, &cs4271); if (ret < 0) return ret; - i2c_set_clientdata(client, cs4271); - cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); - if (IS_ERR(cs4271->regmap)) - return PTR_ERR(cs4271->regmap); + dev_set_drvdata(dev, cs4271); + cs4271->regmap = regmap; - return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, - &cs4271_dai, 1); -} - -static int cs4271_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static struct i2c_driver cs4271_i2c_driver = { - .driver = { - .name = "cs4271", - .owner = THIS_MODULE, - .of_match_table = of_match_ptr(cs4271_dt_ids), - }, - .id_table = cs4271_i2c_id, - .probe = cs4271_i2c_probe, - .remove = cs4271_i2c_remove, -}; -#endif /* IS_ENABLED(CONFIG_I2C) */ - -/* - * We only register our serial bus driver here without - * assignment to particular chip. So if any of the below - * fails, there is some problem with I2C or SPI subsystem. - * In most cases this module will be compiled with support - * of only one serial bus. - */ -static int __init cs4271_modinit(void) -{ - int ret; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&cs4271_i2c_driver); - if (ret) { - pr_err("Failed to register CS4271 I2C driver: %d\n", ret); - return ret; - } -#endif - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&cs4271_spi_driver); - if (ret) { - pr_err("Failed to register CS4271 SPI driver: %d\n", ret); - return ret; - } -#endif - - return 0; -} -module_init(cs4271_modinit); - -static void __exit cs4271_modexit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&cs4271_spi_driver); -#endif - -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&cs4271_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_cs4271, &cs4271_dai, + 1); } -module_exit(cs4271_modexit); +EXPORT_SYMBOL_GPL(cs4271_probe); MODULE_AUTHOR("Alexander Sverdlin "); MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); diff --git a/sound/soc/codecs/cs4271.h b/sound/soc/codecs/cs4271.h new file mode 100644 index 0000000..9adad8e --- /dev/null +++ b/sound/soc/codecs/cs4271.h @@ -0,0 +1,11 @@ +#ifndef _CS4271_PRIV_H +#define _CS4271_PRIV_H + +#include + +extern const struct of_device_id cs4271_dt_ids[]; +extern const struct regmap_config cs4271_regmap_config; + +int cs4271_probe(struct device *dev, struct regmap *regmap); + +#endif -- cgit v1.1 From 051c77e552bf1c80adf3fb2fa5e145ca7c9b0f08 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 15:13:24 -0300 Subject: ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-wm8962.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..48179ff 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) clk_fail: clk_disable_unprepare(data->codec_clk); fail: - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } -- cgit v1.1 From 7e3bc3c1c9017867b88593ecb4c0c1ae9da16fb2 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 15:13:25 -0300 Subject: ASoC: fsl: imx-sgtl5000: Delete unneeded test before of_node_put of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-sgtl5000.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 1cb22dd..1dab963 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -175,10 +175,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } -- cgit v1.1 From aa4ac9201676ddd35aa56ae74bdf8e07454f04fc Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 15:29:56 -0300 Subject: ASoC: fsl: imx-spdif: Delete unneeded test before of_node_put of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-spdif.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e1dc401..0c9068e 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -74,8 +74,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) platform_set_drvdata(pdev, data); end: - if (spdif_np) - of_node_put(spdif_np); + of_node_put(spdif_np); return ret; } -- cgit v1.1 From 89dea487646831d47b0c56723a7c43c62b094c48 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 15:29:57 -0300 Subject: ASoC: fsl: eukrea-tlv320: Delete unneeded test before of_node_put of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index eb093d5..dd931fa 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -217,8 +217,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - if (np) - of_node_put(ssi_np); + of_node_put(ssi_np); return ret; } -- cgit v1.1 From 077661b6ed24e530dabc9db3ab3ae48fbaf19679 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Oct 2014 20:56:29 +0200 Subject: ASoC: eukrea-tlv320: Fix of_node_put() call with uninitialized object The of_node_put() call in eukrea_tlv320_probe() may take an uninitialized pointer, as compiler spotted out: sound/soc/fsl/eukrea-tlv320.c:221:14: warning: 'ssi_np' may be used uninitialized in this function [-Wuninitialized] This patch adds the proper NULL initializations as a fix. (codec_np is also NULL initialized just for consistency.) Fixes: 66f232908de2 ('ASoC: eukrea-tlv320: Add DT support') Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index dd931fa..b175b01 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -105,7 +105,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) int ret; int int_port = 0, ext_port; struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; + struct device_node *ssi_np = NULL, *codec_np = NULL; eukrea_tlv320.dev = &pdev->dev; if (np) { -- cgit v1.1 From 73a2cd9193ae50d4cc7c447f8929e13010b589be Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:04 -0700 Subject: ASoC: fsl_esai: Add indentation for binding doc to increase readability This patch simply adds indentations for DT binding doc to increase readability without changing any contents. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,esai.txt | 44 +++++++++++----------- 1 file changed, 23 insertions(+), 21 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt index 52f5b6b..d3b6b5f 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt @@ -7,37 +7,39 @@ other DSPs. It has up to six transmitters and four receivers. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-esai" or - "fsl,vf610-esai" + - compatible : Compatible list, must contain "fsl,imx35-esai" or + "fsl,vf610-esai" - - reg : Offset and length of the register set for the device. + - reg : Offset and length of the register set for the device. - - interrupts : Contains the spdif interrupt. + - interrupts : Contains the spdif interrupt. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. - - dma-names : Two dmas have to be defined, "tx" and "rx". + - dma-names : Two dmas have to be defined, "tx" and "rx". - - clocks: Contains an entry for each entry in clock-names. + - clocks : Contains an entry for each entry in clock-names. - - clock-names : Includes the following entries: - "core" The core clock used to access registers - "extal" The esai baud clock for esai controller used to derive - HCK, SCK and FS. - "fsys" The system clock derived from ahb clock used to derive - HCK, SCK and FS. + - clock-names : Includes the following entries: + "core" The core clock used to access registers + "extal" The esai baud clock for esai controller used to + derive HCK, SCK and FS. + "fsys" The system clock derived from ahb clock used to + derive HCK, SCK and FS. - - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. - This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM]. + - fsl,fifo-depth : The number of elements in the transmit and receive + FIFOs. This number is the maximum allowed value for + TFCR[TFWM] or RFCR[RFWM]. - fsl,esai-synchronous: This is a boolean property. If present, indicating - that ESAI would work in the synchronous mode, which means all the settings - for Receiving would be duplicated from Transmition related registers. + that ESAI would work in the synchronous mode, which + means all the settings for Receiving would be + duplicated from Transmition related registers. - - big-endian : If this property is absent, the native endian mode will - be in use as default, or the big endian mode will be in use for all the - device registers. + - big-endian : If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. Example: -- cgit v1.1 From 9c4c1045343836b848c3dd546277c955d145d20a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:05 -0700 Subject: ASoC: fsl_spdif: Add indentation for binding doc to increase readability This patch simply adds indentations for DT binding doc to increase readability without changing any contents. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,spdif.txt | 37 +++++++++++----------- 1 file changed, 18 insertions(+), 19 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 3e9e82c8..b5ee32e 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,32 +6,31 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, must contain "fsl,imx35-spdif". - - reg : Offset and length of the register set for the device. + - reg : Offset and length of the register set for the device. - - interrupts : Contains the spdif interrupt. + - interrupts : Contains the spdif interrupt. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. - - dma-names : Two dmas have to be defined, "tx" and "rx". + - dma-names : Two dmas have to be defined, "tx" and "rx". - - clocks : Contains an entry for each entry in clock-names. + - clocks : Contains an entry for each entry in clock-names. - - clock-names : Includes the following entries: - "core" The core clock of spdif controller - "rxtx<0-7>" Clock source list for tx and rx clock. - This clock list should be identical to - the source list connecting to the spdif - clock mux in "SPDIF Transceiver Clock - Diagram" of SoC reference manual. It - can also be referred to TxClk_Source - bit of register SPDIF_STC. + - clock-names : Includes the following entries: + "core" The core clock of spdif controller. + "rxtx<0-7>" Clock source list for tx and rx clock. + This clock list should be identical to the source + list connecting to the spdif clock mux in "SPDIF + Transceiver Clock Diagram" of SoC reference manual. + It can also be referred to TxClk_Source bit of + register SPDIF_STC. - - big-endian : If this property is absent, the native endian mode will - be in use as default, or the big endian mode will be in use for all the - device registers. + - big-endian : If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. Example: -- cgit v1.1 From 0b9938b264d1a76458cf06aab6de7b9cec68efca Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:06 -0700 Subject: ASoC: fsl_sai: Add indentation for binding doc to increase readability This patch refines the DT binding doc for more readability by adding extra blank lines and indentations. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl-sai.txt | 66 ++++++++++++++-------- 1 file changed, 41 insertions(+), 25 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 4956b14..044e5d7 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -5,32 +5,48 @@ which provides a synchronous audio interface that supports fullduplex serial interfaces with frame synchronization such as I2S, AC97, TDM, and codec/DSP interfaces. - Required properties: -- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". -- reg: Offset and length of the register set for the device. -- clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" - "mclk3" for bit clock and frame clock providing. -- dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. -- dma-names : Two dmas have to be defined, "tx" and "rx". -- pinctrl-names: Must contain a "default" entry. -- pinctrl-NNN: One property must exist for each entry in pinctrl-names. - See ../pinctrl/pinctrl-bindings.txt for details of the property values. -- big-endian: Boolean property, required if all the FTM_PWM registers - are big-endian rather than little-endian. -- lsb-first: Configures whether the LSB or the MSB is transmitted first for - the fifo data. If this property is absent, the MSB is transmitted first as - default, or the LSB is transmitted first. -- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx with Rx) which means - both the transimitter and receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. -- fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which means both transimitter - and receiver will send and receive data by following their own bit clocks - and frame sync clocks separately. + + - compatible : Compatible list, contains "fsl,vf610-sai" or + "fsl,imx6sx-sai". + + - reg : Offset and length of the register set for the device. + + - clocks : Must contain an entry for each entry in clock-names. + + - clock-names : Must include the "bus" for register access and + "mclk1", "mclk2", "mclk3" for bit clock and frame + clock providing. + - dmas : Generic dma devicetree binding as described in + Documentation/devicetree/bindings/dma/dma.txt. + + - dma-names : Two dmas have to be defined, "tx" and "rx". + + - pinctrl-names : Must contain a "default" entry. + + - pinctrl-NNN : One property must exist for each entry in + pinctrl-names. See ../pinctrl/pinctrl-bindings.txt + for details of the property values. + + - big-endian : Boolean property, required if all the FTM_PWM + registers are big-endian rather than little-endian. + + - lsb-first : Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx + with Rx) which means both the transimitter and the + receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. + + - fsl,sai-asynchronous: This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which + means both transimitter and receiver will send and + receive data by following their own bit clocks and + frame sync clocks separately. Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the -- cgit v1.1 From d29ae41edde680c00f9f74d448b7d59c91ca0474 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:07 -0700 Subject: ASoC: eukrea-tlv320: Add indentation for binding doc to increase readability This patch refines the DT binding doc for more readability by adding extra blank lines and indentations. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/eukrea-tlv320.txt | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt index 0d7985c..6dfa88c 100644 --- a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -1,11 +1,16 @@ Audio complex for Eukrea boards with tlv320aic23 codec. Required properties: -- compatible : "eukrea,asoc-tlv320" -- eukrea,model : The user-visible name of this sound complex. -- ssi-controller : The phandle of the SSI controller. -- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). -- fsl,mux-ext-port : The external port of the i.MX audio muxer. + + - compatible : "eukrea,asoc-tlv320" + + - eukrea,model : The user-visible name of this sound complex. + + - ssi-controller : The phandle of the SSI controller. + + - fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + + - fsl,mux-ext-port : The external port of the i.MX audio muxer. Note: The AUDMUX port numbering should start at 1, which is consistent with hardware manual. -- cgit v1.1 From 5463c709ddeeacc73ae6844fd8aebc0a217b98d4 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:08 -0700 Subject: ASoC: imx-audmux: Add indentation for binding doc to increase readability This patch refines the DT binding doc for more readability by adding extra blank lines and indentations. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/imx-audmux.txt | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt index f88a00e..b30a737 100644 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt @@ -1,18 +1,24 @@ Freescale Digital Audio Mux (AUDMUX) device Required properties: -- compatible : "fsl,imx21-audmux" for AUDMUX version firstly used on i.MX21, - or "fsl,imx31-audmux" for the version firstly used on i.MX31. -- reg : Should contain AUDMUX registers location and length + + - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used + on i.MX21, or "fsl,imx31-audmux" for the version + firstly used on i.MX31. + + - reg : Should contain AUDMUX registers location and length. An initial configuration can be setup using child nodes. Required properties of optional child nodes: -- fsl,audmux-port : Integer of the audmux port that is configured by this - child node. -- fsl,port-config : List of configuration options for the specific port. For - imx31-audmux and above, it is a list of tuples . For - imx21-audmux it is a list of pcr values. + + - fsl,audmux-port : Integer of the audmux port that is configured by this + child node. + + - fsl,port-config : List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples + . For imx21-audmux it is a list of pcr + values. Example: -- cgit v1.1 From afa1fde676592b9c40e893f4f652f2c842e31683 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:09 -0700 Subject: ASoC: imx-sgtl5000: Add indentation for binding doc to increase readability This patch refines the DT binding doc for more readability by adding extra blank lines and indentations. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../bindings/sound/imx-audio-sgtl5000.txt | 61 ++++++++++++---------- 1 file changed, 34 insertions(+), 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt index e4acdd8..2f89db8 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt @@ -1,33 +1,40 @@ Freescale i.MX audio complex with SGTL5000 codec Required properties: -- compatible : "fsl,imx-audio-sgtl5000" -- model : The user-visible name of this sound complex -- ssi-controller : The phandle of the i.MX SSI controller -- audio-codec : The phandle of the SGTL5000 audio codec -- audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names could be power - supplies, SGTL5000 pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - SGTL5000 pins: - * MIC_IN - * LINE_IN - * HP_OUT - * LINE_OUT - - Board connectors: - * Mic Jack - * Line In Jack - * Headphone Jack - * Line Out Jack - * Ext Spk - -- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) -- mux-ext-port : The external port of the i.MX audio muxer + + - compatible : "fsl,imx-audio-sgtl5000" + + - model : The user-visible name of this sound complex + + - ssi-controller : The phandle of the i.MX SSI controller + + - audio-codec : The phandle of the SGTL5000 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, SGTL5000 + pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + SGTL5000 pins: + * MIC_IN + * LINE_IN + * HP_OUT + * LINE_OUT + + Board connectors: + * Mic Jack + * Line In Jack + * Headphone Jack + * Line Out Jack + * Ext Spk + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer Note: The AUDMUX port numbering should start at 1, which is consistent with hardware manual. -- cgit v1.1 From 6219b082b3099668c32be99bd31216379fc3d97a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:10 -0700 Subject: ASoC: imx-spdif: Add indentation for binding doc to increase readability This patch simply adds indentations for DT binding doc to increase readability without changing any contents. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/imx-audio-spdif.txt | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt index 7d13479..da84a44 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -2,23 +2,25 @@ Freescale i.MX audio complex with S/PDIF transceiver Required properties: - - compatible : "fsl,imx-audio-spdif" + - compatible : "fsl,imx-audio-spdif" - - model : The user-visible name of this sound complex + - model : The user-visible name of this sound complex - - spdif-controller : The phandle of the i.MX S/PDIF controller + - spdif-controller : The phandle of the i.MX S/PDIF controller Optional properties: - - spdif-out : This is a boolean property. If present, the transmitting - function of S/PDIF will be enabled, indicating there's a physical - S/PDIF out connector/jack on the board or it's connecting to some - other IP block, such as an HDMI encoder/display-controller. + - spdif-out : This is a boolean property. If present, the + transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector + or jack on the board or it's connecting to some + other IP block, such as an HDMI encoder or + display-controller. - - spdif-in : This is a boolean property. If present, the receiving - function of S/PDIF will be enabled, indicating there's a physical - S/PDIF in connector/jack on the board. + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there + is a physical S/PDIF in connector/jack on the board. * Note: At least one of these two properties should be set in the DT binding. -- cgit v1.1 From 6a6dec83e5abbc5003dac234970b51afa142defd Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 7 Oct 2014 12:29:11 -0700 Subject: ASoC: imx-wm8962: Add indentation for binding doc to increase readability This patch simply adds indentations for DT binding doc to increase readability without changing any contents. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 +++++++++++++--------- 1 file changed, 26 insertions(+), 19 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt index f49450a..acea71b 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -1,25 +1,32 @@ Freescale i.MX audio complex with WM8962 codec Required properties: -- compatible : "fsl,imx-audio-wm8962" -- model : The user-visible name of this sound complex -- ssi-controller : The phandle of the i.MX SSI controller -- audio-codec : The phandle of the WM8962 audio codec -- audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names could be power - supplies, WM8962 pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - -- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) -- mux-ext-port : The external port of the i.MX audio muxer + + - compatible : "fsl,imx-audio-wm8962" + + - model : The user-visible name of this sound complex + + - ssi-controller : The phandle of the i.MX SSI controller + + - audio-codec : The phandle of the WM8962 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, WM8962 + pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) + + - mux-ext-port : The external port of the i.MX audio muxer Note: The AUDMUX port numbering should start at 1, which is consistent with hardware manual. -- cgit v1.1 From 74d813cf37c210e94d155b0c19598fe269b8f78c Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 14 Oct 2014 20:29:26 +0300 Subject: ASoC: hdmi: Mark the maximum significant bits to HDMI codec HDMI audio can not have more than 24 bits even if on i2s bus there would be 32 bit samples. Mark this by adding .sig_bits = 24 to playback stream definition. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 1087fd5..2a52b90 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -47,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, }, .capture = { .stream_name = "Capture", -- cgit v1.1 From 69434097916bc52a4d6d495a0d719eb02e0cff9e Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 14 Oct 2014 20:29:27 +0300 Subject: ASoC: hdmi: HDMI codec doesn't benefit from pmdown delay Adds .ignore_pmdown_time = true to codec driver struct. HDMI codec is currently a dummy codec and doesn't benefit from pmdown delay. Even if in the future the codec would controll HDMI encoder, it would still be a digital to digital interface that should have no need for pmdown delay. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 2a52b90..1391ad5 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -76,6 +76,7 @@ static struct snd_soc_codec_driver hdmi_codec = { .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), .dapm_routes = hdmi_routes, .num_dapm_routes = ARRAY_SIZE(hdmi_routes), + .ignore_pmdown_time = true, }; static int hdmi_codec_probe(struct platform_device *pdev) -- cgit v1.1 From 4fa805738e497c6f5bad53fcdc76b9759bc9dc80 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:12:56 +0530 Subject: ASoC: Intel: mrfld: add the gain controls The DSP has various gain modules in the path, add these as ALSA gain controls Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/sst-atom-controls.c | 202 +++++++++++++++++++++++++++++++++++- sound/soc/intel/sst-atom-controls.h | 121 +++++++++++++++++++++ 3 files changed, 322 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f5b4a9c7..726f7d8 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,6 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC + depends on INTEL_SCU_IPC || COMPILE_TEST select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM help diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 7104a34..a00d506 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -162,6 +162,190 @@ static int sst_algo_control_set(struct snd_kcontrol *kcontrol, return ret; } +static int sst_gain_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = mc->stereo ? 2 : 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} + +/** + * sst_send_gain_cmd - send the gain algorithm IPC to the FW + * @gv: the stored value of gain (also contains rampduration) + * @mute: flag that indicates whether this was called from the + * digital_mute callback or directly. If called from the + * digital_mute callback, module will be muted/unmuted based on this + * flag. The flag is always 0 if called directly. + * + * Called with sst_data.lock held + * + * The user-set gain value is sent only if the user-controllable 'mute' control + * is OFF (indicated by gv->mute). Otherwise, the mute value (MIN value) is + * sent. + */ +static int sst_send_gain_cmd(struct sst_data *drv, struct sst_gain_value *gv, + u16 task_id, u16 loc_id, u16 module_id, int mute) +{ + struct sst_cmd_set_gain_dual cmd; + + dev_dbg(&drv->pdev->dev, "Enter\n"); + + cmd.header.command_id = MMX_SET_GAIN; + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.gain_cell_num = 1; + + if (mute || gv->mute) { + cmd.cell_gains[0].cell_gain_left = SST_GAIN_MIN_VALUE; + cmd.cell_gains[0].cell_gain_right = SST_GAIN_MIN_VALUE; + } else { + cmd.cell_gains[0].cell_gain_left = gv->l_gain; + cmd.cell_gains[0].cell_gain_right = gv->r_gain; + } + + SST_FILL_DESTINATION(2, cmd.cell_gains[0].dest, + loc_id, module_id); + cmd.cell_gains[0].gain_time_constant = gv->ramp_duration; + + cmd.header.length = sizeof(struct sst_cmd_set_gain_dual) + - sizeof(struct sst_dsp_header); + + /* we are with lock held, so call the unlocked api to send */ + return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS, + SST_FLAG_BLOCKED, task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +static int sst_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + struct sst_gain_value *gv = mc->gain_val; + + switch (mc->type) { + case SST_GAIN_TLV: + ucontrol->value.integer.value[0] = gv->l_gain; + ucontrol->value.integer.value[1] = gv->r_gain; + break; + + case SST_GAIN_MUTE: + ucontrol->value.integer.value[0] = gv->mute ? 1 : 0; + break; + + case SST_GAIN_RAMP_DURATION: + ucontrol->value.integer.value[0] = gv->ramp_duration; + break; + + default: + dev_err(component->dev, "Invalid Input- gain type:%d\n", + mc->type); + return -EINVAL; + }; + + return 0; +} + +static int sst_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = 0; + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt); + struct sst_gain_mixer_control *mc = (void *)kcontrol->private_value; + struct sst_gain_value *gv = mc->gain_val; + + mutex_lock(&drv->lock); + + switch (mc->type) { + case SST_GAIN_TLV: + gv->l_gain = ucontrol->value.integer.value[0]; + gv->r_gain = ucontrol->value.integer.value[1]; + dev_dbg(cmpnt->dev, "%s: Volume %d, %d\n", + mc->pname, gv->l_gain, gv->r_gain); + break; + + case SST_GAIN_MUTE: + gv->mute = !!ucontrol->value.integer.value[0]; + dev_dbg(cmpnt->dev, "%s: Mute %d\n", mc->pname, gv->mute); + break; + + case SST_GAIN_RAMP_DURATION: + gv->ramp_duration = ucontrol->value.integer.value[0]; + dev_dbg(cmpnt->dev, "%s: Ramp Delay%d\n", + mc->pname, gv->ramp_duration); + break; + + default: + mutex_unlock(&drv->lock); + dev_err(cmpnt->dev, "Invalid Input- gain type:%d\n", + mc->type); + return -EINVAL; + }; + + if (mc->w && mc->w->power) + ret = sst_send_gain_cmd(drv, gv, mc->task_id, + mc->pipe_id | mc->instance_id, mc->module_id, 0); + mutex_unlock(&drv->lock); + + return ret; +} + +static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0); + +/* Gain helper with min/max set */ +#define SST_GAIN(name, path_id, task_id, instance, gain_var) \ + SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \ + SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \ + sst_gain_get, sst_gain_put, \ + SST_MODULE_ID_GAIN_CELL, path_id, instance, task_id, \ + sst_gain_tlv_common, gain_var) + +#define SST_VOLUME(name, path_id, task_id, instance, gain_var) \ + SST_GAIN_KCONTROLS(name, "Volume", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \ + SST_GAIN_TC_MIN, SST_GAIN_TC_MAX, \ + sst_gain_get, sst_gain_put, \ + SST_MODULE_ID_VOLUME, path_id, instance, task_id, \ + sst_gain_tlv_common, gain_var) + +static struct sst_gain_value sst_gains[]; + +static const struct snd_kcontrol_new sst_gain_controls[] = { + SST_GAIN("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[0]), + SST_GAIN("media1_in", SST_PATH_INDEX_MEDIA1_IN, SST_TASK_MMX, 0, &sst_gains[1]), + SST_GAIN("media2_in", SST_PATH_INDEX_MEDIA2_IN, SST_TASK_MMX, 0, &sst_gains[2]), + SST_GAIN("media3_in", SST_PATH_INDEX_MEDIA3_IN, SST_TASK_MMX, 0, &sst_gains[3]), + + SST_GAIN("pcm0_in", SST_PATH_INDEX_PCM0_IN, SST_TASK_SBA, 0, &sst_gains[4]), + SST_GAIN("pcm1_in", SST_PATH_INDEX_PCM1_IN, SST_TASK_SBA, 0, &sst_gains[5]), + SST_GAIN("pcm1_out", SST_PATH_INDEX_PCM1_OUT, SST_TASK_SBA, 0, &sst_gains[6]), + SST_GAIN("pcm2_out", SST_PATH_INDEX_PCM2_OUT, SST_TASK_SBA, 0, &sst_gains[7]), + + SST_GAIN("codec_in0", SST_PATH_INDEX_CODEC_IN0, SST_TASK_SBA, 0, &sst_gains[8]), + SST_GAIN("codec_in1", SST_PATH_INDEX_CODEC_IN1, SST_TASK_SBA, 0, &sst_gains[9]), + SST_GAIN("codec_out0", SST_PATH_INDEX_CODEC_OUT0, SST_TASK_SBA, 0, &sst_gains[10]), + SST_GAIN("codec_out1", SST_PATH_INDEX_CODEC_OUT1, SST_TASK_SBA, 0, &sst_gains[11]), + SST_GAIN("media_loop1_out", SST_PATH_INDEX_MEDIA_LOOP1_OUT, SST_TASK_SBA, 0, &sst_gains[12]), + SST_GAIN("media_loop2_out", SST_PATH_INDEX_MEDIA_LOOP2_OUT, SST_TASK_SBA, 0, &sst_gains[13]), + SST_GAIN("sprot_loop_out", SST_PATH_INDEX_SPROT_LOOP_OUT, SST_TASK_SBA, 0, &sst_gains[14]), + SST_VOLUME("media0_in", SST_PATH_INDEX_MEDIA0_IN, SST_TASK_MMX, 0, &sst_gains[15]), +}; + +#define SST_GAIN_NUM_CONTROLS 3 +/* the SST_GAIN macro above will create three alsa controls for each + * instance invoked, gain, mute and ramp duration, which use the same gain + * cell sst_gain to keep track of data + * To calculate number of gain cell instances we need to device by 3 in + * below caulcation for gain cell memory. + * This gets rid of static number and issues while adding new controls + */ +static struct sst_gain_value sst_gains[ARRAY_SIZE(sst_gain_controls)/SST_GAIN_NUM_CONTROLS]; + static const struct snd_kcontrol_new sst_algo_controls[] = { SST_ALGO_KCONTROL_BYTES("media_loop1_out", "fir", 272, SST_MODULE_ID_FIR_24, SST_PATH_INDEX_MEDIA_LOOP1_OUT, 0, SST_TASK_SBA, SBA_VB_SET_FIR), @@ -200,19 +384,33 @@ static int sst_algo_control_init(struct device *dev) int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) { - int ret = 0; + int i, ret = 0; struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + unsigned int gains = ARRAY_SIZE(sst_gain_controls)/3; drv->byte_stream = devm_kzalloc(platform->dev, SST_MAX_BIN_BYTES, GFP_KERNEL); if (!drv->byte_stream) return -ENOMEM; - /*Initialize algo control params*/ + for (i = 0; i < gains; i++) { + sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT; + sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT; + sst_gains[i].r_gain = SST_GAIN_VOLUME_DEFAULT; + sst_gains[i].ramp_duration = SST_GAIN_RAMP_DURATION_DEFAULT; + } + + ret = snd_soc_add_platform_controls(platform, sst_gain_controls, + ARRAY_SIZE(sst_gain_controls)); + if (ret) + return ret; + + /* Initialize algo control params */ ret = sst_algo_control_init(platform->dev); if (ret) return ret; ret = snd_soc_add_platform_controls(platform, sst_algo_controls, ARRAY_SIZE(sst_algo_controls)); + return ret; } diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index a73e894..e530002 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -23,6 +23,9 @@ #ifndef __SST_ATOM_CONTROLS_H__ #define __SST_ATOM_CONTROLS_H__ +#include +#include + enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, @@ -360,16 +363,134 @@ struct sst_dsp_header { struct sst_cmd_generic { struct sst_dsp_header header; } __packed; + +struct gain_cell { + struct sst_destination_id dest; + s16 cell_gain_left; + s16 cell_gain_right; + u16 gain_time_constant; +} __packed; + +#define NUM_GAIN_CELLS 1 +struct sst_cmd_set_gain_dual { + struct sst_dsp_header header; + u16 gain_cell_num; + struct gain_cell cell_gains[NUM_GAIN_CELLS]; +} __packed; struct sst_cmd_set_params { struct sst_destination_id dst; u16 command_id; char params[0]; } __packed; + +/**** widget defines *****/ + +struct sst_ids { + u16 location_id; + u16 module_id; + u8 task_id; + u8 format; + u8 reg; + const char *parent_wname; + struct snd_soc_dapm_widget *parent_w; + struct list_head algo_list; + struct list_head gain_list; + const struct sst_pcm_format *pcm_fmt; +}; +enum sst_gain_kcontrol_type { + SST_GAIN_TLV, + SST_GAIN_MUTE, + SST_GAIN_RAMP_DURATION, +}; + +struct sst_gain_mixer_control { + bool stereo; + enum sst_gain_kcontrol_type type; + struct sst_gain_value *gain_val; + int max; + int min; + u16 instance_id; + u16 module_id; + u16 pipe_id; + u16 task_id; + char pname[44]; + struct snd_soc_dapm_widget *w; +}; + +struct sst_gain_value { + u16 ramp_duration; + s16 l_gain; + s16 r_gain; + bool mute; +}; +#define SST_GAIN_VOLUME_DEFAULT (-1440) +#define SST_GAIN_RAMP_DURATION_DEFAULT 5 /* timeconstant */ +#define SST_GAIN_MUTE_DEFAULT true + +#define SST_GAIN_KCONTROL_TLV(xname, xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, tlv_array, xgain_val, \ + xmin, xmax, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = sst_gain_ctl_info,\ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = true, .max = xmax, .min = xmin, .type = SST_GAIN_TLV, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} + +#define SST_GAIN_KCONTROL_INT(xname, xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, xtype, xgain_val, \ + xmin, xmax, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sst_gain_ctl_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = false, .max = xmax, .min = xmin, .type = xtype, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} + +#define SST_GAIN_KCONTROL_BOOL(xname, xhandler_get, xhandler_put,\ + xmod, xpipe, xinstance, xtask, xgain_val, xpname) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_bool_ext, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct sst_gain_mixer_control) \ + { .stereo = false, .type = SST_GAIN_MUTE, \ + .module_id = xmod, .pipe_id = xpipe, .task_id = xtask,\ + .instance_id = xinstance, .gain_val = xgain_val, .pname = xpname} #define SST_CONTROL_NAME(xpname, xmname, xinstance, xtype) \ xpname " " xmname " " #xinstance " " xtype #define SST_COMBO_CONTROL_NAME(xpname, xmname, xinstance, xtype, xsubmodule) \ xpname " " xmname " " #xinstance " " xtype " " xsubmodule + +/* + * 3 Controls for each Gain module + * e.g. - pcm0_in Gain 0 Volume + * - pcm0_in Gain 0 Ramp Delay + * - pcm0_in Gain 0 Switch + */ +#define SST_GAIN_KCONTROLS(xpname, xmname, xmin_gain, xmax_gain, xmin_tc, xmax_tc, \ + xhandler_get, xhandler_put, \ + xmod, xpipe, xinstance, xtask, tlv_array, xgain_val) \ + { SST_GAIN_KCONTROL_INT(SST_CONTROL_NAME(xpname, xmname, xinstance, "Ramp Delay"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, SST_GAIN_RAMP_DURATION, \ + xgain_val, xmin_tc, xmax_tc, xpname) }, \ + { SST_GAIN_KCONTROL_BOOL(SST_CONTROL_NAME(xpname, xmname, xinstance, "Switch"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, \ + xgain_val, xpname) } ,\ + { SST_GAIN_KCONTROL_TLV(SST_CONTROL_NAME(xpname, xmname, xinstance, "Volume"), \ + xhandler_get, xhandler_put, xmod, xpipe, xinstance, xtask, tlv_array, \ + xgain_val, xmin_gain, xmax_gain, xpname) } + +#define SST_GAIN_TC_MIN 5 +#define SST_GAIN_TC_MAX 5000 +#define SST_GAIN_MIN_VALUE -1440 /* in 0.1 DB units */ +#define SST_GAIN_MAX_VALUE 360 + enum sst_algo_kcontrol_type { SST_ALGO_PARAMS, SST_ALGO_BYPASS, -- cgit v1.1 From 24c8d14192cc63661ca049b423d7baaa0bbafeb3 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:12:57 +0530 Subject: ASoC: Intel: mrfld: add DSP core controls This patch adds core controls like interleavers, SSP BEs, and also logic of sending pipeline and module commands to the DSP. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 754 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.h | 307 +++++++++++++++ 2 files changed, 1061 insertions(+) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index a00d506..9239eff2 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -15,6 +15,9 @@ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * + * In the dpcm driver modelling when a particular FE/BE/Mixer/Pipe is active + * we forward the settings and parameters, rest we keep the values in + * driver and forward when DAPM enables them * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt @@ -81,6 +84,183 @@ static int sst_fill_and_send_cmd(struct sst_data *drv, return ret; } +/** + * tx map value is a bitfield where each bit represents a FW channel + * + * 3 2 1 0 # 0 = codec0, 1 = codec1 + * RLRLRLRL # 3, 4 = reserved + * + * e.g. slot 0 rx map = 00001100b -> data from slot 0 goes into codec_in1 L,R + */ +static u8 sst_ssp_tx_map[SST_MAX_TDM_SLOTS] = { + 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default rx map */ +}; + +/** + * rx map value is a bitfield where each bit represents a slot + * + * 76543210 # 0 = slot 0, 1 = slot 1 + * + * e.g. codec1_0 tx map = 00000101b -> data from codec_out1_0 goes into slot 0, 2 + */ +static u8 sst_ssp_rx_map[SST_MAX_TDM_SLOTS] = { + 0x1, 0x2, 0x4, 0x8, 0x10, 0x20, 0x40, 0x80, /* default tx map */ +}; + +/** + * NOTE: this is invoked with lock held + */ +static int sst_send_slot_map(struct sst_data *drv) +{ + struct sst_param_sba_ssp_slot_map cmd; + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.command_id = SBA_SET_SSP_SLOT_MAP; + cmd.header.length = sizeof(struct sst_param_sba_ssp_slot_map) + - sizeof(struct sst_dsp_header); + + cmd.param_id = SBA_SET_SSP_SLOT_MAP; + cmd.param_len = sizeof(cmd.rx_slot_map) + sizeof(cmd.tx_slot_map) + + sizeof(cmd.ssp_index); + cmd.ssp_index = SSP_CODEC; + + memcpy(cmd.rx_slot_map, &sst_ssp_tx_map[0], sizeof(cmd.rx_slot_map)); + memcpy(cmd.tx_slot_map, &sst_ssp_rx_map[0], sizeof(cmd.tx_slot_map)); + + return sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_SET_PARAMS, + SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +int sst_slot_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct sst_enum *e = (struct sst_enum *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = e->max; + + if (uinfo->value.enumerated.item > e->max - 1) + uinfo->value.enumerated.item = e->max - 1; + strcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item]); + + return 0; +} + +/** + * sst_slot_get - get the status of the interleaver/deinterleaver control + * + * Searches the map where the control status is stored, and gets the + * channel/slot which is currently set for this enumerated control. Since it is + * an enumerated control, there is only one possible value. + */ +static int sst_slot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sst_enum *e = (void *)kcontrol->private_value; + struct snd_soc_component *c = snd_kcontrol_chip(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + unsigned int ctl_no = e->reg; + unsigned int is_tx = e->tx; + unsigned int val, mux; + u8 *map = is_tx ? sst_ssp_rx_map : sst_ssp_tx_map; + + mutex_lock(&drv->lock); + val = 1 << ctl_no; + /* search which slot/channel has this bit set - there should be only one */ + for (mux = e->max; mux > 0; mux--) + if (map[mux - 1] & val) + break; + + ucontrol->value.enumerated.item[0] = mux; + mutex_unlock(&drv->lock); + + dev_dbg(c->dev, "%s - %s map = %#x\n", + is_tx ? "tx channel" : "rx slot", + e->texts[mux], mux ? map[mux - 1] : -1); + return 0; +} + +/* sst_check_and_send_slot_map - helper for checking power state and sending + * slot map cmd + * + * called with lock held + */ +static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol *kcontrol) +{ + struct sst_enum *e = (void *)kcontrol->private_value; + int ret = 0; + + if (e->w && e->w->power) + ret = sst_send_slot_map(drv); + else + dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", + kcontrol->id.name); + return ret; +} + +/** + * sst_slot_put - set the status of interleaver/deinterleaver control + * + * (de)interleaver controls are defined in opposite sense to be user-friendly + * + * Instead of the enum value being the value written to the register, it is the + * register address; and the kcontrol number (register num) is the value written + * to the register. This is so that there can be only one value for each + * slot/channel since there is only one control for each slot/channel. + * + * This means that whenever an enum is set, we need to clear the bit + * for that kcontrol_no for all the interleaver OR deinterleaver registers + */ +static int sst_slot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + struct sst_enum *e = (void *)kcontrol->private_value; + int i, ret = 0; + unsigned int ctl_no = e->reg; + unsigned int is_tx = e->tx; + unsigned int slot_channel_no; + unsigned int val, mux; + u8 *map; + + map = is_tx ? sst_ssp_rx_map : sst_ssp_tx_map; + + val = 1 << ctl_no; + mux = ucontrol->value.enumerated.item[0]; + if (mux > e->max - 1) + return -EINVAL; + + mutex_lock(&drv->lock); + /* first clear all registers of this bit */ + for (i = 0; i < e->max; i++) + map[i] &= ~val; + + if (mux == 0) { + /* kctl set to 'none' and we reset the bits so send IPC */ + ret = sst_check_and_send_slot_map(drv, kcontrol); + + mutex_unlock(&drv->lock); + return ret; + } + + /* offset by one to take "None" into account */ + slot_channel_no = mux - 1; + map[slot_channel_no] |= val; + + dev_dbg(c->dev, "%s %s map = %#x\n", + is_tx ? "tx channel" : "rx slot", + e->texts[mux], map[slot_channel_no]); + + ret = sst_check_and_send_slot_map(drv, kcontrol); + + mutex_unlock(&drv->lock); + return ret; +} + static int sst_send_algo_cmd(struct sst_data *drv, struct sst_algo_control *bc) { @@ -104,6 +284,34 @@ static int sst_send_algo_cmd(struct sst_data *drv, return ret; } +/** + * sst_find_and_send_pipe_algo - send all the algo parameters for a pipe + * + * The algos which are in each pipeline are sent to the firmware one by one + * + * Called with lock held + */ +static int sst_find_and_send_pipe_algo(struct sst_data *drv, + const char *pipe, struct sst_ids *ids) +{ + int ret = 0; + struct sst_algo_control *bc; + struct sst_module *algo = NULL; + + dev_dbg(&drv->pdev->dev, "Enter: widget=%s\n", pipe); + + list_for_each_entry(algo, &ids->algo_list, node) { + bc = (void *)algo->kctl->private_value; + + dev_dbg(&drv->pdev->dev, "Found algo control name=%s pipe=%s\n", + algo->kctl->id.name, pipe); + ret = sst_send_algo_cmd(drv, bc); + if (ret) + return ret; + } + return ret; +} + static int sst_algo_bytes_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -296,8 +504,317 @@ static int sst_gain_put(struct snd_kcontrol *kcontrol, return ret; } +static int sst_set_pipe_gain(struct sst_ids *ids, + struct sst_data *drv, int mute); + +static int sst_send_pipe_module_params(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol) +{ + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + struct sst_ids *ids = w->priv; + + mutex_lock(&drv->lock); + sst_find_and_send_pipe_algo(drv, w->name, ids); + sst_set_pipe_gain(ids, drv, 0); + mutex_unlock(&drv->lock); + + return 0; +} + +static int sst_generic_modules_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + return sst_send_pipe_module_params(w, k); + return 0; +} + static const DECLARE_TLV_DB_SCALE(sst_gain_tlv_common, SST_GAIN_MIN_VALUE * 10, 10, 0); +/* Look up table to convert MIXER SW bit regs to SWM inputs */ +static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = { + [SST_IP_CODEC0] = SST_SWM_IN_CODEC0, + [SST_IP_CODEC1] = SST_SWM_IN_CODEC1, + [SST_IP_LOOP0] = SST_SWM_IN_SPROT_LOOP, + [SST_IP_LOOP1] = SST_SWM_IN_MEDIA_LOOP1, + [SST_IP_LOOP2] = SST_SWM_IN_MEDIA_LOOP2, + [SST_IP_PCM0] = SST_SWM_IN_PCM0, + [SST_IP_PCM1] = SST_SWM_IN_PCM1, + [SST_IP_MEDIA0] = SST_SWM_IN_MEDIA0, + [SST_IP_MEDIA1] = SST_SWM_IN_MEDIA1, + [SST_IP_MEDIA2] = SST_SWM_IN_MEDIA2, + [SST_IP_MEDIA3] = SST_SWM_IN_MEDIA3, +}; + +/** + * called with lock held + */ +static int sst_set_pipe_gain(struct sst_ids *ids, + struct sst_data *drv, int mute) +{ + int ret = 0; + struct sst_gain_mixer_control *mc; + struct sst_gain_value *gv; + struct sst_module *gain = NULL; + + list_for_each_entry(gain, &ids->gain_list, node) { + struct snd_kcontrol *kctl = gain->kctl; + + dev_dbg(&drv->pdev->dev, "control name=%s\n", kctl->id.name); + mc = (void *)kctl->private_value; + gv = mc->gain_val; + + ret = sst_send_gain_cmd(drv, gv, mc->task_id, + mc->pipe_id | mc->instance_id, mc->module_id, mute); + if (ret) + return ret; + } + return ret; +} + +/* + * sst_handle_vb_timer - Start/Stop the DSP scheduler + * + * The DSP expects first cmd to be SBA_VB_START, so at first startup send + * that. + * DSP expects last cmd to be SBA_VB_IDLE, so at last shutdown send that. + * + * Do refcount internally so that we send command only at first start + * and last end. Since SST driver does its own ref count, invoke sst's + * power ops always! + */ +int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable) +{ + int ret = 0; + struct sst_cmd_generic cmd; + struct sst_data *drv = snd_soc_dai_get_drvdata(dai); + static int timer_usage; + + if (enable) + cmd.header.command_id = SBA_VB_START; + else + cmd.header.command_id = SBA_IDLE; + dev_dbg(dai->dev, "enable=%u, usage=%d\n", enable, timer_usage); + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.length = 0; + + if (enable) { + ret = sst->ops->power(sst->dev, true); + if (ret < 0) + return ret; + } + + mutex_lock(&drv->lock); + if (enable) + timer_usage++; + else + timer_usage--; + + /* + * Send the command only if this call is the first enable or last + * disable + */ + if ((enable && (timer_usage == 1)) || + (!enable && (timer_usage == 0))) { + ret = sst_fill_and_send_cmd_unlocked(drv, SST_IPC_IA_CMD, + SST_FLAG_BLOCKED, SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + if (ret && enable) { + timer_usage--; + enable = false; + } + } + mutex_unlock(&drv->lock); + + if (!enable) + sst->ops->power(sst->dev, false); + return ret; +} + +/** + * sst_ssp_config - contains SSP configuration for media UC + */ +static const struct sst_ssp_config sst_ssp_configs = { + .ssp_id = SSP_CODEC, + .bits_per_slot = 24, + .slots = 4, + .ssp_mode = SSP_MODE_MASTER, + .pcm_mode = SSP_PCM_MODE_NETWORK, + .duplex = SSP_DUPLEX, + .ssp_protocol = SSP_MODE_PCM, + .fs_width = 1, + .fs_frequency = SSP_FS_48_KHZ, + .active_slot_map = 0xF, + .start_delay = 0, +}; + +int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable) +{ + struct sst_cmd_sba_hw_set_ssp cmd; + struct sst_data *drv = snd_soc_dai_get_drvdata(dai); + const struct sst_ssp_config *config; + + dev_info(dai->dev, "Enter: enable=%d port_name=%s\n", enable, id); + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + cmd.header.command_id = SBA_HW_SET_SSP; + cmd.header.length = sizeof(struct sst_cmd_sba_hw_set_ssp) + - sizeof(struct sst_dsp_header); + + config = &sst_ssp_configs; + dev_dbg(dai->dev, "ssp_id: %u\n", config->ssp_id); + + if (enable) + cmd.switch_state = SST_SWITCH_ON; + else + cmd.switch_state = SST_SWITCH_OFF; + + cmd.selection = config->ssp_id; + cmd.nb_bits_per_slots = config->bits_per_slot; + cmd.nb_slots = config->slots; + cmd.mode = config->ssp_mode | (config->pcm_mode << 1); + cmd.duplex = config->duplex; + cmd.active_tx_slot_map = config->active_slot_map; + cmd.active_rx_slot_map = config->active_slot_map; + cmd.frame_sync_frequency = config->fs_frequency; + cmd.frame_sync_polarity = SSP_FS_ACTIVE_HIGH; + cmd.data_polarity = 1; + cmd.frame_sync_width = config->fs_width; + cmd.ssp_protocol = config->ssp_protocol; + cmd.start_delay = config->start_delay; + cmd.reserved1 = cmd.reserved2 = 0xFF; + + return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +static int sst_set_be_modules(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + + dev_dbg(c->dev, "Enter: widget=%s\n", w->name); + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = sst_send_slot_map(drv); + if (ret) + return ret; + ret = sst_send_pipe_module_params(w, k); + } + return ret; +} + +static int sst_set_media_path(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct sst_cmd_set_media_path cmd; + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + struct sst_ids *ids = w->priv; + + dev_dbg(c->dev, "widget=%s\n", w->name); + dev_dbg(c->dev, "task=%u, location=%#x\n", + ids->task_id, ids->location_id); + + if (SND_SOC_DAPM_EVENT_ON(event)) + cmd.switch_state = SST_PATH_ON; + else + cmd.switch_state = SST_PATH_OFF; + + SST_FILL_DESTINATION(2, cmd.header.dst, + ids->location_id, SST_DEFAULT_MODULE_ID); + + /* MMX_SET_MEDIA_PATH == SBA_SET_MEDIA_PATH */ + cmd.header.command_id = MMX_SET_MEDIA_PATH; + cmd.header.length = sizeof(struct sst_cmd_set_media_path) + - sizeof(struct sst_dsp_header); + + ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + ids->task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + if (ret) + return ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = sst_send_pipe_module_params(w, k); + return ret; +} + +static int sst_set_media_loop(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + int ret = 0; + struct sst_cmd_sba_set_media_loop_map cmd; + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct sst_data *drv = snd_soc_component_get_drvdata(c); + struct sst_ids *ids = w->priv; + + dev_dbg(c->dev, "Enter:widget=%s\n", w->name); + if (SND_SOC_DAPM_EVENT_ON(event)) + cmd.switch_state = SST_SWITCH_ON; + else + cmd.switch_state = SST_SWITCH_OFF; + + SST_FILL_DESTINATION(2, cmd.header.dst, + ids->location_id, SST_DEFAULT_MODULE_ID); + + cmd.header.command_id = SBA_SET_MEDIA_LOOP_MAP; + cmd.header.length = sizeof(struct sst_cmd_sba_set_media_loop_map) + - sizeof(struct sst_dsp_header); + cmd.param.part.cfg.rate = 2; /* 48khz */ + + cmd.param.part.cfg.format = ids->format; /* stereo/Mono */ + cmd.param.part.cfg.s_length = 1; /* 24bit left justified */ + cmd.map = 0; /* Algo sequence: Gain - DRP - FIR - IIR */ + + ret = sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + SST_TASK_SBA, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); + if (ret) + return ret; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = sst_send_pipe_module_params(w, k); + return ret; +} + +static const char * const slot_names[] = { + "none", + "slot 0", "slot 1", "slot 2", "slot 3", + "slot 4", "slot 5", "slot 6", "slot 7", /* not supported by FW */ +}; + +static const char * const channel_names[] = { + "none", + "codec_out0_0", "codec_out0_1", "codec_out1_0", "codec_out1_1", + "codec_out2_0", "codec_out2_1", "codec_out3_0", "codec_out3_1", /* not supported by FW */ +}; + +#define SST_INTERLEAVER(xpname, slot_name, slotno) \ + SST_SSP_SLOT_CTL(xpname, "tx interleaver", slot_name, slotno, true, \ + channel_names, sst_slot_get, sst_slot_put) + +#define SST_DEINTERLEAVER(xpname, channel_name, channel_no) \ + SST_SSP_SLOT_CTL(xpname, "rx deinterleaver", channel_name, channel_no, false, \ + slot_names, sst_slot_get, sst_slot_put) + +static const struct snd_kcontrol_new sst_slot_controls[] = { + SST_INTERLEAVER("codec_out", "slot 0", 0), + SST_INTERLEAVER("codec_out", "slot 1", 1), + SST_INTERLEAVER("codec_out", "slot 2", 2), + SST_INTERLEAVER("codec_out", "slot 3", 3), + SST_DEINTERLEAVER("codec_in", "codec_in0_0", 0), + SST_DEINTERLEAVER("codec_in", "codec_in0_1", 1), + SST_DEINTERLEAVER("codec_in", "codec_in1_0", 2), + SST_DEINTERLEAVER("codec_in", "codec_in1_1", 3), +}; + /* Gain helper with min/max set */ #define SST_GAIN(name, path_id, task_id, instance, gain_var) \ SST_GAIN_KCONTROLS(name, "Gain", SST_GAIN_MIN_VALUE, SST_GAIN_MAX_VALUE, \ @@ -382,6 +899,234 @@ static int sst_algo_control_init(struct device *dev) return 0; } +static bool is_sst_dapm_widget(struct snd_soc_dapm_widget *w) +{ + switch (w->id) { + case snd_soc_dapm_pga: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_mixer: + return true; + default: + return false; + } +} + +/** + * sst_send_pipe_gains - send gains for the front-end DAIs + * + * The gains in the pipes connected to the front-ends are muted/unmuted + * automatically via the digital_mute() DAPM callback. This function sends the + * gains for the front-end pipes. + */ +int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) +{ + struct sst_data *drv = snd_soc_dai_get_drvdata(dai); + struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_path *p = NULL; + + dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev_dbg(dai->dev, "Stream name=%s\n", + dai->playback_widget->name); + w = dai->playback_widget; + list_for_each_entry(p, &w->sinks, list_source) { + if (p->connected && !p->connected(w, p->sink)) + continue; + + if (p->connect && p->sink->power && + is_sst_dapm_widget(p->sink)) { + struct sst_ids *ids = p->sink->priv; + + dev_dbg(dai->dev, "send gains for widget=%s\n", + p->sink->name); + mutex_lock(&drv->lock); + sst_set_pipe_gain(ids, drv, mute); + mutex_unlock(&drv->lock); + } + } + } else { + dev_dbg(dai->dev, "Stream name=%s\n", + dai->capture_widget->name); + w = dai->capture_widget; + list_for_each_entry(p, &w->sources, list_sink) { + if (p->connected && !p->connected(w, p->sink)) + continue; + + if (p->connect && p->source->power && + is_sst_dapm_widget(p->source)) { + struct sst_ids *ids = p->source->priv; + + dev_dbg(dai->dev, "send gain for widget=%s\n", + p->source->name); + mutex_lock(&drv->lock); + sst_set_pipe_gain(ids, drv, mute); + mutex_unlock(&drv->lock); + } + } + } + return 0; +} + +/** + * sst_fill_module_list - populate the list of modules/gains for a pipe + * + * + * Fills the widget pointer in the kcontrol private data, and also fills the + * kcontrol pointer in the widget private data. + * + * Widget pointer is used to send the algo/gain in the .put() handler if the + * widget is powerd on. + * + * Kcontrol pointer is used to send the algo/gain in the widget power ON/OFF + * event handler. Each widget (pipe) has multiple algos stored in the algo_list. + */ +static int sst_fill_module_list(struct snd_kcontrol *kctl, + struct snd_soc_dapm_widget *w, int type) +{ + struct sst_module *module = NULL; + struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); + struct sst_ids *ids = w->priv; + int ret = 0; + + module = devm_kzalloc(c->dev, sizeof(*module), GFP_KERNEL); + if (!module) + return -ENOMEM; + + if (type == SST_MODULE_GAIN) { + struct sst_gain_mixer_control *mc = (void *)kctl->private_value; + + mc->w = w; + module->kctl = kctl; + list_add_tail(&module->node, &ids->gain_list); + } else if (type == SST_MODULE_ALGO) { + struct sst_algo_control *bc = (void *)kctl->private_value; + + bc->w = w; + module->kctl = kctl; + list_add_tail(&module->node, &ids->algo_list); + } else { + dev_err(c->dev, "invoked for unknown type %d module %s", + type, kctl->id.name); + ret = -EINVAL; + } + + return ret; +} + +/** + * sst_fill_widget_module_info - fill list of gains/algos for the pipe + * @widget: pipe modelled as a DAPM widget + * + * Fill the list of gains/algos for the widget by looking at all the card + * controls and comparing the name of the widget with the first part of control + * name. First part of control name contains the pipe name (widget name). + */ +static int sst_fill_widget_module_info(struct snd_soc_dapm_widget *w, + struct snd_soc_platform *platform) +{ + struct snd_kcontrol *kctl; + int index, ret = 0; + struct snd_card *card = platform->component.card->snd_card; + char *idx; + + down_read(&card->controls_rwsem); + + list_for_each_entry(kctl, &card->controls, list) { + idx = strstr(kctl->id.name, " "); + if (idx == NULL) + continue; + index = strlen(kctl->id.name) - strlen(idx); + + if (strstr(kctl->id.name, "Volume") && + !strncmp(kctl->id.name, w->name, index)) + ret = sst_fill_module_list(kctl, w, SST_MODULE_GAIN); + + else if (strstr(kctl->id.name, "params") && + !strncmp(kctl->id.name, w->name, index)) + ret = sst_fill_module_list(kctl, w, SST_MODULE_ALGO); + + else if (strstr(kctl->id.name, "Switch") && + !strncmp(kctl->id.name, w->name, index) && + strstr(kctl->id.name, "Gain")) { + struct sst_gain_mixer_control *mc = + (void *)kctl->private_value; + + mc->w = w; + + } else if (strstr(kctl->id.name, "interleaver") && + !strncmp(kctl->id.name, w->name, index)) { + struct sst_enum *e = (void *)kctl->private_value; + + e->w = w; + + } else if (strstr(kctl->id.name, "deinterleaver") && + !strncmp(kctl->id.name, w->name, index)) { + + struct sst_enum *e = (void *)kctl->private_value; + + e->w = w; + } + + if (ret < 0) { + up_read(&card->controls_rwsem); + return ret; + } + } + + up_read(&card->controls_rwsem); + return 0; +} + +/** + * sst_fill_linked_widgets - fill the parent pointer for the linked widget + */ +static void sst_fill_linked_widgets(struct snd_soc_platform *platform, + struct sst_ids *ids) +{ + struct snd_soc_dapm_widget *w; + unsigned int len = strlen(ids->parent_wname); + + list_for_each_entry(w, &platform->component.card->widgets, list) { + if (!strncmp(ids->parent_wname, w->name, len)) { + ids->parent_w = w; + break; + } + } +} + +/** + * sst_map_modules_to_pipe - fill algo/gains list for all pipes + */ +static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) +{ + struct snd_soc_dapm_widget *w; + int ret = 0; + + list_for_each_entry(w, &platform->component.card->widgets, list) { + if (platform && is_sst_dapm_widget(w) && (w->priv)) { + struct sst_ids *ids = w->priv; + + dev_dbg(platform->dev, "widget type=%d name=%s\n", + w->id, w->name); + INIT_LIST_HEAD(&ids->algo_list); + INIT_LIST_HEAD(&ids->gain_list); + ret = sst_fill_widget_module_info(w, platform); + + if (ret < 0) + return ret; + + /* fill linked widgets */ + if (ids->parent_wname != NULL) + sst_fill_linked_widgets(platform, ids); + } + } + return 0; +} + int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) { int i, ret = 0; @@ -411,6 +1156,15 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) return ret; ret = snd_soc_add_platform_controls(platform, sst_algo_controls, ARRAY_SIZE(sst_algo_controls)); + if (ret) + return ret; + + ret = snd_soc_add_platform_controls(platform, sst_slot_controls, + ARRAY_SIZE(sst_slot_controls)); + if (ret) + return ret; + + ret = sst_map_modules_to_pipe(platform); return ret; } diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index e530002..dfebfdd 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -364,6 +364,38 @@ struct sst_cmd_generic { struct sst_dsp_header header; } __packed; +struct swm_input_ids { + struct sst_destination_id input_id; +} __packed; + +struct sst_cmd_set_swm { + struct sst_dsp_header header; + struct sst_destination_id output_id; + u16 switch_state; + u16 nb_inputs; + struct swm_input_ids input[SST_CMD_SWM_MAX_INPUTS]; +} __packed; + +struct sst_cmd_set_media_path { + struct sst_dsp_header header; + u16 switch_state; +} __packed; + +struct pcm_cfg { + u8 s_length:2; + u8 rate:3; + u8 format:3; +} __packed; + +struct sst_cmd_set_speech_path { + struct sst_dsp_header header; + u16 switch_state; + struct { + u16 rsvd:8; + struct pcm_cfg cfg; + } config; +} __packed; + struct gain_cell { struct sst_destination_id dest; s16 cell_gain_left; @@ -383,8 +415,162 @@ struct sst_cmd_set_params { char params[0]; } __packed; + +struct sst_cmd_sba_vb_start { + struct sst_dsp_header header; +} __packed; + +union sba_media_loop_params { + struct { + u16 rsvd:8; + struct pcm_cfg cfg; + } part; + u16 full; +} __packed; + +struct sst_cmd_sba_set_media_loop_map { + struct sst_dsp_header header; + u16 switch_state; + union sba_media_loop_params param; + u16 map; +} __packed; + +struct sst_cmd_tone_stop { + struct sst_dsp_header header; + u16 switch_state; +} __packed; + +enum sst_ssp_mode { + SSP_MODE_MASTER = 0, + SSP_MODE_SLAVE = 1, +}; + +enum sst_ssp_pcm_mode { + SSP_PCM_MODE_NORMAL = 0, + SSP_PCM_MODE_NETWORK = 1, +}; + +enum sst_ssp_duplex { + SSP_DUPLEX = 0, + SSP_RX = 1, + SSP_TX = 2, +}; + +enum sst_ssp_fs_frequency { + SSP_FS_8_KHZ = 0, + SSP_FS_16_KHZ = 1, + SSP_FS_44_1_KHZ = 2, + SSP_FS_48_KHZ = 3, +}; + +enum sst_ssp_fs_polarity { + SSP_FS_ACTIVE_LOW = 0, + SSP_FS_ACTIVE_HIGH = 1, +}; + +enum sst_ssp_protocol { + SSP_MODE_PCM = 0, + SSP_MODE_I2S = 1, +}; + +enum sst_ssp_port_id { + SSP_MODEM = 0, + SSP_BT = 1, + SSP_FM = 2, + SSP_CODEC = 3, +}; + +struct sst_cmd_sba_hw_set_ssp { + struct sst_dsp_header header; + u16 selection; /* 0:SSP0(def), 1:SSP1, 2:SSP2 */ + + u16 switch_state; + + u16 nb_bits_per_slots:6; /* 0-32 bits, 24 (def) */ + u16 nb_slots:4; /* 0-8: slots per frame */ + u16 mode:3; /* 0:Master, 1: Slave */ + u16 duplex:3; + + u16 active_tx_slot_map:8; /* Bit map, 0:off, 1:on */ + u16 reserved1:8; + + u16 active_rx_slot_map:8; /* Bit map 0: Off, 1:On */ + u16 reserved2:8; + + u16 frame_sync_frequency; + + u16 frame_sync_polarity:8; + u16 data_polarity:8; + + u16 frame_sync_width; /* 1 to N clocks */ + u16 ssp_protocol:8; + u16 start_delay:8; /* Start delay in terms of clock ticks */ +} __packed; + +#define SST_MAX_TDM_SLOTS 8 + +struct sst_param_sba_ssp_slot_map { + struct sst_dsp_header header; + + u16 param_id; + u16 param_len; + u16 ssp_index; + + u8 rx_slot_map[SST_MAX_TDM_SLOTS]; + u8 tx_slot_map[SST_MAX_TDM_SLOTS]; +} __packed; + +enum { + SST_PROBE_EXTRACTOR = 0, + SST_PROBE_INJECTOR = 1, +}; + /**** widget defines *****/ +#define SST_MODULE_GAIN 1 +#define SST_MODULE_ALGO 2 + +#define SST_FMT_MONO 0 +#define SST_FMT_STEREO 3 + +/* physical SSP numbers */ +enum { + SST_SSP0 = 0, + SST_SSP1, + SST_SSP2, + SST_SSP_LAST = SST_SSP2, +}; + +#define SST_NUM_SSPS (SST_SSP_LAST + 1) /* physical SSPs */ +#define SST_MAX_SSP_MUX 2 /* single SSP muxed between pipes */ +#define SST_MAX_SSP_DOMAINS 2 /* domains present in each pipe */ + +struct sst_module { + struct snd_kcontrol *kctl; + struct list_head node; +}; + +struct sst_ssp_config { + u8 ssp_id; + u8 bits_per_slot; + u8 slots; + u8 ssp_mode; + u8 pcm_mode; + u8 duplex; + u8 ssp_protocol; + u8 fs_frequency; + u8 active_slot_map; + u8 start_delay; + u16 fs_width; +}; + +struct sst_ssp_cfg { + const u8 ssp_number; + const int *mux_shift; + const int (*domain_shift)[SST_MAX_SSP_MUX]; + const struct sst_ssp_config (*ssp_config)[SST_MAX_SSP_MUX][SST_MAX_SSP_DOMAINS]; +}; + struct sst_ids { u16 location_id; u16 module_id; @@ -397,6 +583,102 @@ struct sst_ids { struct list_head gain_list; const struct sst_pcm_format *pcm_fmt; }; + + +#define SST_AIF_IN(wname, wevent) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_AIF_OUT(wname, wevent) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_INPUT(wname, wevent) \ +{ .id = snd_soc_dapm_input, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_OUTPUT(wname, wevent) \ +{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .task_id = 0, .location_id = 0 } \ +} + +#define SST_DAPM_OUTPUT(wname, wloc_id, wtask_id, wformat, wevent) \ +{ .id = snd_soc_dapm_output, .name = wname, .sname = NULL, \ + .reg = SND_SOC_NOPM, .shift = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .priv = (void *)&(struct sst_ids) { .location_id = wloc_id, .task_id = wtask_id,\ + .pcm_fmt = wformat, } \ +} + +#define SST_PATH(wname, wtask, wloc_id, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, } \ +} + +#define SST_LINKED_PATH(wname, wtask, wloc_id, linked_wname, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .on_val = 1, .off_val = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .parent_wname = linked_wname} \ +} + +#define SST_PATH_MEDIA_LOOP(wname, wtask, wloc_id, wformat, wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = NULL, .num_kcontrols = 0, \ + .event = wevent, .event_flags = wflags, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .format = wformat,} \ +} + +/* output is triggered before input */ +#define SST_PATH_INPUT(name, task_id, loc_id, event) \ + SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) + +#define SST_PATH_LINKED_INPUT(name, task_id, loc_id, linked_wname, event) \ + SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \ + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD) + +#define SST_PATH_OUTPUT(name, task_id, loc_id, event) \ + SST_PATH(name, task_id, loc_id, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + +#define SST_PATH_LINKED_OUTPUT(name, task_id, loc_id, linked_wname, event) \ + SST_LINKED_PATH(name, task_id, loc_id, linked_wname, event, \ + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + +#define SST_PATH_MEDIA_LOOP_OUTPUT(name, task_id, loc_id, format, event) \ + SST_PATH_MEDIA_LOOP(name, task_id, loc_id, format, event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD) + + +#define SST_SWM_MIXER(wname, wreg, wtask, wloc_id, wcontrols, wevent) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = SND_SOC_NOPM, .shift = 0, \ + .kcontrol_news = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols),\ + .event = wevent, .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD | \ + SND_SOC_DAPM_POST_REG, \ + .priv = (void *)&(struct sst_ids) { .task_id = wtask, .location_id = wloc_id, \ + .reg = wreg } \ +} + enum sst_gain_kcontrol_type { SST_GAIN_TLV, SST_GAIN_MUTE, @@ -560,4 +842,29 @@ struct sst_enum { struct snd_soc_dapm_widget *w; }; +/* only 4 slots/channels supported atm */ +#define SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts) \ + (struct sst_enum){ .reg = s_ch_no, .tx = is_tx, .max = 4+1, .texts = xtexts, } + +#define SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name) \ + xpname " " xmname " " s_ch_name + +#define SST_SSP_SLOT_CTL(xpname, xmname, s_ch_name, s_ch_no, is_tx, xtexts, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = SST_SLOT_CTL_NAME(xpname, xmname, s_ch_name), \ + .info = sst_slot_enum_info, \ + .get = xget, .put = xput, \ + .private_value = (unsigned long)&SST_SSP_SLOT_ENUM(s_ch_no, is_tx, xtexts), \ +} + +#define SST_MUX_CTL_NAME(xpname, xinstance) \ + xpname " " #xinstance + +#define SST_SSP_MUX_ENUM(xreg, xshift, xtexts) \ + (struct soc_enum) SOC_ENUM_DOUBLE(xreg, xshift, xshift, ARRAY_SIZE(xtexts), xtexts) + +#define SST_SSP_MUX_CTL(xpname, xinstance, xreg, xshift, xtexts) \ + SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \ + SST_SSP_MUX_ENUM(xreg, xshift, xtexts)) + #endif -- cgit v1.1 From e4f5ccd050e5c366ee1301b5b318bdc2780ce94a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:12:58 +0530 Subject: ASoC: Intel: mrfld: add the DSP DAPM widgets This patch adds all DAPM widgets and the event handlers for DSP except the mixers. Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 226 ++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 4 + 2 files changed, 230 insertions(+) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 9239eff2..9aa09db 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -548,6 +548,41 @@ static const uint swm_mixer_input_ids[SST_SWM_INPUT_COUNT] = { }; /** + * fill_swm_input - fill in the SWM input ids given the register + * + * The register value is a bit-field inicated which mixer inputs are ON. Use the + * lookup table to get the input-id and fill it in the structure. + */ +static int fill_swm_input(struct snd_soc_component *cmpnt, + struct swm_input_ids *swm_input, unsigned int reg) +{ + uint i, is_set, nb_inputs = 0; + u16 input_loc_id; + + dev_dbg(cmpnt->dev, "reg: %#x\n", reg); + for (i = 0; i < SST_SWM_INPUT_COUNT; i++) { + is_set = reg & BIT(i); + if (!is_set) + continue; + + input_loc_id = swm_mixer_input_ids[i]; + SST_FILL_DESTINATION(2, swm_input->input_id, + input_loc_id, SST_DEFAULT_MODULE_ID); + nb_inputs++; + swm_input++; + dev_dbg(cmpnt->dev, "input id: %#x, nb_inputs: %d\n", + input_loc_id, nb_inputs); + + if (nb_inputs == SST_CMD_SWM_MAX_INPUTS) { + dev_warn(cmpnt->dev, "SET_SWM cmd max inputs reached"); + break; + } + } + return nb_inputs; +} + + +/** * called with lock held */ static int sst_set_pipe_gain(struct sst_ids *ids, @@ -573,6 +608,112 @@ static int sst_set_pipe_gain(struct sst_ids *ids, return ret; } +static int sst_swm_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct sst_cmd_set_swm cmd; + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct sst_data *drv = snd_soc_component_get_drvdata(cmpnt); + struct sst_ids *ids = w->priv; + bool set_mixer = false; + struct soc_mixer_control *mc; + int val = 0; + int i = 0; + + dev_dbg(cmpnt->dev, "widget = %s\n", w->name); + /* + * Identify which mixer input is on and send the bitmap of the + * inputs as an IPC to the DSP. + */ + for (i = 0; i < w->num_kcontrols; i++) { + if (dapm_kcontrol_get_value(w->kcontrols[i])) { + mc = (struct soc_mixer_control *)(w->kcontrols[i])->private_value; + val |= 1 << mc->shift; + } + } + dev_dbg(cmpnt->dev, "val = %#x\n", val); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMD: + set_mixer = true; + break; + case SND_SOC_DAPM_POST_REG: + if (w->power) + set_mixer = true; + break; + default: + set_mixer = false; + } + + if (set_mixer == false) + return 0; + + if (SND_SOC_DAPM_EVENT_ON(event) || + event == SND_SOC_DAPM_POST_REG) + cmd.switch_state = SST_SWM_ON; + else + cmd.switch_state = SST_SWM_OFF; + + SST_FILL_DEFAULT_DESTINATION(cmd.header.dst); + /* MMX_SET_SWM == SBA_SET_SWM */ + cmd.header.command_id = SBA_SET_SWM; + + SST_FILL_DESTINATION(2, cmd.output_id, + ids->location_id, SST_DEFAULT_MODULE_ID); + cmd.nb_inputs = fill_swm_input(cmpnt, &cmd.input[0], val); + cmd.header.length = offsetof(struct sst_cmd_set_swm, input) + - sizeof(struct sst_dsp_header) + + (cmd.nb_inputs * sizeof(cmd.input[0])); + + return sst_fill_and_send_cmd(drv, SST_IPC_IA_CMD, SST_FLAG_BLOCKED, + ids->task_id, 0, &cmd, + sizeof(cmd.header) + cmd.header.length); +} + +/* SBA mixers - 16 inputs */ +#define SST_SBA_DECLARE_MIX_CONTROLS(kctl_name) \ + static const struct snd_kcontrol_new kctl_name[] = { \ + SOC_DAPM_SINGLE("codec_in0 Switch", SND_SOC_NOPM, SST_IP_CODEC0, 1, 0), \ + SOC_DAPM_SINGLE("codec_in1 Switch", SND_SOC_NOPM, SST_IP_CODEC1, 1, 0), \ + SOC_DAPM_SINGLE("sprot_loop_in Switch", SND_SOC_NOPM, SST_IP_LOOP0, 1, 0), \ + SOC_DAPM_SINGLE("media_loop1_in Switch", SND_SOC_NOPM, SST_IP_LOOP1, 1, 0), \ + SOC_DAPM_SINGLE("media_loop2_in Switch", SND_SOC_NOPM, SST_IP_LOOP2, 1, 0), \ + SOC_DAPM_SINGLE("pcm0_in Switch", SND_SOC_NOPM, SST_IP_PCM0, 1, 0), \ + SOC_DAPM_SINGLE("pcm1_in Switch", SND_SOC_NOPM, SST_IP_PCM1, 1, 0), \ + } + +#define SST_SBA_MIXER_GRAPH_MAP(mix_name) \ + { mix_name, "codec_in0 Switch", "codec_in0" }, \ + { mix_name, "codec_in1 Switch", "codec_in1" }, \ + { mix_name, "sprot_loop_in Switch", "sprot_loop_in" }, \ + { mix_name, "media_loop1_in Switch", "media_loop1_in" }, \ + { mix_name, "media_loop2_in Switch", "media_loop2_in" }, \ + { mix_name, "pcm0_in Switch", "pcm0_in" }, \ + { mix_name, "pcm1_in Switch", "pcm1_in" } + +#define SST_MMX_DECLARE_MIX_CONTROLS(kctl_name) \ + static const struct snd_kcontrol_new kctl_name[] = { \ + SOC_DAPM_SINGLE("media0_in Switch", SND_SOC_NOPM, SST_IP_MEDIA0, 1, 0), \ + SOC_DAPM_SINGLE("media1_in Switch", SND_SOC_NOPM, SST_IP_MEDIA1, 1, 0), \ + SOC_DAPM_SINGLE("media2_in Switch", SND_SOC_NOPM, SST_IP_MEDIA2, 1, 0), \ + SOC_DAPM_SINGLE("media3_in Switch", SND_SOC_NOPM, SST_IP_MEDIA3, 1, 0), \ + } + +SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media0_controls); +SST_MMX_DECLARE_MIX_CONTROLS(sst_mix_media1_controls); + +/* 18 SBA mixers */ +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm0_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm1_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls); +SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls); + /* * sst_handle_vb_timer - Start/Stop the DSP scheduler * @@ -784,6 +925,83 @@ static int sst_set_media_loop(struct snd_soc_dapm_widget *w, return ret; } +static const struct snd_soc_dapm_widget sst_dapm_widgets[] = { + SST_AIF_IN("codec_in0", sst_set_be_modules), + SST_AIF_IN("codec_in1", sst_set_be_modules), + SST_AIF_OUT("codec_out0", sst_set_be_modules), + SST_AIF_OUT("codec_out1", sst_set_be_modules), + + /* Media Paths */ + /* MediaX IN paths are set via ALLOC, so no SET_MEDIA_PATH command */ + SST_PATH_INPUT("media0_in", SST_TASK_MMX, SST_SWM_IN_MEDIA0, sst_generic_modules_event), + SST_PATH_INPUT("media1_in", SST_TASK_MMX, SST_SWM_IN_MEDIA1, NULL), + SST_PATH_INPUT("media2_in", SST_TASK_MMX, SST_SWM_IN_MEDIA2, sst_set_media_path), + SST_PATH_INPUT("media3_in", SST_TASK_MMX, SST_SWM_IN_MEDIA3, NULL), + SST_PATH_OUTPUT("media0_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA0, sst_set_media_path), + SST_PATH_OUTPUT("media1_out", SST_TASK_MMX, SST_SWM_OUT_MEDIA1, sst_set_media_path), + + /* SBA PCM Paths */ + SST_PATH_INPUT("pcm0_in", SST_TASK_SBA, SST_SWM_IN_PCM0, sst_set_media_path), + SST_PATH_INPUT("pcm1_in", SST_TASK_SBA, SST_SWM_IN_PCM1, sst_set_media_path), + SST_PATH_OUTPUT("pcm0_out", SST_TASK_SBA, SST_SWM_OUT_PCM0, sst_set_media_path), + SST_PATH_OUTPUT("pcm1_out", SST_TASK_SBA, SST_SWM_OUT_PCM1, sst_set_media_path), + SST_PATH_OUTPUT("pcm2_out", SST_TASK_SBA, SST_SWM_OUT_PCM2, sst_set_media_path), + + /* SBA Loops */ + SST_PATH_INPUT("sprot_loop_in", SST_TASK_SBA, SST_SWM_IN_SPROT_LOOP, NULL), + SST_PATH_INPUT("media_loop1_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP1, NULL), + SST_PATH_INPUT("media_loop2_in", SST_TASK_SBA, SST_SWM_IN_MEDIA_LOOP2, NULL), + SST_PATH_MEDIA_LOOP_OUTPUT("sprot_loop_out", SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, SST_FMT_MONO, sst_set_media_loop), + SST_PATH_MEDIA_LOOP_OUTPUT("media_loop1_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, SST_FMT_MONO, sst_set_media_loop), + SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop), + + /* Media Mixers */ +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"media0_in", NULL, "Compress Playback"}, + {"media1_in", NULL, "Headset Playback"}, + {"media2_in", NULL, "pcm0_out"}, + + {"media0_out mix 0", "media0_in Switch", "media0_in"}, + {"media0_out mix 0", "media1_in Switch", "media1_in"}, + {"media0_out mix 0", "media2_in Switch", "media2_in"}, + {"media0_out mix 0", "media3_in Switch", "media3_in"}, + {"media1_out mix 0", "media0_in Switch", "media0_in"}, + {"media1_out mix 0", "media1_in Switch", "media1_in"}, + {"media1_out mix 0", "media2_in Switch", "media2_in"}, + {"media1_out mix 0", "media3_in Switch", "media3_in"}, + + {"media0_out", NULL, "media0_out mix 0"}, + {"media1_out", NULL, "media1_out mix 0"}, + {"pcm0_in", NULL, "media0_out"}, + {"pcm1_in", NULL, "media1_out"}, + + {"Headset Capture", NULL, "pcm1_out"}, + {"Headset Capture", NULL, "pcm2_out"}, + {"pcm0_out", NULL, "pcm0_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm0_out mix 0"), + {"pcm1_out", NULL, "pcm1_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm1_out mix 0"), + {"pcm2_out", NULL, "pcm2_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("pcm2_out mix 0"), + + {"media_loop1_in", NULL, "media_loop1_out"}, + {"media_loop1_out", NULL, "media_loop1_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("media_loop1_out mix 0"), + {"media_loop2_in", NULL, "media_loop2_out"}, + {"media_loop2_out", NULL, "media_loop2_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("media_loop2_out mix 0"), + {"sprot_loop_in", NULL, "sprot_loop_out"}, + {"sprot_loop_out", NULL, "sprot_loop_out mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("sprot_loop_out mix 0"), + + {"codec_out0", NULL, "codec_out0 mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("codec_out0 mix 0"), + {"codec_out1", NULL, "codec_out1 mix 0"}, + SST_SBA_MIXER_GRAPH_MAP("codec_out1 mix 0"), + +}; static const char * const slot_names[] = { "none", "slot 0", "slot 1", "slot 2", "slot 3", @@ -1130,6 +1348,8 @@ static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) { int i, ret = 0; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(&platform->component); struct sst_data *drv = snd_soc_platform_get_drvdata(platform); unsigned int gains = ARRAY_SIZE(sst_gain_controls)/3; @@ -1138,6 +1358,12 @@ int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) if (!drv->byte_stream) return -ENOMEM; + snd_soc_dapm_new_controls(dapm, sst_dapm_widgets, + ARRAY_SIZE(sst_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, + ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(dapm->card); + for (i = 0; i < gains; i++) { sst_gains[i].mute = SST_GAIN_MUTE_DEFAULT; sst_gains[i].l_gain = SST_GAIN_VOLUME_DEFAULT; diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 19f83ec..d41d1c3 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -153,6 +153,10 @@ struct sst_device { struct sst_data; int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); +int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute); +int send_ssp_cmd(struct snd_soc_dai *dai, const char *id, bool enable); +int sst_handle_vb_timer(struct snd_soc_dai *dai, bool enable); + void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); -- cgit v1.1 From c82351da2e9f2b14d5664e41b021ec1fd948b932 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:12:59 +0530 Subject: ASoC: Intel: mfld-pcm: add FE and BE ops Now that we have added code for managing DSP pipelines we need to add the code for DSPs FrontEnd and Backend dai. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 153 ++++++++++++++++++++++++++------ 1 file changed, 125 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index aa9b600..e7cf18d 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -101,35 +101,11 @@ static struct sst_dev_stream_map dpcm_strm_map[] = { {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, }; -/* MFLD - MSIC */ -static struct snd_soc_dai_driver sst_platform_dai[] = { +static int sst_media_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { - .name = "Headset-cpu-dai", - .id = 0, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 5, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "Compress-cpu-dai", - .compress_dai = 1, - .playback = { - .channels_min = SST_STEREO, - .channels_max = SST_STEREO, - .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -}; + + return sst_send_pipe_gains(dai, stream, mute); +} /* helper functions */ void sst_set_stream_status(struct sst_runtime_stream *stream, @@ -451,12 +427,133 @@ static int sst_media_hw_free(struct snd_pcm_substream *substream, return snd_pcm_lib_free_pages(substream); } +static int sst_enable_ssp(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + + if (!dai->active) { + ret = sst_handle_vb_timer(dai, true); + if (ret) + return ret; + ret = send_ssp_cmd(dai, dai->name, 1); + } + return ret; +} + +static void sst_disable_ssp(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (!dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } +} + static struct snd_soc_dai_ops sst_media_dai_ops = { .startup = sst_media_open, .shutdown = sst_media_close, .prepare = sst_media_prepare, .hw_params = sst_media_hw_params, .hw_free = sst_media_hw_free, + .mute_stream = sst_media_digital_mute, +}; + +static struct snd_soc_dai_ops sst_compr_dai_ops = { + .mute_stream = sst_media_digital_mute, +}; + +static struct snd_soc_dai_ops sst_be_dai_ops = { + .startup = sst_enable_ssp, + .shutdown = sst_disable_ssp, +}; + +static struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "media-cpu-dai", + .ops = &sst_media_dai_ops, + .playback = { + .stream_name = "Headset Playback", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Headset Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "compress-cpu-dai", + .compress_dai = 1, + .ops = &sst_compr_dai_ops, + .playback = { + .stream_name = "Compress Playback", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +/* BE CPU Dais */ +{ + .name = "ssp0-port", + .ops = &sst_be_dai_ops, + .playback = { + .stream_name = "ssp0 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp0 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "ssp1-port", + .ops = &sst_be_dai_ops, + .playback = { + .stream_name = "ssp1 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp1 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "ssp2-port", + .ops = &sst_be_dai_ops, + .playback = { + .stream_name = "ssp2 Tx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp2 Rx", + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, }; static int sst_platform_open(struct snd_pcm_substream *substream) -- cgit v1.1 From f2b3a93973ca7cda6e6365c0a8ff7c4438778a6f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 15 Oct 2014 20:13:00 +0530 Subject: ASoC: Intel: mrfld: add the DSP mixers Signed-off-by: Vinod Koul Signed-off-by: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 9aa09db..dcdeb28 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -956,6 +956,32 @@ static const struct snd_soc_dapm_widget sst_dapm_widgets[] = { SST_PATH_MEDIA_LOOP_OUTPUT("media_loop2_out", SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, SST_FMT_STEREO, sst_set_media_loop), /* Media Mixers */ + SST_SWM_MIXER("media0_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA0, + sst_mix_media0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media1_out mix 0", SND_SOC_NOPM, SST_TASK_MMX, SST_SWM_OUT_MEDIA1, + sst_mix_media1_controls, sst_swm_mixer_event), + + /* SBA PCM mixers */ + SST_SWM_MIXER("pcm0_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM0, + sst_mix_pcm0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("pcm1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM1, + sst_mix_pcm1_controls, sst_swm_mixer_event), + SST_SWM_MIXER("pcm2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_PCM2, + sst_mix_pcm2_controls, sst_swm_mixer_event), + + /* SBA Loop mixers */ + SST_SWM_MIXER("sprot_loop_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_SPROT_LOOP, + sst_mix_sprot_l0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media_loop1_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP1, + sst_mix_media_l1_controls, sst_swm_mixer_event), + SST_SWM_MIXER("media_loop2_out mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_MEDIA_LOOP2, + sst_mix_media_l2_controls, sst_swm_mixer_event), + + /* SBA Backend mixers */ + SST_SWM_MIXER("codec_out0 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC0, + sst_mix_codec0_controls, sst_swm_mixer_event), + SST_SWM_MIXER("codec_out1 mix 0", SND_SOC_NOPM, SST_TASK_SBA, SST_SWM_OUT_CODEC1, + sst_mix_codec1_controls, sst_swm_mixer_event), }; static const struct snd_soc_dapm_route intercon[] = { -- cgit v1.1 From 5914ccf47bb0954210d64a92396632442c4f2a80 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 13:54:29 +0530 Subject: ASoC: intel: turn off COMPILE_TEST for medfield Since medfield machine uses SCU_IPC which is not availble for all archs, so compile test fails on these Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 726f7d8..f5b4a9c7 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,6 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC || COMPILE_TEST + depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM help -- cgit v1.1 From f07e51c51e44a6e4e6d003f3bccbbf8a1b2cda0d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 16 Oct 2014 15:29:15 +0100 Subject: ASoC: Intel: Add TDM support to HSW/BDW SSP port Add TDM support to SSP port via DSP IPC SetDeviceFormat message. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 4 ++++ sound/soc/intel/sst-haswell-ipc.h | 4 +++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b629151..92d625a 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1630,6 +1630,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, config.clock_frequency = mclk; config.mode = mode; config.clock_divider = clock_divider; + if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) + config.channels = 4; + else + config.channels = 2; trace_hsw_device_config_req(&config); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 2ac194a..063dd6b 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -84,6 +84,7 @@ enum sst_hsw_device_mclk { enum sst_hsw_device_mode { SST_HSW_DEVICE_CLOCK_SLAVE = 0, SST_HSW_DEVICE_CLOCK_MASTER = 1, + SST_HSW_DEVICE_TDM_CLOCK_MASTER = 2, }; /* DX Power State */ @@ -295,7 +296,8 @@ struct sst_hsw_ipc_device_config_req { u32 clock_frequency; u32 mode; u16 clock_divider; - u16 reserved; + u8 channels; + u8 reserved; } __attribute__((packed)); /* Audio Data formats */ -- cgit v1.1 From 48dc326f6ba71ba0ee5b1bbfc128a6577ba98608 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 16 Oct 2014 15:29:16 +0100 Subject: ASoC: Intel: Add 4 channel support to DSP. DSP can now support 4 channels in certain use cases. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 4 ---- sound/soc/intel/sst-haswell-pcm.c | 8 +------- 2 files changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 92d625a..4799768 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1256,10 +1256,6 @@ int sst_hsw_stream_set_channels(struct sst_hsw *hsw, return -EINVAL; } - /* stereo is only supported atm */ - if (channels != 2) - return -EINVAL; - stream->request.format.ch_num = channels; return 0; } diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 33fc5c3..32a33b9 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -421,13 +421,7 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } - /* we only support stereo atm */ channels = params_channels(params); - if (channels != 2) { - dev_err(rtd->dev, "error: invalid channels %d\n", channels); - return -EINVAL; - } - map = create_channel_map(SST_HSW_CHANNEL_CONFIG_STEREO); sst_hsw_stream_set_map_config(hsw, pcm_data->stream, map, SST_HSW_CHANNEL_CONFIG_STEREO); @@ -743,7 +737,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { .capture = { .stream_name = "Analog Capture", .channels_min = 2, - .channels_max = 2, + .channels_max = 4, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, -- cgit v1.1 From 8046249d3ef18a1093fbee9ab8eb16c05c13edc7 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 16 Oct 2014 15:29:17 +0100 Subject: ASoC: Intel: Make HSW/BDW pointer debug verbose Improve the debug SNR by making the positional pointer debug more verbose. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 32a33b9..32a6470 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -552,7 +552,7 @@ static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) pos = frames_to_bytes(runtime, (runtime->control->appl_ptr % runtime->buffer_size)); - dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + dev_vdbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); /* let alsa know we have play a period */ snd_pcm_period_elapsed(substream); @@ -574,7 +574,7 @@ static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) offset = bytes_to_frames(runtime, position); ppos = sst_hsw_get_dsp_presentation_position(hsw, pcm_data->stream); - dev_dbg(rtd->dev, "PCM: DMA pointer %du bytes, pos %llu\n", + dev_vdbg(rtd->dev, "PCM: DMA pointer %du bytes, pos %llu\n", position, ppos); return offset; } -- cgit v1.1 From 3750a8f7d1df5e6442e0fc537d1d37b1b48d3712 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Fri, 17 Oct 2014 00:14:19 +0800 Subject: ASoC: Intel: mrfld: fix semicolon.cocci warnings sound/soc/intel/sst-atom-controls.c:249:2-3: Unneeded semicolon sound/soc/intel/sst-atom-controls.c:289:2-3: Unneeded semicolon Removes unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Signed-off-by: Fengguang Wu Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index dcdeb28..309a8f3 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -454,7 +454,7 @@ static int sst_gain_get(struct snd_kcontrol *kcontrol, dev_err(component->dev, "Invalid Input- gain type:%d\n", mc->type); return -EINVAL; - }; + } return 0; } @@ -494,7 +494,7 @@ static int sst_gain_put(struct snd_kcontrol *kcontrol, dev_err(cmpnt->dev, "Invalid Input- gain type:%d\n", mc->type); return -EINVAL; - }; + } if (mc->w && mc->w->power) ret = sst_send_gain_cmd(drv, gv, mc->task_id, -- cgit v1.1 From 163d2089d226ab184469f53561f1a63f151757c3 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:13 +0530 Subject: ASoC: Intel: mrfld - add the dsp sst driver The SST driver is the missing piece in our driver stack not upstreamed, so push it now :) This driver currently supports PCI device on Merrifield. Future updates will bring support for ACPI device as well as future update to PCI devices as well In subsequent patches support is added for DSP loading using memcpy, pcm operations and compressed ops. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- arch/x86/include/asm/platform_sst_audio.h | 34 ++ sound/soc/intel/sst/sst.c | 434 ++++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 539 ++++++++++++++++++++++++++++++ 3 files changed, 1007 insertions(+) create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 0a4e140..268a96ae 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -16,6 +16,9 @@ #include +#define MAX_NUM_STREAMS_MRFLD 25 +#define MAX_NUM_STREAMS MAX_NUM_STREAMS_MRFLD + enum sst_audio_task_id_mrfld { SST_TASK_ID_NONE = 0, SST_TASK_ID_SBA = 1, @@ -73,6 +76,37 @@ struct sst_platform_data { unsigned int strm_map_size; }; +struct sst_info { + u32 iram_start; + u32 iram_end; + bool iram_use; + u32 dram_start; + u32 dram_end; + bool dram_use; + u32 imr_start; + u32 imr_end; + bool imr_use; + u32 mailbox_start; + bool use_elf; + bool lpe_viewpt_rqd; + unsigned int max_streams; + u32 dma_max_len; + u8 num_probes; +}; + +struct sst_lib_dnld_info { + unsigned int mod_base; + unsigned int mod_end; + unsigned int mod_table_offset; + unsigned int mod_table_size; + bool mod_ddr_dnld; +}; + +struct sst_platform_info { + const struct sst_info *probe_data; + const struct sst_ipc_info *ipc_info; + const struct sst_lib_dnld_info *lib_info; +}; int add_sst_platform_device(void); #endif diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c new file mode 100644 index 0000000..d88cdd9 --- /dev/null +++ b/sound/soc/intel/sst/sst.c @@ -0,0 +1,434 @@ +/* + * sst.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine Driver"); +MODULE_LICENSE("GPL v2"); + +static inline bool sst_is_process_reply(u32 msg_id) +{ + return ((msg_id & PROCESS_MSG) ? true : false); +} + +static inline bool sst_validate_mailbox_size(unsigned int size) +{ + return ((size <= SST_MAILBOX_SIZE) ? true : false); +} + +static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) +{ + union interrupt_reg_mrfld isr; + union ipc_header_mrfld header; + union sst_imr_reg_mrfld imr; + struct ipc_post *msg = NULL; + unsigned int size = 0; + struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + irqreturn_t retval = IRQ_HANDLED; + + /* Interrupt arrived, check src */ + isr.full = sst_shim_read64(drv->shim, SST_ISRX); + + if (isr.part.done_interrupt) { + /* Clear done bit */ + spin_lock(&drv->ipc_spin_lock); + header.full = sst_shim_read64(drv->shim, + drv->ipc_reg.ipcx); + header.p.header_high.part.done = 0; + sst_shim_write64(drv->shim, drv->ipc_reg.ipcx, header.full); + + /* write 1 to clear status register */; + isr.part.done_interrupt = 1; + sst_shim_write64(drv->shim, SST_ISRX, isr.full); + spin_unlock(&drv->ipc_spin_lock); + + /* we can send more messages to DSP so trigger work */ + queue_work(drv->post_msg_wq, &drv->ipc_post_msg_wq); + retval = IRQ_HANDLED; + } + + if (isr.part.busy_interrupt) { + /* message from dsp so copy that */ + spin_lock(&drv->ipc_spin_lock); + imr.full = sst_shim_read64(drv->shim, SST_IMRX); + imr.part.busy_interrupt = 1; + sst_shim_write64(drv->shim, SST_IMRX, imr.full); + spin_unlock(&drv->ipc_spin_lock); + header.full = sst_shim_read64(drv->shim, drv->ipc_reg.ipcd); + + if (sst_create_ipc_msg(&msg, header.p.header_high.part.large)) { + drv->ops->clear_interrupt(drv); + return IRQ_HANDLED; + } + + if (header.p.header_high.part.large) { + size = header.p.header_low_payload; + if (sst_validate_mailbox_size(size)) { + memcpy_fromio(msg->mailbox_data, + drv->mailbox + drv->mailbox_recv_offset, size); + } else { + dev_err(drv->dev, + "Mailbox not copied, payload size is: %u\n", size); + header.p.header_low_payload = 0; + } + } + + msg->mrfld_header = header; + msg->is_process_reply = + sst_is_process_reply(header.p.header_high.part.msg_id); + spin_lock(&drv->rx_msg_lock); + list_add_tail(&msg->node, &drv->rx_list); + spin_unlock(&drv->rx_msg_lock); + drv->ops->clear_interrupt(drv); + retval = IRQ_WAKE_THREAD; + } + return retval; +} + +static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) +{ + struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + struct ipc_post *__msg, *msg = NULL; + unsigned long irq_flags; + + spin_lock_irqsave(&drv->rx_msg_lock, irq_flags); + if (list_empty(&drv->rx_list)) { + spin_unlock_irqrestore(&drv->rx_msg_lock, irq_flags); + return IRQ_HANDLED; + } + + list_for_each_entry_safe(msg, __msg, &drv->rx_list, node) { + list_del(&msg->node); + spin_unlock_irqrestore(&drv->rx_msg_lock, irq_flags); + if (msg->is_process_reply) + drv->ops->process_message(msg); + else + drv->ops->process_reply(drv, msg); + + if (msg->is_large) + kfree(msg->mailbox_data); + kfree(msg); + spin_lock_irqsave(&drv->rx_msg_lock, irq_flags); + } + spin_unlock_irqrestore(&drv->rx_msg_lock, irq_flags); + return IRQ_HANDLED; +} + +static struct intel_sst_ops mrfld_ops = { + .interrupt = intel_sst_interrupt_mrfld, + .irq_thread = intel_sst_irq_thread_mrfld, + .clear_interrupt = intel_sst_clear_intr_mrfld, + .start = sst_start_mrfld, + .reset = intel_sst_reset_dsp_mrfld, + .post_message = sst_post_message_mrfld, + .process_reply = sst_process_reply_mrfld, + .alloc_stream = sst_alloc_stream_mrfld, + .post_download = sst_post_download_mrfld, +}; + +int sst_driver_ops(struct intel_sst_drv *sst) +{ + + switch (sst->pci_id) { + case SST_MRFLD_PCI_ID: + sst->tstamp = SST_TIME_STAMP_MRFLD; + sst->ops = &mrfld_ops; + return 0; + + default: + dev_err(sst->dev, + "SST Driver capablities missing for pci_id: %x", sst->pci_id); + return -EINVAL; + }; +} + +void sst_process_pending_msg(struct work_struct *work) +{ + struct intel_sst_drv *ctx = container_of(work, + struct intel_sst_drv, ipc_post_msg_wq); + + ctx->ops->post_message(ctx, NULL, false); +} + +/* +* intel_sst_probe - PCI probe function +* +* @pci: PCI device structure +* @pci_id: PCI device ID structure +* +*/ +static int intel_sst_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int i, ret = 0; + struct intel_sst_drv *sst_drv_ctx; + struct intel_sst_ops *ops; + struct sst_platform_info *sst_pdata = pci->dev.platform_data; + int ddr_base; + + dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); + sst_drv_ctx = devm_kzalloc(&pci->dev, sizeof(*sst_drv_ctx), GFP_KERNEL); + if (!sst_drv_ctx) + return -ENOMEM; + + sst_drv_ctx->dev = &pci->dev; + sst_drv_ctx->pci_id = pci->device; + if (!sst_pdata) + return -EINVAL; + + sst_drv_ctx->pdata = sst_pdata; + if (!sst_drv_ctx->pdata->probe_data) + return -EINVAL; + + memcpy(&sst_drv_ctx->info, sst_drv_ctx->pdata->probe_data, + sizeof(sst_drv_ctx->info)); + + if (0 != sst_driver_ops(sst_drv_ctx)) + return -EINVAL; + + ops = sst_drv_ctx->ops; + mutex_init(&sst_drv_ctx->sst_lock); + + /* pvt_id 0 reserved for async messages */ + sst_drv_ctx->pvt_id = 1; + sst_drv_ctx->stream_cnt = 0; + sst_drv_ctx->fw_in_mem = NULL; + + /* we use memcpy, so set to 0 */ + sst_drv_ctx->use_dma = 0; + sst_drv_ctx->use_lli = 0; + + INIT_LIST_HEAD(&sst_drv_ctx->memcpy_list); + INIT_LIST_HEAD(&sst_drv_ctx->ipc_dispatch_list); + INIT_LIST_HEAD(&sst_drv_ctx->block_list); + INIT_LIST_HEAD(&sst_drv_ctx->rx_list); + + sst_drv_ctx->post_msg_wq = + create_singlethread_workqueue("sst_post_msg_wq"); + if (!sst_drv_ctx->post_msg_wq) { + ret = -EINVAL; + goto do_free_drv_ctx; + } + INIT_WORK(&sst_drv_ctx->ipc_post_msg_wq, sst_process_pending_msg); + init_waitqueue_head(&sst_drv_ctx->wait_queue); + + spin_lock_init(&sst_drv_ctx->ipc_spin_lock); + spin_lock_init(&sst_drv_ctx->block_lock); + spin_lock_init(&sst_drv_ctx->rx_msg_lock); + + dev_info(sst_drv_ctx->dev, "Got drv data max stream %d\n", + sst_drv_ctx->info.max_streams); + for (i = 1; i <= sst_drv_ctx->info.max_streams; i++) { + struct stream_info *stream = &sst_drv_ctx->streams[i]; + + memset(stream, 0, sizeof(*stream)); + stream->pipe_id = PIPE_RSVD; + mutex_init(&stream->lock); + } + + /* Init the device */ + ret = pcim_enable_device(pci); + if (ret) { + dev_err(sst_drv_ctx->dev, + "device can't be enabled. Returned err: %d\n", ret); + goto do_free_mem; + } + sst_drv_ctx->pci = pci_dev_get(pci); + ret = pci_request_regions(pci, SST_DRV_NAME); + if (ret) + goto do_free_mem; + + /* map registers */ + /* DDR base */ + if (sst_drv_ctx->pci_id == SST_MRFLD_PCI_ID) { + sst_drv_ctx->ddr_base = pci_resource_start(pci, 0); + /* check that the relocated IMR base matches with FW Binary */ + ddr_base = relocate_imr_addr_mrfld(sst_drv_ctx->ddr_base); + if (!sst_drv_ctx->pdata->lib_info) { + dev_err(sst_drv_ctx->dev, "lib_info pointer NULL\n"); + ret = -EINVAL; + goto do_release_regions; + } + if (ddr_base != sst_drv_ctx->pdata->lib_info->mod_base) { + dev_err(sst_drv_ctx->dev, + "FW LSP DDR BASE does not match with IFWI\n"); + ret = -EINVAL; + goto do_release_regions; + } + sst_drv_ctx->ddr_end = pci_resource_end(pci, 0); + + sst_drv_ctx->ddr = pcim_iomap(pci, 0, + pci_resource_len(pci, 0)); + if (!sst_drv_ctx->ddr) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(sst_drv_ctx->dev, "sst: DDR Ptr %p\n", sst_drv_ctx->ddr); + } else { + sst_drv_ctx->ddr = NULL; + } + + /* SHIM */ + sst_drv_ctx->shim_phy_add = pci_resource_start(pci, 1); + sst_drv_ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); + if (!sst_drv_ctx->shim) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(sst_drv_ctx->dev, "SST Shim Ptr %p\n", sst_drv_ctx->shim); + + /* Shared SRAM */ + sst_drv_ctx->mailbox_add = pci_resource_start(pci, 2); + sst_drv_ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); + if (!sst_drv_ctx->mailbox) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(sst_drv_ctx->dev, "SRAM Ptr %p\n", sst_drv_ctx->mailbox); + + /* IRAM */ + sst_drv_ctx->iram_end = pci_resource_end(pci, 3); + sst_drv_ctx->iram_base = pci_resource_start(pci, 3); + sst_drv_ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); + if (!sst_drv_ctx->iram) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(sst_drv_ctx->dev, "IRAM Ptr %p\n", sst_drv_ctx->iram); + + /* DRAM */ + sst_drv_ctx->dram_end = pci_resource_end(pci, 4); + sst_drv_ctx->dram_base = pci_resource_start(pci, 4); + sst_drv_ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); + if (!sst_drv_ctx->dram) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(sst_drv_ctx->dev, "DRAM Ptr %p\n", sst_drv_ctx->dram); + + + sst_set_fw_state_locked(sst_drv_ctx, SST_RESET); + sst_drv_ctx->irq_num = pci->irq; + /* Register the ISR */ + ret = devm_request_threaded_irq(&pci->dev, pci->irq, + sst_drv_ctx->ops->interrupt, + sst_drv_ctx->ops->irq_thread, 0, SST_DRV_NAME, + sst_drv_ctx); + if (ret) + goto do_release_regions; + dev_dbg(sst_drv_ctx->dev, "Registered IRQ 0x%x\n", pci->irq); + + /* default intr are unmasked so set this as masked */ + if (sst_drv_ctx->pci_id == SST_MRFLD_PCI_ID) + sst_shim_write64(sst_drv_ctx->shim, SST_IMRX, 0xFFFF0038); + + pci_set_drvdata(pci, sst_drv_ctx); + pm_runtime_set_autosuspend_delay(sst_drv_ctx->dev, SST_SUSPEND_DELAY); + pm_runtime_use_autosuspend(sst_drv_ctx->dev); + pm_runtime_allow(sst_drv_ctx->dev); + pm_runtime_put_noidle(sst_drv_ctx->dev); + sst_register(sst_drv_ctx->dev); + sst_drv_ctx->qos = devm_kzalloc(&pci->dev, + sizeof(struct pm_qos_request), GFP_KERNEL); + if (!sst_drv_ctx->qos) { + ret = -ENOMEM; + goto do_release_regions; + } + pm_qos_add_request(sst_drv_ctx->qos, PM_QOS_CPU_DMA_LATENCY, + PM_QOS_DEFAULT_VALUE); + + return ret; + +do_release_regions: + pci_release_regions(pci); +do_free_mem: + destroy_workqueue(sst_drv_ctx->post_msg_wq); +do_free_drv_ctx: + dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); + return ret; +} + +/** +* intel_sst_remove - PCI remove function +* +* @pci: PCI device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +static void intel_sst_remove(struct pci_dev *pci) +{ + struct intel_sst_drv *sst_drv_ctx = pci_get_drvdata(pci); + + pm_runtime_get_noresume(sst_drv_ctx->dev); + pm_runtime_forbid(sst_drv_ctx->dev); + sst_unregister(sst_drv_ctx->dev); + pci_dev_put(sst_drv_ctx->pci); + sst_set_fw_state_locked(sst_drv_ctx, SST_SHUTDOWN); + + flush_scheduled_work(); + destroy_workqueue(sst_drv_ctx->post_msg_wq); + pm_qos_remove_request(sst_drv_ctx->qos); + kfree(sst_drv_ctx->fw_sg_list.src); + kfree(sst_drv_ctx->fw_sg_list.dst); + sst_drv_ctx->fw_sg_list.list_len = 0; + kfree(sst_drv_ctx->fw_in_mem); + sst_drv_ctx->fw_in_mem = NULL; + sst_memcpy_free_resources(sst_drv_ctx); + sst_drv_ctx = NULL; + pci_release_regions(pci); + pci_set_drvdata(pci, NULL); +} + +/* PCI Routines */ +static struct pci_device_id intel_sst_ids[] = { + { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, + { 0, } +}; + +static struct pci_driver sst_driver = { + .name = SST_DRV_NAME, + .id_table = intel_sst_ids, + .probe = intel_sst_probe, + .remove = intel_sst_remove, +}; + +module_pci_driver(sst_driver); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h new file mode 100644 index 0000000..bfcf51a --- /dev/null +++ b/sound/soc/intel/sst/sst.h @@ -0,0 +1,539 @@ +/* + * sst.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corporation + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * Common private declarations for SST + */ +#ifndef __SST_H__ +#define __SST_H__ + +#include + +/* driver names */ +#define SST_DRV_NAME "intel_sst_driver" +#define SST_MRFLD_PCI_ID 0x119A + +#define SST_SUSPEND_DELAY 2000 +#define FW_CONTEXT_MEM (64*1024) +#define SST_ICCM_BOUNDARY 4 +#define SST_CONFIG_SSP_SIGN 0x7ffe8001 + +#define MRFLD_FW_VIRTUAL_BASE 0xC0000000 +#define MRFLD_FW_DDR_BASE_OFFSET 0x0 +#define MRFLD_FW_FEATURE_BASE_OFFSET 0x4 +#define MRFLD_FW_BSS_RESET_BIT 0 + +enum sst_states { + SST_FW_LOADING = 1, + SST_FW_RUNNING, + SST_RESET, + SST_SHUTDOWN, +}; + +enum sst_algo_ops { + SST_SET_ALGO = 0, + SST_GET_ALGO = 1, +}; + +#define SST_BLOCK_TIMEOUT 1000 + +#define FW_SIGNATURE_SIZE 4 + +/* stream states */ +enum sst_stream_states { + STREAM_UN_INIT = 0, /* Freed/Not used stream */ + STREAM_RUNNING = 1, /* Running */ + STREAM_PAUSED = 2, /* Paused stream */ + STREAM_DECODE = 3, /* stream is in decoding only state */ + STREAM_INIT = 4, /* stream init, waiting for data */ + STREAM_RESET = 5, /* force reset on recovery */ +}; + +enum sst_ram_type { + SST_IRAM = 1, + SST_DRAM = 2, + SST_DDR = 5, + SST_CUSTOM_INFO = 7, /* consists of FW binary information */ +}; + +/* SST shim registers to structure mapping */ +union interrupt_reg { + struct { + u64 done_interrupt:1; + u64 busy_interrupt:1; + u64 rsvd:62; + } part; + u64 full; +}; + +union sst_pisr_reg { + struct { + u32 pssp0:1; + u32 pssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:26; + } part; + u32 full; +}; + +union sst_pimr_reg { + struct { + u32 ssp0:1; + u32 ssp1:1; + u32 rsvd0:3; + u32 dmac:1; + u32 rsvd1:10; + u32 ssp0_sc:1; + u32 ssp1_sc:1; + u32 rsvd2:3; + u32 dmac_sc:1; + u32 rsvd3:10; + } part; + u32 full; +}; + +union config_status_reg_mrfld { + struct { + u64 lpe_reset:1; + u64 lpe_reset_vector:1; + u64 runstall:1; + u64 pwaitmode:1; + u64 clk_sel:3; + u64 rsvd2:1; + u64 sst_clk:3; + u64 xt_snoop:1; + u64 rsvd3:4; + u64 clk_sel1:6; + u64 clk_enable:3; + u64 rsvd4:6; + u64 slim0baseclk:1; + u64 rsvd:32; + } part; + u64 full; +}; + +union interrupt_reg_mrfld { + struct { + u64 done_interrupt:1; + u64 busy_interrupt:1; + u64 rsvd:62; + } part; + u64 full; +}; + +union sst_imr_reg_mrfld { + struct { + u64 done_interrupt:1; + u64 busy_interrupt:1; + u64 rsvd:62; + } part; + u64 full; +}; + +/** + * struct sst_block - This structure is used to block a user/fw data call to another + * fw/user call + * + * @condition: condition for blocking check + * @ret_code: ret code when block is released + * @data: data ptr + * @size: size of data + * @on: block condition + * @msg_id: msg_id = msgid in mfld/ctp, mrfld = NULL + * @drv_id: str_id in mfld/ctp, = drv_id in mrfld + * @node: list head node + */ +struct sst_block { + bool condition; + int ret_code; + void *data; + u32 size; + bool on; + u32 msg_id; + u32 drv_id; + struct list_head node; +}; + +/** + * struct stream_info - structure that holds the stream information + * + * @status : stream current state + * @prev : stream prev state + * @ops : stream operation pb/cp/drm... + * @bufs: stream buffer list + * @lock : stream mutex for protecting state + * @pcm_substream : PCM substream + * @period_elapsed : PCM period elapsed callback + * @sfreq : stream sampling freq + * @str_type : stream type + * @cumm_bytes : cummulative bytes decoded + * @str_type : stream type + * @src : stream source + */ +struct stream_info { + unsigned int status; + unsigned int prev; + unsigned int ops; + struct mutex lock; + + void *pcm_substream; + void (*period_elapsed)(void *pcm_substream); + + unsigned int sfreq; + u32 cumm_bytes; + + void *compr_cb_param; + void (*compr_cb)(void *compr_cb_param); + + void *drain_cb_param; + void (*drain_notify)(void *drain_cb_param); + + unsigned int num_ch; + unsigned int pipe_id; + unsigned int str_id; + unsigned int task_id; +}; + +#define SST_FW_SIGN "$SST" +#define SST_FW_LIB_SIGN "$LIB" + +/** + * struct sst_fw_header - FW file headers + * + * @signature : FW signature + * @file_size: size of fw image + * @modules : # of modules + * @file_format : version of header format + * @reserved : reserved fields + */ +struct sst_fw_header { + unsigned char signature[FW_SIGNATURE_SIZE]; + u32 file_size; + u32 modules; + u32 file_format; + u32 reserved[4]; +}; + +/** + * struct fw_module_header - module header in FW + * + * @signature: module signature + * @mod_size: size of module + * @blocks: block count + * @type: block type + * @entry_point: module netry point + */ +struct fw_module_header { + unsigned char signature[FW_SIGNATURE_SIZE]; + u32 mod_size; + u32 blocks; + u32 type; + u32 entry_point; +}; + +/** + * struct fw_block_info - block header for FW + * + * @type: block ram type I/D + * @size: size of block + * @ram_offset: offset in ram + */ +struct fw_block_info { + enum sst_ram_type type; + u32 size; + u32 ram_offset; + u32 rsvd; +}; + +struct sst_runtime_param { + struct snd_sst_runtime_params param; +}; + +struct sst_sg_list { + struct scatterlist *src; + struct scatterlist *dst; + int list_len; + unsigned int sg_idx; +}; + +struct sst_memcpy_list { + struct list_head memcpylist; + void *dstn; + const void *src; + u32 size; + bool is_io; +}; + +/*Firmware Module Information*/ +enum sst_lib_dwnld_status { + SST_LIB_NOT_FOUND = 0, + SST_LIB_FOUND, + SST_LIB_DOWNLOADED, +}; + +struct sst_module_info { + const char *name; /*Library name*/ + u32 id; /*Module ID*/ + u32 entry_pt; /*Module entry point*/ + u8 status; /*module status*/ + u8 rsvd1; + u16 rsvd2; +}; + +/* + * Structure for managing the Library Region(1.5MB) + * in DDR in Merrifield + */ +struct sst_mem_mgr { + phys_addr_t current_base; + int avail; + unsigned int count; +}; + +struct sst_ipc_reg { + int ipcx; + int ipcd; +}; + +struct sst_shim_regs64 { + u64 csr; + u64 pisr; + u64 pimr; + u64 isrx; + u64 isrd; + u64 imrx; + u64 imrd; + u64 ipcx; + u64 ipcd; + u64 isrsc; + u64 isrlpesc; + u64 imrsc; + u64 imrlpesc; + u64 ipcsc; + u64 ipclpesc; + u64 clkctl; + u64 csr2; +}; + +/** + * struct intel_sst_drv - driver ops + * + * @sst_state : current sst device state + * @pci_id : PCI device id loaded + * @shim : SST shim pointer + * @mailbox : SST mailbox pointer + * @iram : SST IRAM pointer + * @dram : SST DRAM pointer + * @pdata : SST info passed as a part of pci platform data + * @shim_phy_add : SST shim phy addr + * @shim_regs64: Struct to save shim registers + * @ipc_dispatch_list : ipc messages dispatched + * @rx_list : to copy the process_reply/process_msg from DSP + * @ipc_post_msg_wq : wq to post IPC messages context + * @mad_ops : MAD driver operations registered + * @mad_wq : MAD driver wq + * @post_msg_wq : wq to post IPC messages + * @streams : sst stream contexts + * @list_lock : sst driver list lock (deprecated) + * @ipc_spin_lock : spin lock to handle audio shim access and ipc queue + * @block_lock : spin lock to add block to block_list and assign pvt_id + * @rx_msg_lock : spin lock to handle the rx messages from the DSP + * @scard_ops : sst card ops + * @pci : sst pci device struture + * @dev : pointer to current device struct + * @sst_lock : sst device lock + * @pvt_id : sst private id + * @stream_cnt : total sst active stream count + * @pb_streams : total active pb streams + * @cp_streams : total active cp streams + * @audio_start : audio status + * @qos : PM Qos struct + * firmware_name : Firmware / Library name + */ +struct intel_sst_drv { + int sst_state; + int irq_num; + unsigned int pci_id; + void __iomem *ddr; + void __iomem *shim; + void __iomem *mailbox; + void __iomem *iram; + void __iomem *dram; + unsigned int mailbox_add; + unsigned int iram_base; + unsigned int dram_base; + unsigned int shim_phy_add; + unsigned int iram_end; + unsigned int dram_end; + unsigned int ddr_end; + unsigned int ddr_base; + unsigned int mailbox_recv_offset; + struct sst_shim_regs64 *shim_regs64; + struct list_head block_list; + struct list_head ipc_dispatch_list; + struct sst_platform_info *pdata; + struct list_head rx_list; + struct work_struct ipc_post_msg_wq; + wait_queue_head_t wait_queue; + struct workqueue_struct *post_msg_wq; + unsigned int tstamp; + /* str_id 0 is not used */ + struct stream_info streams[MAX_NUM_STREAMS+1]; + spinlock_t ipc_spin_lock; + spinlock_t block_lock; + spinlock_t rx_msg_lock; + struct pci_dev *pci; + struct device *dev; + volatile long unsigned pvt_id; + struct mutex sst_lock; + unsigned int stream_cnt; + unsigned int csr_value; + void *fw_in_mem; + struct sst_sg_list fw_sg_list, library_list; + struct intel_sst_ops *ops; + struct sst_info info; + struct pm_qos_request *qos; + unsigned int use_dma; + unsigned int use_lli; + atomic_t fw_clear_context; + bool lib_dwnld_reqd; + struct list_head memcpy_list; + struct sst_ipc_reg ipc_reg; + struct sst_mem_mgr lib_mem_mgr; + /* + * Holder for firmware name. Due to async call it needs to be + * persistent till worker thread gets called + */ + char firmware_name[20]; +}; + +/* misc definitions */ +#define FW_DWNL_ID 0x01 + +struct intel_sst_ops { + irqreturn_t (*interrupt)(int, void *); + irqreturn_t (*irq_thread)(int, void *); + void (*clear_interrupt)(struct intel_sst_drv *ctx); + int (*start)(struct intel_sst_drv *ctx); + int (*reset)(struct intel_sst_drv *ctx); + void (*process_reply)(struct intel_sst_drv *ctx, struct ipc_post *msg); + int (*post_message)(struct intel_sst_drv *ctx, + struct ipc_post *msg, bool sync); + void (*process_message)(struct ipc_post *msg); + void (*set_bypass)(bool set); + int (*save_dsp_context)(struct intel_sst_drv *sst); + void (*restore_dsp_context)(void); + int (*alloc_stream)(struct intel_sst_drv *ctx, void *params); + void (*post_download)(struct intel_sst_drv *sst); +}; + +int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int id); +int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int id); +int sst_drop_stream(struct intel_sst_drv *sst_drv_ctx, int id); +int sst_free_stream(struct intel_sst_drv *sst_drv_ctx, int id); +int sst_start_stream(struct intel_sst_drv *sst_drv_ctx, int str_id); +int sst_send_byte_stream_mrfld(struct intel_sst_drv *ctx, + struct snd_sst_bytes_v2 *sbytes); +int sst_set_stream_param(int str_id, struct snd_sst_params *str_param); +int sst_set_metadata(int str_id, char *params); +int sst_get_stream(struct intel_sst_drv *sst_drv_ctx, + struct snd_sst_params *str_param); +int sst_get_stream_allocated(struct intel_sst_drv *ctx, + struct snd_sst_params *str_param, + struct snd_sst_lib_download **lib_dnld); +int sst_drain_stream(struct intel_sst_drv *sst_drv_ctx, + int str_id, bool partial_drain); +int sst_post_message_mrfld(struct intel_sst_drv *ctx, + struct ipc_post *msg, bool sync); +void sst_process_reply_mrfld(struct intel_sst_drv *ctx, struct ipc_post *msg); +int sst_start_mrfld(struct intel_sst_drv *ctx); +int intel_sst_reset_dsp_mrfld(struct intel_sst_drv *ctx); +void intel_sst_clear_intr_mrfld(struct intel_sst_drv *ctx); + +int sst_load_fw(struct intel_sst_drv *ctx); +int sst_load_library(struct snd_sst_lib_download *lib, u8 ops); +void sst_post_download_mrfld(struct intel_sst_drv *ctx); +int sst_get_block_stream(struct intel_sst_drv *sst_drv_ctx); +void sst_memcpy_free_resources(struct intel_sst_drv *ctx); + +int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, + struct sst_block *block); +int sst_wait_timeout(struct intel_sst_drv *sst_drv_ctx, + struct sst_block *block); +int sst_create_ipc_msg(struct ipc_post **arg, bool large); +int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id); +void sst_clean_stream(struct stream_info *stream); +int intel_sst_register_compress(struct intel_sst_drv *sst); +int intel_sst_remove_compress(struct intel_sst_drv *sst); +void sst_cdev_fragment_elapsed(struct intel_sst_drv *ctx, int str_id); +int sst_send_sync_msg(int ipc, int str_id); +int sst_get_num_channel(struct snd_sst_params *str_param); +int sst_get_sfreq(struct snd_sst_params *str_param); +int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params); +void sst_restore_fw_context(void); +struct sst_block *sst_create_block(struct intel_sst_drv *ctx, + u32 msg_id, u32 drv_id); +int sst_create_block_and_ipc_msg(struct ipc_post **arg, bool large, + struct intel_sst_drv *sst_drv_ctx, struct sst_block **block, + u32 msg_id, u32 drv_id); +int sst_free_block(struct intel_sst_drv *ctx, struct sst_block *freed); +int sst_wake_up_block(struct intel_sst_drv *ctx, int result, + u32 drv_id, u32 ipc, void *data, u32 size); +int sst_request_firmware_async(struct intel_sst_drv *ctx); +int sst_driver_ops(struct intel_sst_drv *sst); +struct sst_platform_info *sst_get_acpi_driver_data(const char *hid); +void sst_firmware_load_cb(const struct firmware *fw, void *context); +int sst_prepare_and_post_msg(struct intel_sst_drv *sst, + int task_id, int ipc_msg, int cmd_id, int pipe_id, + size_t mbox_data_len, const void *mbox_data, void **data, + bool large, bool fill_dsp, bool sync, bool response); + +void sst_save_shim64(struct intel_sst_drv *ctx, void __iomem *shim, + struct sst_shim_regs64 *shim_regs); +void sst_process_pending_msg(struct work_struct *work); +int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); +void sst_init_stream(struct stream_info *stream, + int codec, int sst_id, int ops, u8 slot); +int sst_validate_strid(struct intel_sst_drv *sst_drv_ctx, int str_id); +struct stream_info *get_stream_info(struct intel_sst_drv *sst_drv_ctx, + int str_id); +int get_stream_id_mrfld(struct intel_sst_drv *sst_drv_ctx, + u32 pipe_id); +u32 relocate_imr_addr_mrfld(u32 base_addr); +void sst_add_to_dispatch_list_and_post(struct intel_sst_drv *sst, + struct ipc_post *msg); +int sst_pm_runtime_put(struct intel_sst_drv *sst_drv); +int sst_shim_write(void __iomem *addr, int offset, int value); +u32 sst_shim_read(void __iomem *addr, int offset); +u64 sst_reg_read64(void __iomem *addr, int offset); +int sst_shim_write64(void __iomem *addr, int offset, u64 value); +u64 sst_shim_read64(void __iomem *addr, int offset); +void sst_set_fw_state_locked( + struct intel_sst_drv *sst_drv_ctx, int sst_state); +void sst_fill_header_mrfld(union ipc_header_mrfld *header, + int msg, int task_id, int large, int drv_id); +void sst_fill_header_dsp(struct ipc_dsp_hdr *dsp, int msg, + int pipe_id, int len); + +int sst_register(struct device *); +int sst_unregister(struct device *); + +#endif -- cgit v1.1 From 9012c9544eeac485b2193fea721233907f0847fa Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:14 +0530 Subject: ASoC: Intel: mrfld - Add DSP load and management This patch contains all dsp controlling functions like firmware download, setting/resetting dsp cores, etc. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_loader.c | 461 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 461 insertions(+) create mode 100644 sound/soc/intel/sst/sst_loader.c diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c new file mode 100644 index 0000000..b6d27c1 --- /dev/null +++ b/sound/soc/intel/sst/sst_loader.c @@ -0,0 +1,461 @@ +/* + * sst_dsp.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This file contains all dsp controlling functions like firmware download, + * setting/resetting dsp cores, etc + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + +static void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + int i; + const u32 *src_32 = src; + u32 *dst_32 = dst; + + for (i = 0; i < count/sizeof(u32); i++) + writel(*src_32++, dst_32++); +} + +/** + * intel_sst_reset_dsp_mrfld - Resetting SST DSP + * + * This resets DSP in case of MRFLD platfroms + */ +int intel_sst_reset_dsp_mrfld(struct intel_sst_drv *sst_drv_ctx) +{ + union config_status_reg_mrfld csr; + + dev_dbg(sst_drv_ctx->dev, "sst: Resetting the DSP in mrfld\n"); + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + + dev_dbg(sst_drv_ctx->dev, "value:0x%llx\n", csr.full); + + csr.full |= 0x7; + sst_shim_write64(sst_drv_ctx->shim, SST_CSR, csr.full); + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + + dev_dbg(sst_drv_ctx->dev, "value:0x%llx\n", csr.full); + + csr.full &= ~(0x1); + sst_shim_write64(sst_drv_ctx->shim, SST_CSR, csr.full); + + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + dev_dbg(sst_drv_ctx->dev, "value:0x%llx\n", csr.full); + return 0; +} + +/** + * sst_start_merrifield - Start the SST DSP processor + * + * This starts the DSP in MERRIFIELD platfroms + */ +int sst_start_mrfld(struct intel_sst_drv *sst_drv_ctx) +{ + union config_status_reg_mrfld csr; + + dev_dbg(sst_drv_ctx->dev, "sst: Starting the DSP in mrfld LALALALA\n"); + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + dev_dbg(sst_drv_ctx->dev, "value:0x%llx\n", csr.full); + + csr.full |= 0x7; + sst_shim_write64(sst_drv_ctx->shim, SST_CSR, csr.full); + + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + dev_dbg(sst_drv_ctx->dev, "value:0x%llx\n", csr.full); + + csr.part.xt_snoop = 1; + csr.full &= ~(0x5); + sst_shim_write64(sst_drv_ctx->shim, SST_CSR, csr.full); + + csr.full = sst_shim_read64(sst_drv_ctx->shim, SST_CSR); + dev_dbg(sst_drv_ctx->dev, "sst: Starting the DSP_merrifield:%llx\n", + csr.full); + return 0; +} + +static int sst_validate_fw_image(struct intel_sst_drv *ctx, unsigned long size, + struct fw_module_header **module, u32 *num_modules) +{ + struct sst_fw_header *header; + const void *sst_fw_in_mem = ctx->fw_in_mem; + + dev_dbg(ctx->dev, "Enter\n"); + + /* Read the header information from the data pointer */ + header = (struct sst_fw_header *)sst_fw_in_mem; + dev_dbg(ctx->dev, + "header sign=%s size=%x modules=%x fmt=%x size=%zx\n", + header->signature, header->file_size, header->modules, + header->file_format, sizeof(*header)); + + /* verify FW */ + if ((strncmp(header->signature, SST_FW_SIGN, 4) != 0) || + (size != header->file_size + sizeof(*header))) { + /* Invalid FW signature */ + dev_err(ctx->dev, "InvalidFW sign/filesize mismatch\n"); + return -EINVAL; + } + *num_modules = header->modules; + *module = (void *)sst_fw_in_mem + sizeof(*header); + + return 0; +} + +/* + * sst_fill_memcpy_list - Fill the memcpy list + * + * @memcpy_list: List to be filled + * @destn: Destination addr to be filled in the list + * @src: Source addr to be filled in the list + * @size: Size to be filled in the list + * + * Adds the node to the list after required fields + * are populated in the node + */ +static int sst_fill_memcpy_list(struct list_head *memcpy_list, + void *destn, const void *src, u32 size, bool is_io) +{ + struct sst_memcpy_list *listnode; + + listnode = kzalloc(sizeof(*listnode), GFP_KERNEL); + if (listnode == NULL) + return -ENOMEM; + listnode->dstn = destn; + listnode->src = src; + listnode->size = size; + listnode->is_io = is_io; + list_add_tail(&listnode->memcpylist, memcpy_list); + + return 0; +} + +/** + * sst_parse_module_memcpy - Parse audio FW modules and populate the memcpy list + * + * @sst_drv_ctx : driver context + * @module : FW module header + * @memcpy_list : Pointer to the list to be populated + * Create the memcpy list as the number of block to be copied + * returns error or 0 if module sizes are proper + */ +static int sst_parse_module_memcpy(struct intel_sst_drv *sst_drv_ctx, + struct fw_module_header *module, struct list_head *memcpy_list) +{ + struct fw_block_info *block; + u32 count; + int ret_val = 0; + void __iomem *ram_iomem; + + dev_dbg(sst_drv_ctx->dev, "module sign %s size %x blocks %x type %x\n", + module->signature, module->mod_size, + module->blocks, module->type); + dev_dbg(sst_drv_ctx->dev, "module entrypoint 0x%x\n", module->entry_point); + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + if (block->size <= 0) { + dev_err(sst_drv_ctx->dev, "block size invalid\n"); + return -EINVAL; + } + switch (block->type) { + case SST_IRAM: + ram_iomem = sst_drv_ctx->iram; + break; + case SST_DRAM: + ram_iomem = sst_drv_ctx->dram; + break; + case SST_DDR: + ram_iomem = sst_drv_ctx->ddr; + break; + case SST_CUSTOM_INFO: + block = (void *)block + sizeof(*block) + block->size; + continue; + default: + dev_err(sst_drv_ctx->dev, "wrong ram type0x%x in block0x%x\n", + block->type, count); + return -EINVAL; + } + + ret_val = sst_fill_memcpy_list(memcpy_list, + ram_iomem + block->ram_offset, + (void *)block + sizeof(*block), block->size, 1); + if (ret_val) + return ret_val; + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +/** + * sst_parse_fw_memcpy - parse the firmware image & populate the list for memcpy + * + * @ctx : pointer to drv context + * @size : size of the firmware + * @fw_list : pointer to list_head to be populated + * This function parses the FW image and saves the parsed image in the list + * for memcpy + */ +static int sst_parse_fw_memcpy(struct intel_sst_drv *ctx, unsigned long size, + struct list_head *fw_list) +{ + struct fw_module_header *module; + u32 count, num_modules; + int ret_val; + + ret_val = sst_validate_fw_image(ctx, size, &module, &num_modules); + if (ret_val) + return ret_val; + + for (count = 0; count < num_modules; count++) { + ret_val = sst_parse_module_memcpy(ctx, module, fw_list); + if (ret_val) + return ret_val; + module = (void *)module + sizeof(*module) + module->mod_size; + } + + return 0; +} + +/** + * sst_do_memcpy - function initiates the memcpy + * + * @memcpy_list: Pter to memcpy list on which the memcpy needs to be initiated + * + * Triggers the memcpy + */ +static void sst_do_memcpy(struct list_head *memcpy_list) +{ + struct sst_memcpy_list *listnode; + + list_for_each_entry(listnode, memcpy_list, memcpylist) { + if (listnode->is_io == true) + memcpy32_toio((void __iomem *)listnode->dstn, + listnode->src, listnode->size); + else + memcpy(listnode->dstn, listnode->src, listnode->size); + } +} + +void sst_memcpy_free_resources(struct intel_sst_drv *sst_drv_ctx) +{ + struct sst_memcpy_list *listnode, *tmplistnode; + + /* Free the list */ + if (!list_empty(&sst_drv_ctx->memcpy_list)) { + list_for_each_entry_safe(listnode, tmplistnode, + &sst_drv_ctx->memcpy_list, memcpylist) { + list_del(&listnode->memcpylist); + kfree(listnode); + } + } +} + +static int sst_cache_and_parse_fw(struct intel_sst_drv *sst, + const struct firmware *fw) +{ + int retval = 0; + + sst->fw_in_mem = kzalloc(fw->size, GFP_KERNEL); + if (!sst->fw_in_mem) { + retval = -ENOMEM; + goto end_release; + } + dev_dbg(sst->dev, "copied fw to %p", sst->fw_in_mem); + dev_dbg(sst->dev, "phys: %lx", (unsigned long)virt_to_phys(sst->fw_in_mem)); + memcpy(sst->fw_in_mem, fw->data, fw->size); + retval = sst_parse_fw_memcpy(sst, fw->size, &sst->memcpy_list); + if (retval) { + dev_err(sst->dev, "Failed to parse fw\n"); + kfree(sst->fw_in_mem); + sst->fw_in_mem = NULL; + } + +end_release: + release_firmware(fw); + return retval; + +} + +void sst_firmware_load_cb(const struct firmware *fw, void *context) +{ + struct intel_sst_drv *ctx = context; + + dev_dbg(ctx->dev, "Enter\n"); + + if (fw == NULL) { + dev_err(ctx->dev, "request fw failed\n"); + return; + } + + mutex_lock(&ctx->sst_lock); + + if (ctx->sst_state != SST_RESET || + ctx->fw_in_mem != NULL) { + if (fw != NULL) + release_firmware(fw); + mutex_unlock(&ctx->sst_lock); + return; + } + + dev_dbg(ctx->dev, "Request Fw completed\n"); + sst_cache_and_parse_fw(ctx, fw); + mutex_unlock(&ctx->sst_lock); +} + +/* + * sst_request_fw - requests audio fw from kernel and saves a copy + * + * This function requests the SST FW from the kernel, parses it and + * saves a copy in the driver context + */ +static int sst_request_fw(struct intel_sst_drv *sst) +{ + int retval = 0; + char name[20]; + const struct firmware *fw; + + dev_dbg(sst->dev, "Requesting FW %s now...\n", name); + + retval = request_firmware(&fw, name, sst->dev); + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } + if (retval) { + dev_err(sst->dev, "request fw failed %d\n", retval); + return retval; + } + mutex_lock(&sst->sst_lock); + retval = sst_cache_and_parse_fw(sst, fw); + mutex_unlock(&sst->sst_lock); + + return retval; +} + +/* + * Writing the DDR physical base to DCCM offset + * so that FW can use it to setup TLB + */ +static void sst_dccm_config_write(void __iomem *dram_base, + unsigned int ddr_base) +{ + void __iomem *addr; + u32 bss_reset = 0; + + addr = (void __iomem *)(dram_base + MRFLD_FW_DDR_BASE_OFFSET); + memcpy32_toio(addr, (void *)&ddr_base, sizeof(u32)); + bss_reset |= (1 << MRFLD_FW_BSS_RESET_BIT); + addr = (void __iomem *)(dram_base + MRFLD_FW_FEATURE_BASE_OFFSET); + memcpy32_toio(addr, &bss_reset, sizeof(u32)); + +} + +void sst_post_download_mrfld(struct intel_sst_drv *ctx) +{ + sst_dccm_config_write(ctx->dram, ctx->ddr_base); + dev_dbg(ctx->dev, "config written to DCCM\n"); +} + +/** + * sst_load_fw - function to load FW into DSP + * Transfers the FW to DSP using dma/memcpy + */ +int sst_load_fw(struct intel_sst_drv *sst_drv_ctx) +{ + int ret_val = 0; + struct sst_block *block; + + dev_dbg(sst_drv_ctx->dev, "sst_load_fw\n"); + + if (sst_drv_ctx->sst_state != SST_RESET || + sst_drv_ctx->sst_state == SST_SHUTDOWN) + return -EAGAIN; + + if (!sst_drv_ctx->fw_in_mem) { + dev_dbg(sst_drv_ctx->dev, "sst: FW not in memory retry to download\n"); + ret_val = sst_request_fw(sst_drv_ctx); + if (ret_val) + return ret_val; + } + + BUG_ON(!sst_drv_ctx->fw_in_mem); + block = sst_create_block(sst_drv_ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + /* Prevent C-states beyond C6 */ + pm_qos_update_request(sst_drv_ctx->qos, 0); + + sst_drv_ctx->sst_state = SST_FW_LOADING; + + ret_val = sst_drv_ctx->ops->reset(sst_drv_ctx); + if (ret_val) + goto restore; + + sst_do_memcpy(&sst_drv_ctx->memcpy_list); + + /* Write the DRAM/DCCM config before enabling FW */ + if (sst_drv_ctx->ops->post_download) + sst_drv_ctx->ops->post_download(sst_drv_ctx); + + /* bring sst out of reset */ + ret_val = sst_drv_ctx->ops->start(sst_drv_ctx); + if (ret_val) + goto restore; + + ret_val = sst_wait_timeout(sst_drv_ctx, block); + if (ret_val) { + dev_err(sst_drv_ctx->dev, "fw download failed %d\n" , ret_val); + /* FW download failed due to timeout */ + ret_val = -EBUSY; + + } + + +restore: + /* Re-enable Deeper C-states beyond C6 */ + pm_qos_update_request(sst_drv_ctx->qos, PM_QOS_DEFAULT_VALUE); + sst_free_block(sst_drv_ctx, block); + dev_dbg(sst_drv_ctx->dev, "fw load successful!!!\n"); + + if (sst_drv_ctx->ops->restore_dsp_context) + sst_drv_ctx->ops->restore_dsp_context(); + sst_drv_ctx->sst_state = SST_FW_RUNNING; + return ret_val; +} + -- cgit v1.1 From cc547054d3122646cb9be5d4fc80699bf3e281d8 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:15 +0530 Subject: ASoC: Intel: sst - add pcm ops handling This patch adds low level IPC handling for pcm stream operations Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 403 ++++++++++++++++++++++++++++++++ 1 file changed, 403 insertions(+) create mode 100644 sound/soc/intel/sst/sst_drv_interface.c diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c new file mode 100644 index 0000000..aadb0db --- /dev/null +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -0,0 +1,403 @@ +/* + * sst_drv_interface.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + + + +#define NUM_CODEC 2 +#define MIN_FRAGMENT 2 +#define MAX_FRAGMENT 4 +#define MIN_FRAGMENT_SIZE (50 * 1024) +#define MAX_FRAGMENT_SIZE (1024 * 1024) +#define SST_GET_BYTES_PER_SAMPLE(pcm_wd_sz) (((pcm_wd_sz + 15) >> 4) << 1) + +int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) +{ + struct stream_info *stream; + int ret = 0; + + stream = get_stream_info(ctx, str_id); + if (stream) { + /* str_id is valid, so stream is alloacted */ + ret = sst_free_stream(ctx, str_id); + if (ret) + sst_clean_stream(&ctx->streams[str_id]); + return ret; + } + return ret; +} + +int sst_get_stream_allocated(struct intel_sst_drv *ctx, + struct snd_sst_params *str_param, + struct snd_sst_lib_download **lib_dnld) +{ + int retval; + + retval = ctx->ops->alloc_stream(ctx, str_param); + if (retval > 0) + dev_dbg(ctx->dev, "Stream allocated %d\n", retval); + return retval; + +} + +/* + * sst_get_sfreq - this function returns the frequency of the stream + * + * @str_param : stream params + */ +int sst_get_sfreq(struct snd_sst_params *str_param) +{ + switch (str_param->codec) { + case SST_CODEC_TYPE_PCM: + return str_param->sparams.uc.pcm_params.sfreq; + case SST_CODEC_TYPE_AAC: + return str_param->sparams.uc.aac_params.externalsr; + case SST_CODEC_TYPE_MP3: + return 0; + default: + return -EINVAL; + } +} + +/* + * sst_get_sfreq - this function returns the frequency of the stream + * + * @str_param : stream params + */ +int sst_get_num_channel(struct snd_sst_params *str_param) +{ + switch (str_param->codec) { + case SST_CODEC_TYPE_PCM: + return str_param->sparams.uc.pcm_params.num_chan; + case SST_CODEC_TYPE_MP3: + return str_param->sparams.uc.mp3_params.num_chan; + case SST_CODEC_TYPE_AAC: + return str_param->sparams.uc.aac_params.num_chan; + default: + return -EINVAL; + } +} + +/* + * sst_get_stream - this function prepares for stream allocation + * + * @str_param : stream param + */ +int sst_get_stream(struct intel_sst_drv *ctx, + struct snd_sst_params *str_param) +{ + int retval; + struct stream_info *str_info; + + /* stream is not allocated, we are allocating */ + retval = ctx->ops->alloc_stream(ctx, str_param); + if (retval <= 0) { + return -EIO; + } + /* store sampling freq */ + str_info = &ctx->streams[retval]; + str_info->sfreq = sst_get_sfreq(str_param); + + return retval; +} + +static int sst_power_control(struct device *dev, bool state) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + dev_dbg(ctx->dev, "state:%d", state); + if (state == true) + return pm_runtime_get_sync(dev); + else + return sst_pm_runtime_put(ctx); +} + +/* + * sst_open_pcm_stream - Open PCM interface + * + * @str_param: parameters of pcm stream + * + * This function is called by MID sound card driver to open + * a new pcm interface + */ +static int sst_open_pcm_stream(struct device *dev, + struct snd_sst_params *str_param) +{ + int retval; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (!str_param) + return -EINVAL; + + retval = pm_runtime_get_sync(ctx->dev); + if (retval < 0) + return retval; + retval = sst_get_stream(ctx, str_param); + if (retval > 0) { + ctx->stream_cnt++; + } else { + dev_err(ctx->dev, "sst_get_stream returned err %d\n", retval); + sst_pm_runtime_put(ctx); + } + + return retval; +} + +/* + * sst_close_pcm_stream - Close PCM interface + * + * @str_id: stream id to be closed + * + * This function is called by MID sound card driver to close + * an existing pcm interface + */ +static int sst_close_pcm_stream(struct device *dev, unsigned int str_id) +{ + struct stream_info *stream; + int retval = 0; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + stream = get_stream_info(ctx, str_id); + if (!stream) { + dev_err(ctx->dev, "stream info is NULL for str %d!!!\n", str_id); + return -EINVAL; + } + + if (stream->status == STREAM_RESET) { + /* silently fail here as we have cleaned the stream earlier */ + dev_dbg(ctx->dev, "stream in reset state...\n"); + + retval = 0; + goto put; + } + + retval = free_stream_context(ctx, str_id); +put: + stream->pcm_substream = NULL; + stream->status = STREAM_UN_INIT; + stream->period_elapsed = NULL; + ctx->stream_cnt--; + + sst_pm_runtime_put(ctx); + + dev_dbg(ctx->dev, "Exit\n"); + return 0; +} + +static inline int sst_calc_tstamp(struct intel_sst_drv *ctx, + struct pcm_stream_info *info, + struct snd_pcm_substream *substream, + struct snd_sst_tstamp *fw_tstamp) +{ + size_t delay_bytes, delay_frames; + size_t buffer_sz; + u32 pointer_bytes, pointer_samples; + + dev_dbg(ctx->dev, "mrfld ring_buffer_counter %llu in bytes\n", + fw_tstamp->ring_buffer_counter); + dev_dbg(ctx->dev, "mrfld hardware_counter %llu in bytes\n", + fw_tstamp->hardware_counter); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay_bytes = (size_t) (fw_tstamp->ring_buffer_counter - + fw_tstamp->hardware_counter); + else + delay_bytes = (size_t) (fw_tstamp->hardware_counter - + fw_tstamp->ring_buffer_counter); + delay_frames = bytes_to_frames(substream->runtime, delay_bytes); + buffer_sz = snd_pcm_lib_buffer_bytes(substream); + div_u64_rem(fw_tstamp->ring_buffer_counter, buffer_sz, &pointer_bytes); + pointer_samples = bytes_to_samples(substream->runtime, pointer_bytes); + + dev_dbg(ctx->dev, "pcm delay %zu in bytes\n", delay_bytes); + + info->buffer_ptr = pointer_samples / substream->runtime->channels; + + info->pcm_delay = delay_frames / substream->runtime->channels; + dev_dbg(ctx->dev, "buffer ptr %llu pcm_delay rep: %llu\n", + info->buffer_ptr, info->pcm_delay); + return 0; +} + +static int sst_read_timestamp(struct device *dev, struct pcm_stream_info *info) +{ + struct stream_info *stream; + struct snd_pcm_substream *substream; + struct snd_sst_tstamp fw_tstamp; + unsigned int str_id; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + str_id = info->str_id; + stream = get_stream_info(ctx, str_id); + if (!stream) + return -EINVAL; + + if (!stream->pcm_substream) + return -EINVAL; + substream = stream->pcm_substream; + + memcpy_fromio(&fw_tstamp, + ((void *)(ctx->mailbox + ctx->tstamp) + + (str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + return sst_calc_tstamp(ctx, info, substream, &fw_tstamp); +} + +static int sst_stream_start(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + str_info->prev = str_info->status; + str_info->status = STREAM_RUNNING; + sst_start_stream(ctx, str_id); + + return 0; +} + +static int sst_stream_drop(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + str_info->prev = STREAM_UN_INIT; + str_info->status = STREAM_INIT; + return sst_drop_stream(ctx, str_id); +} + +static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) +{ + int str_id = 0; + struct stream_info *stream; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + str_id = str_info->str_id; + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + stream = get_stream_info(ctx, str_id); + if (!stream) + return -EINVAL; + + dev_dbg(ctx->dev, "setting the period ptrs\n"); + stream->pcm_substream = str_info->arg; + stream->period_elapsed = str_info->period_elapsed; + stream->sfreq = str_info->sfreq; + stream->prev = stream->status; + stream->status = STREAM_INIT; + dev_dbg(ctx->dev, + "pcm_substream %p, period_elapsed %p, sfreq %d, status %d\n", + stream->pcm_substream, stream->period_elapsed, + stream->sfreq, stream->status); + + return 0; +} + +/* + * sst_set_byte_stream - Set generic params + * + * @cmd: control cmd to be set + * @arg: command argument + * + * This function is called by MID sound card driver to configure + * SST runtime params. + */ +static int sst_send_byte_stream(struct device *dev, + struct snd_sst_bytes_v2 *bytes) +{ + int ret_val = 0; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (NULL == bytes) + return -EINVAL; + ret_val = pm_runtime_get_sync(ctx->dev); + if (ret_val < 0) + return ret_val; + + ret_val = sst_send_byte_stream_mrfld(ctx, bytes); + sst_pm_runtime_put(ctx); + + return ret_val; +} + +static struct sst_ops pcm_ops = { + .open = sst_open_pcm_stream, + .stream_init = sst_stream_init, + .stream_start = sst_stream_start, + .stream_drop = sst_stream_drop, + .stream_read_tstamp = sst_read_timestamp, + .send_byte_stream = sst_send_byte_stream, + .close = sst_close_pcm_stream, + .power = sst_power_control, +}; + +static struct sst_device sst_dsp_device = { + .name = "Intel(R) SST LPE", + .dev = NULL, + .ops = &pcm_ops, +}; + +/* + * sst_register - function to register DSP + * + * This functions registers DSP with the platform driver + */ +int sst_register(struct device *dev) +{ + int ret_val; + + sst_dsp_device.dev = dev; + ret_val = sst_register_dsp(&sst_dsp_device); + if (ret_val) + dev_err(dev, "Unable to register DSP with platform driver\n"); + + return ret_val; +} + +int sst_unregister(struct device *dev) +{ + return sst_unregister_dsp(&sst_dsp_device); +} -- cgit v1.1 From ea12aa4acd703b507a20354b7af378b1497369e4 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:16 +0530 Subject: ASoC: Intel: sst: Add IPC handling This patch adds APIs to post IPCs and process reply messages. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_ipc.c | 358 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 358 insertions(+) create mode 100644 sound/soc/intel/sst/sst_ipc.c diff --git a/sound/soc/intel/sst/sst_ipc.c b/sound/soc/intel/sst/sst_ipc.c new file mode 100644 index 0000000..41a2b41 --- /dev/null +++ b/sound/soc/intel/sst/sst_ipc.c @@ -0,0 +1,358 @@ +/* + * sst_ipc.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corporation + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + +struct sst_block *sst_create_block(struct intel_sst_drv *ctx, + u32 msg_id, u32 drv_id) +{ + struct sst_block *msg = NULL; + + dev_dbg(ctx->dev, "Enter\n"); + msg = kzalloc(sizeof(*msg), GFP_KERNEL); + if (!msg) + return NULL; + msg->condition = false; + msg->on = true; + msg->msg_id = msg_id; + msg->drv_id = drv_id; + spin_lock_bh(&ctx->block_lock); + list_add_tail(&msg->node, &ctx->block_list); + spin_unlock_bh(&ctx->block_lock); + + return msg; +} + +int sst_wake_up_block(struct intel_sst_drv *ctx, int result, + u32 drv_id, u32 ipc, void *data, u32 size) +{ + struct sst_block *block = NULL; + + dev_dbg(ctx->dev, "Enter\n"); + + spin_lock_bh(&ctx->block_lock); + list_for_each_entry(block, &ctx->block_list, node) { + dev_dbg(ctx->dev, "Block ipc %d, drv_id %d\n", block->msg_id, + block->drv_id); + if (block->msg_id == ipc && block->drv_id == drv_id) { + dev_dbg(ctx->dev, "free up the block\n"); + block->ret_code = result; + block->data = data; + block->size = size; + block->condition = true; + spin_unlock_bh(&ctx->block_lock); + wake_up(&ctx->wait_queue); + return 0; + } + } + spin_unlock_bh(&ctx->block_lock); + dev_dbg(ctx->dev, + "Block not found or a response received for a short msg for ipc %d, drv_id %d\n", + ipc, drv_id); + return -EINVAL; +} + +int sst_free_block(struct intel_sst_drv *ctx, struct sst_block *freed) +{ + struct sst_block *block = NULL, *__block; + + dev_dbg(ctx->dev, "Enter\n"); + spin_lock_bh(&ctx->block_lock); + list_for_each_entry_safe(block, __block, &ctx->block_list, node) { + if (block == freed) { + pr_debug("pvt_id freed --> %d\n", freed->drv_id); + /* toggle the index position of pvt_id */ + list_del(&freed->node); + spin_unlock_bh(&ctx->block_lock); + kfree(freed->data); + freed->data = NULL; + kfree(freed); + return 0; + } + } + spin_unlock_bh(&ctx->block_lock); + dev_err(ctx->dev, "block is already freed!!!\n"); + return -EINVAL; +} + +int sst_post_message_mrfld(struct intel_sst_drv *sst_drv_ctx, + struct ipc_post *ipc_msg, bool sync) +{ + struct ipc_post *msg = ipc_msg; + union ipc_header_mrfld header; + unsigned int loop_count = 0; + int retval = 0; + unsigned long irq_flags; + + dev_dbg(sst_drv_ctx->dev, "Enter: sync: %d\n", sync); + spin_lock_irqsave(&sst_drv_ctx->ipc_spin_lock, irq_flags); + header.full = sst_shim_read64(sst_drv_ctx->shim, SST_IPCX); + if (sync) { + while (header.p.header_high.part.busy) { + if (loop_count > 25) { + dev_err(sst_drv_ctx->dev, + "sst: Busy wait failed, cant send this msg\n"); + retval = -EBUSY; + goto out; + } + cpu_relax(); + loop_count++; + header.full = sst_shim_read64(sst_drv_ctx->shim, SST_IPCX); + } + } else { + if (list_empty(&sst_drv_ctx->ipc_dispatch_list)) { + /* queue is empty, nothing to send */ + spin_unlock_irqrestore(&sst_drv_ctx->ipc_spin_lock, irq_flags); + dev_dbg(sst_drv_ctx->dev, + "Empty msg queue... NO Action\n"); + return 0; + } + + if (header.p.header_high.part.busy) { + spin_unlock_irqrestore(&sst_drv_ctx->ipc_spin_lock, irq_flags); + dev_dbg(sst_drv_ctx->dev, "Busy not free... post later\n"); + return 0; + } + + /* copy msg from list */ + msg = list_entry(sst_drv_ctx->ipc_dispatch_list.next, + struct ipc_post, node); + list_del(&msg->node); + } + dev_dbg(sst_drv_ctx->dev, "sst: Post message: header = %x\n", + msg->mrfld_header.p.header_high.full); + dev_dbg(sst_drv_ctx->dev, "sst: size = 0x%x\n", + msg->mrfld_header.p.header_low_payload); + + if (msg->mrfld_header.p.header_high.part.large) + memcpy_toio(sst_drv_ctx->mailbox + SST_MAILBOX_SEND, + msg->mailbox_data, + msg->mrfld_header.p.header_low_payload); + + sst_shim_write64(sst_drv_ctx->shim, SST_IPCX, msg->mrfld_header.full); + +out: + spin_unlock_irqrestore(&sst_drv_ctx->ipc_spin_lock, irq_flags); + kfree(msg->mailbox_data); + kfree(msg); + return retval; +} + +void intel_sst_clear_intr_mrfld(struct intel_sst_drv *sst_drv_ctx) +{ + union interrupt_reg_mrfld isr; + union interrupt_reg_mrfld imr; + union ipc_header_mrfld clear_ipc; + unsigned long irq_flags; + + spin_lock_irqsave(&sst_drv_ctx->ipc_spin_lock, irq_flags); + imr.full = sst_shim_read64(sst_drv_ctx->shim, SST_IMRX); + isr.full = sst_shim_read64(sst_drv_ctx->shim, SST_ISRX); + + /* write 1 to clear*/ + isr.part.busy_interrupt = 1; + sst_shim_write64(sst_drv_ctx->shim, SST_ISRX, isr.full); + + /* Set IA done bit */ + clear_ipc.full = sst_shim_read64(sst_drv_ctx->shim, SST_IPCD); + + clear_ipc.p.header_high.part.busy = 0; + clear_ipc.p.header_high.part.done = 1; + clear_ipc.p.header_low_payload = IPC_ACK_SUCCESS; + sst_shim_write64(sst_drv_ctx->shim, SST_IPCD, clear_ipc.full); + /* un mask busy interrupt */ + imr.part.busy_interrupt = 0; + sst_shim_write64(sst_drv_ctx->shim, SST_IMRX, imr.full); + spin_unlock_irqrestore(&sst_drv_ctx->ipc_spin_lock, irq_flags); +} + + +/* + * process_fw_init - process the FW init msg + * + * @msg: IPC message mailbox data from FW + * + * This function processes the FW init msg from FW + * marks FW state and prints debug info of loaded FW + */ +static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, + void *msg) +{ + struct ipc_header_fw_init *init = + (struct ipc_header_fw_init *)msg; + int retval = 0; + + dev_dbg(sst_drv_ctx->dev, "*** FW Init msg came***\n"); + if (init->result) { + sst_drv_ctx->sst_state = SST_RESET; + dev_err(sst_drv_ctx->dev, "FW Init failed, Error %x\n", + init->result); + retval = init->result; + goto ret; + } + +ret: + sst_wake_up_block(sst_drv_ctx, retval, FW_DWNL_ID, 0 , NULL, 0); +} + +static void process_fw_async_msg(struct intel_sst_drv *sst_drv_ctx, + struct ipc_post *msg) +{ + u32 msg_id; + int str_id; + u32 data_size, i; + void *data_offset; + struct stream_info *stream; + union ipc_header_high msg_high; + u32 msg_low, pipe_id; + + msg_high = msg->mrfld_header.p.header_high; + msg_low = msg->mrfld_header.p.header_low_payload; + msg_id = ((struct ipc_dsp_hdr *)msg->mailbox_data)->cmd_id; + data_offset = (msg->mailbox_data + sizeof(struct ipc_dsp_hdr)); + data_size = msg_low - (sizeof(struct ipc_dsp_hdr)); + + switch (msg_id) { + case IPC_SST_PERIOD_ELAPSED_MRFLD: + pipe_id = ((struct ipc_dsp_hdr *)msg->mailbox_data)->pipe_id; + str_id = get_stream_id_mrfld(sst_drv_ctx, pipe_id); + if (str_id > 0) { + dev_dbg(sst_drv_ctx->dev, + "Period elapsed rcvd for pipe id 0x%x\n", + pipe_id); + stream = &sst_drv_ctx->streams[str_id]; + if (stream->period_elapsed) + stream->period_elapsed(stream->pcm_substream); + if (stream->compr_cb) + stream->compr_cb(stream->compr_cb_param); + } + break; + + case IPC_IA_DRAIN_STREAM_MRFLD: + pipe_id = ((struct ipc_dsp_hdr *)msg->mailbox_data)->pipe_id; + str_id = get_stream_id_mrfld(sst_drv_ctx, pipe_id); + if (str_id > 0) { + stream = &sst_drv_ctx->streams[str_id]; + if (stream->drain_notify) + stream->drain_notify(stream->drain_cb_param); + } + break; + + case IPC_IA_FW_ASYNC_ERR_MRFLD: + dev_err(sst_drv_ctx->dev, "FW sent async error msg:\n"); + for (i = 0; i < (data_size/4); i++) + print_hex_dump(KERN_DEBUG, NULL, DUMP_PREFIX_NONE, + 16, 4, data_offset, data_size, false); + break; + + case IPC_IA_FW_INIT_CMPLT_MRFLD: + process_fw_init(sst_drv_ctx, data_offset); + break; + + case IPC_IA_BUF_UNDER_RUN_MRFLD: + pipe_id = ((struct ipc_dsp_hdr *)msg->mailbox_data)->pipe_id; + str_id = get_stream_id_mrfld(sst_drv_ctx, pipe_id); + if (str_id > 0) + dev_err(sst_drv_ctx->dev, + "Buffer under-run for pipe:%#x str_id:%d\n", + pipe_id, str_id); + break; + + default: + dev_err(sst_drv_ctx->dev, + "Unrecognized async msg from FW msg_id %#x\n", msg_id); + } +} + +void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, + struct ipc_post *msg) +{ + unsigned int drv_id; + void *data; + union ipc_header_high msg_high; + u32 msg_low; + struct ipc_dsp_hdr *dsp_hdr; + unsigned int cmd_id; + + msg_high = msg->mrfld_header.p.header_high; + msg_low = msg->mrfld_header.p.header_low_payload; + + dev_dbg(sst_drv_ctx->dev, "IPC process message header %x payload %x\n", + msg->mrfld_header.p.header_high.full, + msg->mrfld_header.p.header_low_payload); + + drv_id = msg_high.part.drv_id; + + /* Check for async messages first */ + if (drv_id == SST_ASYNC_DRV_ID) { + /*FW sent async large message*/ + process_fw_async_msg(sst_drv_ctx, msg); + return; + } + + /* FW sent short error response for an IPC */ + if (msg_high.part.result && drv_id && !msg_high.part.large) { + /* 32-bit FW error code in msg_low */ + dev_err(sst_drv_ctx->dev, "FW sent error response 0x%x", msg_low); + sst_wake_up_block(sst_drv_ctx, msg_high.part.result, + msg_high.part.drv_id, + msg_high.part.msg_id, NULL, 0); + return; + } + + /* + * Process all valid responses + * if it is a large message, the payload contains the size to + * copy from mailbox + **/ + if (msg_high.part.large) { + data = kzalloc(msg_low, GFP_KERNEL); + if (!data) + return; + memcpy(data, (void *) msg->mailbox_data, msg_low); + /* Copy command id so that we can use to put sst to reset */ + dsp_hdr = (struct ipc_dsp_hdr *)data; + cmd_id = dsp_hdr->cmd_id; + dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); + if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, + msg_high.part.drv_id, + msg_high.part.msg_id, data, msg_low)) + kfree(data); + } else { + sst_wake_up_block(sst_drv_ctx, msg_high.part.result, + msg_high.part.drv_id, + msg_high.part.msg_id, NULL, 0); + } + +} -- cgit v1.1 From 3d9ff34622badd65543430a784f7af9838c5c3fc Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:17 +0530 Subject: ASoC: Intel: sst: add stream operations This patch adds pcm and compressed stream control operations. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_stream.c | 437 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 437 insertions(+) create mode 100644 sound/soc/intel/sst/sst_stream.c diff --git a/sound/soc/intel/sst/sst_stream.c b/sound/soc/intel/sst/sst_stream.c new file mode 100644 index 0000000..dae2a41 --- /dev/null +++ b/sound/soc/intel/sst/sst_stream.c @@ -0,0 +1,437 @@ +/* + * sst_stream.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + +int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) +{ + struct snd_sst_alloc_mrfld alloc_param; + struct snd_sst_params *str_params; + struct snd_sst_tstamp fw_tstamp; + struct stream_info *str_info; + struct snd_sst_alloc_response *response; + unsigned int str_id, pipe_id, task_id; + int i, num_ch, ret = 0; + void *data = NULL; + + dev_dbg(sst_drv_ctx->dev, "Enter\n"); + BUG_ON(!params); + + str_params = (struct snd_sst_params *)params; + memset(&alloc_param, 0, sizeof(alloc_param)); + alloc_param.operation = str_params->ops; + alloc_param.codec_type = str_params->codec; + alloc_param.sg_count = str_params->aparams.sg_count; + alloc_param.ring_buf_info[0].addr = + str_params->aparams.ring_buf_info[0].addr; + alloc_param.ring_buf_info[0].size = + str_params->aparams.ring_buf_info[0].size; + alloc_param.frag_size = str_params->aparams.frag_size; + + memcpy(&alloc_param.codec_params, &str_params->sparams, + sizeof(struct snd_sst_stream_params)); + + /* + * fill channel map params for multichannel support. + * Ideally channel map should be received from upper layers + * for multichannel support. + * Currently hardcoding as per FW reqm. + */ + num_ch = sst_get_num_channel(str_params); + for (i = 0; i < 8; i++) { + if (i < num_ch) + alloc_param.codec_params.uc.pcm_params.channel_map[i] = i; + else + alloc_param.codec_params.uc.pcm_params.channel_map[i] = 0xFF; + } + + str_id = str_params->stream_id; + str_info = get_stream_info(sst_drv_ctx, str_id); + if (str_info == NULL) { + dev_err(sst_drv_ctx->dev, "get stream info returned null\n"); + return -EINVAL; + } + + pipe_id = str_params->device_type; + task_id = str_params->task; + sst_drv_ctx->streams[str_id].pipe_id = pipe_id; + sst_drv_ctx->streams[str_id].task_id = task_id; + sst_drv_ctx->streams[str_id].num_ch = num_ch; + + if (sst_drv_ctx->info.lpe_viewpt_rqd) + alloc_param.ts = sst_drv_ctx->info.mailbox_start + + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); + else + alloc_param.ts = sst_drv_ctx->mailbox_add + + sst_drv_ctx->tstamp + (str_id * sizeof(fw_tstamp)); + + dev_dbg(sst_drv_ctx->dev, "alloc tstamp location = 0x%x\n", + alloc_param.ts); + dev_dbg(sst_drv_ctx->dev, "assigned pipe id 0x%x to task %d\n", + pipe_id, task_id); + + /* allocate device type context */ + sst_init_stream(&sst_drv_ctx->streams[str_id], alloc_param.codec_type, + str_id, alloc_param.operation, 0); + + dev_info(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", + str_id, pipe_id); + ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, + IPC_IA_ALLOC_STREAM_MRFLD, pipe_id, sizeof(alloc_param), + &alloc_param, data, true, true, false, true); + + if (ret < 0) { + dev_err(sst_drv_ctx->dev, "FW alloc failed ret %d\n", ret); + /* alloc failed, so reset the state to uninit */ + str_info->status = STREAM_UN_INIT; + str_id = ret; + } else if (data) { + response = (struct snd_sst_alloc_response *)data; + ret = response->str_type.result; + if (!ret) + goto out; + dev_err(sst_drv_ctx->dev, "FW alloc failed ret %d\n", ret); + if (ret == SST_ERR_STREAM_IN_USE) { + dev_err(sst_drv_ctx->dev, + "FW not in clean state, send free for:%d\n", str_id); + sst_free_stream(sst_drv_ctx, str_id); + } + str_id = -ret; + } +out: + kfree(data); + return str_id; +} + +/** +* sst_start_stream - Send msg for a starting stream +* @str_id: stream ID +* +* This function is called by any function which wants to start +* a stream. +*/ +int sst_start_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + int retval = 0; + struct stream_info *str_info; + u16 data = 0; + + dev_dbg(sst_drv_ctx->dev, "sst_start_stream for %d\n", str_id); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + if (str_info->status != STREAM_RUNNING) + return -EBADRQC; + + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, + IPC_CMD, IPC_IA_START_STREAM_MRFLD, str_info->pipe_id, + sizeof(u16), &data, NULL, true, true, true, false); + + return retval; +} + +int sst_send_byte_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, + struct snd_sst_bytes_v2 *bytes) +{ struct ipc_post *msg = NULL; + u32 length; + int pvt_id, ret = 0; + struct sst_block *block = NULL; + + dev_dbg(sst_drv_ctx->dev, + "type:%u ipc_msg:%u block:%u task_id:%u pipe: %#x length:%#x\n", + bytes->type, bytes->ipc_msg, bytes->block, bytes->task_id, + bytes->pipe_id, bytes->len); + + if (sst_create_ipc_msg(&msg, true)) + return -ENOMEM; + + pvt_id = sst_assign_pvt_id(sst_drv_ctx); + sst_fill_header_mrfld(&msg->mrfld_header, bytes->ipc_msg, + bytes->task_id, 1, pvt_id); + msg->mrfld_header.p.header_high.part.res_rqd = bytes->block; + length = bytes->len; + msg->mrfld_header.p.header_low_payload = length; + dev_dbg(sst_drv_ctx->dev, "length is %d\n", length); + memcpy(msg->mailbox_data, &bytes->bytes, bytes->len); + if (bytes->block) { + block = sst_create_block(sst_drv_ctx, bytes->ipc_msg, pvt_id); + if (block == NULL) { + kfree(msg); + ret = -ENOMEM; + goto out; + } + } + + sst_add_to_dispatch_list_and_post(sst_drv_ctx, msg); + dev_dbg(sst_drv_ctx->dev, "msg->mrfld_header.p.header_low_payload:%d", + msg->mrfld_header.p.header_low_payload); + + if (bytes->block) { + ret = sst_wait_timeout(sst_drv_ctx, block); + if (ret) { + dev_err(sst_drv_ctx->dev, "fw returned err %d\n", ret); + sst_free_block(sst_drv_ctx, block); + goto out; + } + } + if (bytes->type == SND_SST_BYTES_GET) { + /* + * copy the reply and send back + * we need to update only sz and payload + */ + if (bytes->block) { + unsigned char *r = block->data; + + dev_dbg(sst_drv_ctx->dev, "read back %d bytes", + bytes->len); + memcpy(bytes->bytes, r, bytes->len); + } + } + if (bytes->block) + sst_free_block(sst_drv_ctx, block); +out: + test_and_clear_bit(pvt_id, &sst_drv_ctx->pvt_id); + return 0; +} + +/* + * sst_pause_stream - Send msg for a pausing stream + * @str_id: stream ID + * + * This function is called by any function which wants to pause + * an already running stream. + */ +int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + int retval = 0; + struct stream_info *str_info; + + dev_dbg(sst_drv_ctx->dev, "SST DBG:sst_pause_stream for %d\n", str_id); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + if (str_info->status == STREAM_PAUSED) + return 0; + if (str_info->status == STREAM_RUNNING || + str_info->status == STREAM_INIT) { + if (str_info->prev == STREAM_UN_INIT) + return -EBADRQC; + + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, + IPC_IA_PAUSE_STREAM_MRFLD, str_info->pipe_id, + 0, NULL, NULL, true, true, false, true); + + if (retval == 0) { + str_info->prev = str_info->status; + str_info->status = STREAM_PAUSED; + } else if (retval == SST_ERR_INVALID_STREAM_ID) { + retval = -EINVAL; + mutex_lock(&sst_drv_ctx->sst_lock); + sst_clean_stream(str_info); + mutex_unlock(&sst_drv_ctx->sst_lock); + } + } else { + retval = -EBADRQC; + dev_dbg(sst_drv_ctx->dev, "SST DBG:BADRQC for stream\n "); + } + + return retval; +} + +/** + * sst_resume_stream - Send msg for resuming stream + * @str_id: stream ID + * + * This function is called by any function which wants to resume + * an already paused stream. + */ +int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + int retval = 0; + struct stream_info *str_info; + + dev_dbg(sst_drv_ctx->dev, "SST DBG:sst_resume_stream for %d\n", str_id); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + if (str_info->status == STREAM_RUNNING) + return 0; + if (str_info->status == STREAM_PAUSED) { + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, + IPC_CMD, IPC_IA_RESUME_STREAM_MRFLD, + str_info->pipe_id, 0, NULL, NULL, + true, true, false, true); + + if (!retval) { + if (str_info->prev == STREAM_RUNNING) + str_info->status = STREAM_RUNNING; + else + str_info->status = STREAM_INIT; + str_info->prev = STREAM_PAUSED; + } else if (retval == -SST_ERR_INVALID_STREAM_ID) { + retval = -EINVAL; + mutex_lock(&sst_drv_ctx->sst_lock); + sst_clean_stream(str_info); + mutex_unlock(&sst_drv_ctx->sst_lock); + } + } else { + retval = -EBADRQC; + dev_err(sst_drv_ctx->dev, "SST ERR: BADQRC for stream\n"); + } + + return retval; +} + + +/** + * sst_drop_stream - Send msg for stopping stream + * @str_id: stream ID + * + * This function is called by any function which wants to stop + * a stream. + */ +int sst_drop_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + int retval = 0; + struct stream_info *str_info; + + dev_dbg(sst_drv_ctx->dev, "SST DBG:sst_drop_stream for %d\n", str_id); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + + if (str_info->status != STREAM_UN_INIT) { + str_info->prev = STREAM_UN_INIT; + str_info->status = STREAM_INIT; + str_info->cumm_bytes = 0; + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, + IPC_CMD, IPC_IA_DROP_STREAM_MRFLD, + str_info->pipe_id, 0, NULL, NULL, + true, true, true, false); + } else { + retval = -EBADRQC; + dev_dbg(sst_drv_ctx->dev, "BADQRC for stream, state %x\n", + str_info->status); + } + return retval; +} + +/** +* sst_drain_stream - Send msg for draining stream +* @str_id: stream ID +* +* This function is called by any function which wants to drain +* a stream. +*/ +int sst_drain_stream(struct intel_sst_drv *sst_drv_ctx, + int str_id, bool partial_drain) +{ + int retval = 0; + struct stream_info *str_info; + + dev_dbg(sst_drv_ctx->dev, "SST DBG:sst_drain_stream for %d\n", str_id); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + if (str_info->status != STREAM_RUNNING && + str_info->status != STREAM_INIT && + str_info->status != STREAM_PAUSED) { + dev_err(sst_drv_ctx->dev, "SST ERR: BADQRC for stream = %d\n", + str_info->status); + return -EBADRQC; + } + + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, + IPC_IA_DRAIN_STREAM_MRFLD, str_info->pipe_id, + sizeof(u8), &partial_drain, NULL, true, true, false, false); + /* + * with new non blocked drain implementation in core we dont need to + * wait for respsonse, and need to only invoke callback for drain + * complete + */ + + return retval; +} + +/** + * sst_free_stream - Frees a stream + * @str_id: stream ID + * + * This function is called by any function which wants to free + * a stream. + */ +int sst_free_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + int retval = 0; + struct stream_info *str_info; + struct intel_sst_ops *ops; + + dev_dbg(sst_drv_ctx->dev, "SST DBG:sst_free_stream for %d\n", str_id); + + mutex_lock(&sst_drv_ctx->sst_lock); + if (sst_drv_ctx->sst_state == SST_RESET) { + mutex_unlock(&sst_drv_ctx->sst_lock); + return -ENODEV; + } + mutex_unlock(&sst_drv_ctx->sst_lock); + str_info = get_stream_info(sst_drv_ctx, str_id); + if (!str_info) + return -EINVAL; + ops = sst_drv_ctx->ops; + + mutex_lock(&str_info->lock); + if (str_info->status != STREAM_UN_INIT) { + str_info->prev = str_info->status; + str_info->status = STREAM_UN_INIT; + mutex_unlock(&str_info->lock); + + dev_info(sst_drv_ctx->dev, "Free for str %d pipe %#x\n", + str_id, str_info->pipe_id); + retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, + IPC_IA_FREE_STREAM_MRFLD, str_info->pipe_id, 0, + NULL, NULL, true, true, false, true); + + dev_dbg(sst_drv_ctx->dev, "sst: wait for free returned %d\n", + retval); + mutex_lock(&sst_drv_ctx->sst_lock); + sst_clean_stream(str_info); + mutex_unlock(&sst_drv_ctx->sst_lock); + dev_dbg(sst_drv_ctx->dev, "SST DBG:Stream freed\n"); + } else { + mutex_unlock(&str_info->lock); + retval = -EBADRQC; + dev_dbg(sst_drv_ctx->dev, "SST DBG:BADQRC for stream\n"); + } + + return retval; +} -- cgit v1.1 From 60dc8dbacb001b6400624ee72990b85d6d44bce6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:18 +0530 Subject: ASoC: Intel: sst: Add some helper functions This patch adds helper functions like wait, creating ipc headers, shim wrappers. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_pvt.c | 446 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 446 insertions(+) create mode 100644 sound/soc/intel/sst/sst_pvt.c diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c new file mode 100644 index 0000000..9e5f69b --- /dev/null +++ b/sound/soc/intel/sst/sst_pvt.c @@ -0,0 +1,446 @@ +/* + * sst_pvt.c - Intel SST Driver for audio engine + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" +#include "../sst-dsp.h" + +int sst_shim_write(void __iomem *addr, int offset, int value) +{ + writel(value, addr + offset); + return 0; +} + +u32 sst_shim_read(void __iomem *addr, int offset) +{ + return readl(addr + offset); +} + +u64 sst_reg_read64(void __iomem *addr, int offset) +{ + u64 val = 0; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + + return val; +} + +int sst_shim_write64(void __iomem *addr, int offset, u64 value) +{ + memcpy_toio(addr + offset, &value, sizeof(value)); + return 0; +} + +u64 sst_shim_read64(void __iomem *addr, int offset) +{ + u64 val = 0; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + return val; +} + +void sst_set_fw_state_locked( + struct intel_sst_drv *sst_drv_ctx, int sst_state) +{ + mutex_lock(&sst_drv_ctx->sst_lock); + sst_drv_ctx->sst_state = sst_state; + mutex_unlock(&sst_drv_ctx->sst_lock); +} + +/* + * sst_wait_interruptible - wait on event + * + * @sst_drv_ctx: Driver context + * @block: Driver block to wait on + * + * This function waits without a timeout (and is interruptable) for a + * given block event + */ +int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, + struct sst_block *block) +{ + int retval = 0; + + if (!wait_event_interruptible(sst_drv_ctx->wait_queue, + block->condition)) { + /* event wake */ + if (block->ret_code < 0) { + dev_err(sst_drv_ctx->dev, + "stream failed %d\n", block->ret_code); + retval = -EBUSY; + } else { + dev_dbg(sst_drv_ctx->dev, "event up\n"); + retval = 0; + } + } else { + dev_err(sst_drv_ctx->dev, "signal interrupted\n"); + retval = -EINTR; + } + return retval; + +} + +unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) +{ + unsigned long long val = 0; + + switch (sst->pci_id) { + case SST_MRFLD_PCI_ID: + val = sst_shim_read64(sst->shim, addr); + break; + } + return val; +} + +void write_shim_data(struct intel_sst_drv *sst, int addr, + unsigned long long data) +{ + switch (sst->pci_id) { + case SST_MRFLD_PCI_ID: + sst_shim_write64(sst->shim, addr, (u64) data); + break; + } +} + +/* + * sst_wait_timeout - wait on event for timeout + * + * @sst_drv_ctx: Driver context + * @block: Driver block to wait on + * + * This function waits with a timeout value (and is not interruptible) on a + * given block event + */ +int sst_wait_timeout(struct intel_sst_drv *sst_drv_ctx, struct sst_block *block) +{ + int retval = 0; + + /* + * NOTE: + * Observed that FW processes the alloc msg and replies even + * before the alloc thread has finished execution + */ + dev_dbg(sst_drv_ctx->dev, + "waiting for condition %x ipc %d drv_id %d\n", + block->condition, block->msg_id, block->drv_id); + if (wait_event_timeout(sst_drv_ctx->wait_queue, + block->condition, + msecs_to_jiffies(SST_BLOCK_TIMEOUT))) { + /* event wake */ + dev_dbg(sst_drv_ctx->dev, "Event wake %x\n", + block->condition); + dev_dbg(sst_drv_ctx->dev, "message ret: %d\n", + block->ret_code); + retval = -block->ret_code; + } else { + block->on = false; + dev_err(sst_drv_ctx->dev, + "Wait timed-out condition:%#x, msg_id:%#x fw_state %#x\n", + block->condition, block->msg_id, sst_drv_ctx->sst_state); + sst_drv_ctx->sst_state = SST_RESET; + + retval = -EBUSY; + } + return retval; +} + +/* + * sst_create_ipc_msg - create a IPC message + * + * @arg: ipc message + * @large: large or short message + * + * this function allocates structures to send a large or short + * message to the firmware + */ +int sst_create_ipc_msg(struct ipc_post **arg, bool large) +{ + struct ipc_post *msg; + + msg = kzalloc(sizeof(struct ipc_post), GFP_ATOMIC); + if (!msg) + return -ENOMEM; + if (large) { + msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC); + if (!msg->mailbox_data) { + kfree(msg); + return -ENOMEM; + } + } else { + msg->mailbox_data = NULL; + } + msg->is_large = large; + *arg = msg; + return 0; +} + +/* + * sst_create_block_and_ipc_msg - Creates IPC message and sst block + * @arg: passed to sst_create_ipc_message API + * @large: large or short message + * @sst_drv_ctx: sst driver context + * @block: return block allocated + * @msg_id: IPC + * @drv_id: stream id or private id + */ +int sst_create_block_and_ipc_msg(struct ipc_post **arg, bool large, + struct intel_sst_drv *sst_drv_ctx, struct sst_block **block, + u32 msg_id, u32 drv_id) +{ + int retval = 0; + + retval = sst_create_ipc_msg(arg, large); + if (retval) + return retval; + *block = sst_create_block(sst_drv_ctx, msg_id, drv_id); + if (*block == NULL) { + kfree(*arg); + return -ENOMEM; + } + return retval; +} + +/* + * sst_clean_stream - clean the stream context + * + * @stream: stream structure + * + * this function resets the stream contexts + * should be called in free + */ +void sst_clean_stream(struct stream_info *stream) +{ + stream->status = STREAM_UN_INIT; + stream->prev = STREAM_UN_INIT; + mutex_lock(&stream->lock); + stream->cumm_bytes = 0; + mutex_unlock(&stream->lock); +} + +int sst_prepare_and_post_msg(struct intel_sst_drv *sst, + int task_id, int ipc_msg, int cmd_id, int pipe_id, + size_t mbox_data_len, const void *mbox_data, void **data, + bool large, bool fill_dsp, bool sync, bool response) +{ + struct ipc_post *msg = NULL; + struct ipc_dsp_hdr dsp_hdr; + struct sst_block *block; + int ret = 0, pvt_id; + + pvt_id = sst_assign_pvt_id(sst); + if (pvt_id < 0) + return pvt_id; + + if (response) + ret = sst_create_block_and_ipc_msg( + &msg, large, sst, &block, ipc_msg, pvt_id); + else + ret = sst_create_ipc_msg(&msg, large); + + if (ret < 0) { + test_and_clear_bit(pvt_id, &sst->pvt_id); + return -ENOMEM; + } + + dev_dbg(sst->dev, "pvt_id = %d, pipe id = %d, task = %d ipc_msg: %d\n", + pvt_id, pipe_id, task_id, ipc_msg); + sst_fill_header_mrfld(&msg->mrfld_header, ipc_msg, + task_id, large, pvt_id); + msg->mrfld_header.p.header_low_payload = sizeof(dsp_hdr) + mbox_data_len; + msg->mrfld_header.p.header_high.part.res_rqd = !sync; + dev_dbg(sst->dev, "header:%x\n", + msg->mrfld_header.p.header_high.full); + dev_dbg(sst->dev, "response rqd: %x", + msg->mrfld_header.p.header_high.part.res_rqd); + dev_dbg(sst->dev, "msg->mrfld_header.p.header_low_payload:%d", + msg->mrfld_header.p.header_low_payload); + if (fill_dsp) { + sst_fill_header_dsp(&dsp_hdr, cmd_id, pipe_id, mbox_data_len); + memcpy(msg->mailbox_data, &dsp_hdr, sizeof(dsp_hdr)); + if (mbox_data_len) { + memcpy(msg->mailbox_data + sizeof(dsp_hdr), + mbox_data, mbox_data_len); + } + } + + if (sync) + sst->ops->post_message(sst, msg, true); + else + sst_add_to_dispatch_list_and_post(sst, msg); + + if (response) { + ret = sst_wait_timeout(sst, block); + if (ret < 0) { + goto out; + } else if(block->data) { + if (!data) + goto out; + *data = kzalloc(block->size, GFP_KERNEL); + if (!(*data)) { + ret = -ENOMEM; + goto out; + } else + memcpy(data, (void *) block->data, block->size); + } + } +out: + if (response) + sst_free_block(sst, block); + test_and_clear_bit(pvt_id, &sst->pvt_id); + return ret; +} + +int sst_pm_runtime_put(struct intel_sst_drv *sst_drv) +{ + int ret; + + pm_runtime_mark_last_busy(sst_drv->dev); + ret = pm_runtime_put_autosuspend(sst_drv->dev); + if (ret < 0) + return ret; + return 0; +} + +void sst_fill_header_mrfld(union ipc_header_mrfld *header, + int msg, int task_id, int large, int drv_id) +{ + header->full = 0; + header->p.header_high.part.msg_id = msg; + header->p.header_high.part.task_id = task_id; + header->p.header_high.part.large = large; + header->p.header_high.part.drv_id = drv_id; + header->p.header_high.part.done = 0; + header->p.header_high.part.busy = 1; + header->p.header_high.part.res_rqd = 1; +} + +void sst_fill_header_dsp(struct ipc_dsp_hdr *dsp, int msg, + int pipe_id, int len) +{ + dsp->cmd_id = msg; + dsp->mod_index_id = 0xff; + dsp->pipe_id = pipe_id; + dsp->length = len; + dsp->mod_id = 0; +} + +#define SST_MAX_BLOCKS 15 +/* + * sst_assign_pvt_id - assign a pvt id for stream + * + * @sst_drv_ctx : driver context + * + * this function assigns a private id for calls that dont have stream + * context yet, should be called with lock held + * uses bits for the id, and finds first free bits and assigns that + */ +int sst_assign_pvt_id(struct intel_sst_drv *drv) +{ + int local; + + spin_lock(&drv->block_lock); + /* find first zero index from lsb */ + local = ffz(drv->pvt_id); + dev_dbg(drv->dev, "pvt_id assigned --> %d\n", local); + if (local >= SST_MAX_BLOCKS){ + spin_unlock(&drv->block_lock); + dev_err(drv->dev, "PVT _ID error: no free id blocks "); + return -EINVAL; + } + /* toggle the index */ + change_bit(local, &drv->pvt_id); + spin_unlock(&drv->block_lock); + return local; +} + +void sst_init_stream(struct stream_info *stream, + int codec, int sst_id, int ops, u8 slot) +{ + stream->status = STREAM_INIT; + stream->prev = STREAM_UN_INIT; + stream->ops = ops; +} + +int sst_validate_strid( + struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + if (str_id <= 0 || str_id > sst_drv_ctx->info.max_streams) { + dev_err(sst_drv_ctx->dev, + "SST ERR: invalid stream id : %d, max %d\n", + str_id, sst_drv_ctx->info.max_streams); + return -EINVAL; + } + + return 0; +} + +struct stream_info *get_stream_info( + struct intel_sst_drv *sst_drv_ctx, int str_id) +{ + if (sst_validate_strid(sst_drv_ctx, str_id)) + return NULL; + return &sst_drv_ctx->streams[str_id]; +} + +int get_stream_id_mrfld(struct intel_sst_drv *sst_drv_ctx, + u32 pipe_id) +{ + int i; + + for (i = 1; i <= sst_drv_ctx->info.max_streams; i++) + if (pipe_id == sst_drv_ctx->streams[i].pipe_id) + return i; + + dev_dbg(sst_drv_ctx->dev, "no such pipe_id(%u)", pipe_id); + return -1; +} + +u32 relocate_imr_addr_mrfld(u32 base_addr) +{ + /* Get the difference from 512MB aligned base addr */ + /* relocate the base */ + base_addr = MRFLD_FW_VIRTUAL_BASE + (base_addr % (512 * 1024 * 1024)); + return base_addr; +} + +void sst_add_to_dispatch_list_and_post(struct intel_sst_drv *sst, + struct ipc_post *msg) +{ + unsigned long irq_flags; + + spin_lock_irqsave(&sst->ipc_spin_lock, irq_flags); + list_add_tail(&msg->node, &sst->ipc_dispatch_list); + spin_unlock_irqrestore(&sst->ipc_spin_lock, irq_flags); + sst->ops->post_message(sst, NULL, false); +} -- cgit v1.1 From 0fbc7d7320202489383f520ecd1758b15a00e17c Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Oct 2014 20:00:19 +0530 Subject: ASoC: Intel: sst: Add makefile and kconfig changes Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 4 ++++ sound/soc/intel/Makefile | 3 +++ sound/soc/intel/sst/Makefile | 3 +++ 3 files changed, 10 insertions(+) create mode 100644 sound/soc/intel/sst/Makefile diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f5b4a9c7..c719438 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -3,6 +3,7 @@ config SND_MFLD_MACHINE depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM + select SND_SST_IPC help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices @@ -12,6 +13,9 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate +config SND_SST_IPC + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index f841786..9ab43be 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,3 +31,6 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o + +# DSP driver +obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile new file mode 100644 index 0000000..4d0e79b --- /dev/null +++ b/sound/soc/intel/sst/Makefile @@ -0,0 +1,3 @@ +snd-intel-sst-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o + +obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst.o -- cgit v1.1 From 282a331fe25c74c23800bb7da1bb62c9e54fd738 Mon Sep 17 00:00:00 2001 From: Ricardo Neri Date: Thu, 16 Oct 2014 18:22:01 -0700 Subject: ASoC: Intel: Add new dependency for Broadwell machine I2C support for the RT286 codec is provided through the Designware I2C platform adapter in this machine. Thus, the adapter driver must be present. Signed-off-by: Ricardo Neri Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c719438..774fab9 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -65,7 +65,8 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\ + I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 -- cgit v1.1 From 161aa49ef1b99891b82d3599885eb8d5cbf0ebfc Mon Sep 17 00:00:00 2001 From: Ricardo Neri Date: Thu, 16 Oct 2014 18:22:02 -0700 Subject: ASoC: Intel: Add new dependency for Haswell machine I2C support for the RT5640 codec is provided through the Designware I2C platform adapter in this machine. Thus, the driver must be present. Signed-off-by: Ricardo Neri Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 774fab9..2a3af88 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -36,7 +36,8 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\ + I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help -- cgit v1.1 From cd6e82b814ca73c474b1a2fa48a54b251da44655 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 7 Oct 2014 10:25:37 +0800 Subject: ASoC: rt5645: Add the workqueue of the jack detect function for the debouncing Add the workqueue of the jack detect function for the debouncing. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 15 ++++++++++++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..57ba742 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2166,11 +2166,20 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +static void rt5645_jack_detect_work(struct work_struct *work) +{ + struct rt5645_priv *rt5645 = + container_of(work, struct rt5645_priv, jack_detect_work.work); + + rt5645_jack_detect(rt5645->codec, rt5645->jack); +} + static irqreturn_t rt5645_irq(int irq, void *data) { struct rt5645_priv *rt5645 = data; - rt5645_jack_detect(rt5645->codec, rt5645->jack); + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -2436,6 +2445,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Fail gpio_direction hp_det_gpio\n"); } + INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, rt5645_dai, ARRAY_SIZE(rt5645_dai)); } @@ -2447,6 +2458,8 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + cancel_delayed_work_sync(&rt5645->jack_detect_work); + if (gpio_is_valid(rt5645->pdata.hp_det_gpio)) gpio_free(rt5645->pdata.hp_det_gpio); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 50c62c5..5ec2520 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2168,6 +2168,7 @@ struct rt5645_priv { struct regmap *regmap; struct i2c_client *i2c; struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; int sysclk; int sysclk_src; -- cgit v1.1 From 80fff6bf65dcae62255bdb592603dfc247c8cacf Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Fri, 3 Oct 2014 14:37:24 -0700 Subject: ASoC: rt5677: Include gpio driver header The header file is needed because struct gpio_chip is placed in struct rt5677_priv. Signed-off-by: Ben Zhang Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d4eb6d5..99fd023 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -13,6 +13,7 @@ #define __RT5677_H__ #include +#include /* Info */ #define RT5677_RESET 0x00 -- cgit v1.1 From 40eb90a18e93fbd4fd0e6892b31241356c3c8e43 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Fri, 10 Oct 2014 20:46:36 -0700 Subject: ASoC: rt5677: Add option to configure gpio as floating/pullup/pulldown gpio_config is array of 6 elements that allows to set GPIO as floating, pullup, pulldown. Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5677.txt | 7 ++++ include/sound/rt5677.h | 3 ++ sound/soc/codecs/rt5677.c | 39 ++++++++++++++++++++++ 3 files changed, 49 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 0701b83..f82f0e9 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -27,6 +27,12 @@ Optional properties: Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential, rather than single-ended. +- realtek,gpio-config + Array of six 8bit elements that configures GPIO. + 0 - floating (reset value) + 1 - pull down + 2 - pull up + Pins on the device (for linking into audio routes): * IN1P @@ -56,4 +62,5 @@ rt5677 { realtek,pow-ldo2-gpio = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; realtek,in1-differential = "true"; + realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index 082670e..a56b429 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -27,6 +27,9 @@ struct rt5677_platform_data { bool lout3_diff; /* DMIC2 clock source selection */ enum rt5677_dmic2_clk dmic2_clk_pin; + + /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ + u8 gpio_config[6]; }; #endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a454df3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3309,6 +3309,38 @@ static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) return 0; } +/** Configures the gpio as + * 0 - floating + * 1 - pull down + * 2 - pull up + */ +static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, + int value) +{ + int shift; + + switch (offset) { + case RT5677_GPIO1 ... RT5677_GPIO2: + shift = 2 * (1 - offset); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL2, + 0x3 << shift, + (value & 0x3) << shift); + break; + + case RT5677_GPIO3 ... RT5677_GPIO6: + shift = 2 * (9 - offset); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL3, + 0x3 << shift, + (value & 0x3) << shift); + break; + + default: + break; + } +} + static struct gpio_chip rt5677_template_chip = { .label = "rt5677", .owner = THIS_MODULE, @@ -3353,6 +3385,7 @@ static void rt5677_free_gpio(struct i2c_client *i2c) static int rt5677_probe(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int i; rt5677->codec = codec; @@ -3371,6 +3404,9 @@ static int rt5677_probe(struct snd_soc_codec *codec) regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00); + for (i = 0; i < RT5677_GPIO_NUM; i++) + rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + return 0; } @@ -3590,6 +3626,9 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) (rt5677->pow_ldo2 != -ENOENT)) return rt5677->pow_ldo2; + of_property_read_u8_array(np, "realtek,gpio-config", + rt5677->pdata.gpio_config, RT5677_GPIO_NUM); + return 0; } -- cgit v1.1 From af48f1d08a5474184da9aaf8b77f4a2848b1875e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 6 Oct 2014 16:30:51 +0800 Subject: ASoC: rt5677: Support DSP function for VAD application The ALC5677 has a programmable DSP, and there is a SPI that is dadicated for DSP firmware loading. Therefore, the patch includes a SPI driver for writing the DSP firmware. The VAD(Voice Activaty Detection) has be implemented and use the DSP to recognize the key phase. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/rt5677-spi.c | 128 +++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 21 +++ sound/soc/codecs/rt5677.c | 288 +++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5677.h | 6 + 5 files changed, 440 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/rt5677-spi.c create mode 100644 sound/soc/codecs/rt5677-spi.h diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..4435f9f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,7 +79,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o +snd-soc-rt5677-objs := rt5677.o rt5677-spi.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c new file mode 100644 index 0000000..11c38f3 --- /dev/null +++ b/sound/soc/codecs/rt5677-spi.c @@ -0,0 +1,128 @@ +/* + * rt5677-spi.c -- RT5677 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rt5677-spi.h" + +static struct spi_device *g_spi; + +/** + * rt5677_spi_write - Write data to SPI. + * @txbuf: Data Buffer for writing. + * @len: Data length. + * + * + * Returns true for success. + */ +int rt5677_spi_write(u8 *txbuf, size_t len) +{ + int status; + + status = spi_write(g_spi, txbuf, len); + + if (status) + dev_err(&g_spi->dev, "rt5677_spi_write error %d\n", status); + + return status; +} + +/** + * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) +{ + u8 spi_cmd = RT5677_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5677_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < fw->size) { + if (offset + RT5677_SPI_BUF_LEN <= fw->size) + end = RT5677_SPI_BUF_LEN; + else + end = fw->size % RT5677_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = fw->data[offset + i + 0]; + write_buf[i + 11] = fw->data[offset + i + 1]; + write_buf[i + 10] = fw->data[offset + i + 2]; + write_buf[i + 9] = fw->data[offset + i + 3]; + write_buf[i + 8] = fw->data[offset + i + 4]; + write_buf[i + 7] = fw->data[offset + i + 5]; + write_buf[i + 6] = fw->data[offset + i + 6]; + write_buf[i + 5] = fw->data[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + rt5677_spi_write(write_buf, end + 6); + + offset += RT5677_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} + +static int rt5677_spi_probe(struct spi_device *spi) +{ + g_spi = spi; + return 0; +} + +static struct spi_driver rt5677_spi_driver = { + .driver = { + .name = "rt5677", + .owner = THIS_MODULE, + }, + .probe = rt5677_spi_probe, +}; +module_spi_driver(rt5677_spi_driver); + +MODULE_DESCRIPTION("ASoC RT5677 SPI driver"); +MODULE_AUTHOR("Oder Chiou "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h new file mode 100644 index 0000000..7528bfd --- /dev/null +++ b/sound/soc/codecs/rt5677-spi.h @@ -0,0 +1,21 @@ +/* + * rt5677-spi.h -- RT5677 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Oder Chiou + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5671_SPI_H__ +#define __RT5671_SPI_H__ + +#define RT5677_SPI_BUF_LEN 240 +#define RT5677_SPI_CMD_BURST_WRITE 0x05 + +int rt5677_spi_write(u8 *txbuf, size_t len); +int rt5677_spi_burst_write(u32 addr, const struct firmware *fw); + +#endif /* __RT5677_SPI_H__ */ diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index a454df3..e6e54fa 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -31,6 +32,7 @@ #include "rl6231.h" #include "rt5677.h" +#include "rt5677-spi.h" #define RT5677_DEVICE_ID 0x6327 @@ -537,6 +539,243 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. + * @codec: SoC audio codec device. + * @addr: Address index. + * @value: Address data. + * + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, + unsigned int addr, unsigned int value, unsigned int opcode) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&rt5677->dsp_cmd_lock); + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + addr & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + value >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + value & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + if (ret < 0) { + dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + goto err; + } + +err: + mutex_unlock(&rt5677->dsp_cmd_lock); + + return ret; +} + +/** + * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. + * @codec: SoC audio codec device. + * @addr: Address index. + * @value: Address data. + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_read_addr( + struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int msb, lsb; + + mutex_lock(&rt5677->dsp_cmd_lock); + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + addr & 0xffff); + if (ret < 0) { + dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + goto err; + } + + ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + if (ret < 0) { + dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + goto err; + } + + regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); + regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + *value = (msb << 16) | lsb; + +err: + mutex_unlock(&rt5677->dsp_cmd_lock); + + return ret; +} + +/** + * rt5677_dsp_mode_i2c_write - Write register on DSP mode. + * @codec: SoC audio codec device. + * @reg: Register index. + * @value: Register data. + * + * + * Returns 0 for success or negative error code. + */ +static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + value, 0x0001); +} + +/** + * rt5677_dsp_mode_i2c_read - Read register on DSP mode. + * @codec: SoC audio codec device. + * @reg: Register index. + * + * + * Returns Register value. + */ +static unsigned int rt5677_dsp_mode_i2c_read( + struct snd_soc_codec *codec, unsigned int reg) +{ + unsigned int value = 0; + + rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + + return value; +} + +/** + * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. + * @codec: audio codec + * @reg: register index. + * @mask: register mask + * @value: new value + * + * + * Returns 1 for change, 0 for no change, or negative error code. + */ +static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, + unsigned int reg, unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + ret = rt5677_dsp_mode_i2c_read(codec, reg); + if (ret < 0) { + dev_err(codec->dev, "Failed to read reg: %d\n", ret); + goto err; + } + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = rt5677_dsp_mode_i2c_write(codec, reg, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) +{ + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + static bool activity; + int ret; + + if (on && !activity) { + activity = true; + + regcache_cache_only(rt5677->regmap, false); + regcache_cache_bypass(rt5677->regmap, true); + + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x1); + regmap_update_bits(rt5677->regmap, + RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0f00); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_LDO1_SEL_MASK, 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, + RT5677_PWR_LDO1, RT5677_PWR_LDO1); + regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + + ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, + codec->dev); + if (ret == 0) { + rt5677_spi_burst_write(0x50000000, rt5677->fw1); + release_firmware(rt5677->fw1); + } + + ret = request_firmware(&rt5677->fw2, RT5677_FIRMWARE2, + codec->dev); + if (ret == 0) { + rt5677_spi_burst_write(0x60000000, rt5677->fw2); + release_firmware(rt5677->fw2); + } + + rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, + 0x0); + + regcache_cache_bypass(rt5677->regmap, false); + regcache_cache_only(rt5677->regmap, true); + } else if (!on && activity) { + activity = false; + + regcache_cache_only(rt5677->regmap, false); + regcache_cache_bypass(rt5677->regmap, true); + + rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, + 0x1); + rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + + regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); + + regcache_cache_bypass(rt5677->regmap, false); + regcache_mark_dirty(rt5677->regmap); + regcache_sync(rt5677->regmap); + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -556,6 +795,31 @@ static unsigned int bst_tlv[] = { 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; +static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; + + return 0; +} + +static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; + + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + rt5677_set_dsp_vad(codec, rt5677->dsp_vad_en); + + return 0; +} + static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* OUTPUT Control */ SOC_SINGLE("OUT1 Playback Switch", RT5677_LOUT1, @@ -627,6 +891,9 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2, RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + SOC_SINGLE_EXT("DSP VAD Switch", SND_SOC_NOPM, 0, 1, 0, + rt5677_dsp_vad_get, rt5677_dsp_vad_put), }; /** @@ -3181,6 +3448,8 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + rt5677_set_dsp_vad(codec, false); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, RT5677_LDO1_SEL_MASK | RT5677_LDO2_SEL_MASK, 0x0055); @@ -3214,6 +3483,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000); regmap_update_bits(rt5677->regmap, RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000); + + if (rt5677->dsp_vad_en) + rt5677_set_dsp_vad(codec, true); break; default: @@ -3407,6 +3679,8 @@ static int rt5677_probe(struct snd_soc_codec *codec) for (i = 0; i < RT5677_GPIO_NUM; i++) rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + mutex_init(&rt5677->dsp_cmd_lock); + return 0; } @@ -3426,8 +3700,11 @@ static int rt5677_suspend(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - regcache_cache_only(rt5677->regmap, true); - regcache_mark_dirty(rt5677->regmap); + if (!rt5677->dsp_vad_en) { + regcache_cache_only(rt5677->regmap, true); + regcache_mark_dirty(rt5677->regmap); + } + if (gpio_is_valid(rt5677->pow_ldo2)) gpio_set_value_cansleep(rt5677->pow_ldo2, 0); @@ -3442,8 +3719,11 @@ static int rt5677_resume(struct snd_soc_codec *codec) gpio_set_value_cansleep(rt5677->pow_ldo2, 1); msleep(10); } - regcache_cache_only(rt5677->regmap, false); - regcache_sync(rt5677->regmap); + + if (!rt5677->dsp_vad_en) { + regcache_cache_only(rt5677->regmap, false); + regcache_sync(rt5677->regmap); + } return 0; } diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 99fd023..20efa4a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1507,6 +1507,9 @@ #define RT5677_GPIO5_FUNC_GPIO (0x0 << 9) #define RT5677_GPIO5_FUNC_DMIC (0x1 << 9) +#define RT5677_FIRMWARE1 "rt5677_dsp_fw1.bin" +#define RT5677_FIRMWARE2 "rt5677_dsp_fw2.bin" + /* System Clock Source */ enum { RT5677_SCLK_S_MCLK, @@ -1546,6 +1549,8 @@ struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; struct regmap *regmap; + const struct firmware *fw1, *fw2; + struct mutex dsp_cmd_lock; int sysclk; int sysclk_src; @@ -1559,6 +1564,7 @@ struct rt5677_priv { #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif + bool dsp_vad_en; }; #endif /* __RT5677_H__ */ -- cgit v1.1 From 8a4bd60af4cbdfbdaab6dec6ab86471733197a4f Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Wed, 15 Oct 2014 13:55:32 -0700 Subject: ASoC: rt5677: Print more information if setting DAI clock failed Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index e6e54fa..caada91 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3104,7 +3104,8 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream, rt5677->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting: sysclk=%dHz lrck=%dHz\n", + rt5677->sysclk, rt5677->lrck[dai->id]); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); -- cgit v1.1 From 45b6e1d300b034678c42369aad3b27d37854d1fb Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Thu, 16 Oct 2014 09:40:58 -0700 Subject: ASoC: rt5677: fix build when kernel compiled without GPIOLIB support Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index caada91..d17d079 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3646,6 +3646,11 @@ static void rt5677_free_gpio(struct i2c_client *i2c) gpiochip_remove(&rt5677->gpio_chip); } #else +static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, + int value) +{ +} + static void rt5677_init_gpio(struct i2c_client *i2c) { } -- cgit v1.1 From ac884fc47b7750b1e7eaf04f0236610c84ceee54 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Fri, 17 Oct 2014 11:56:42 -0700 Subject: ASoC: rt5677: add build dependency to spi Since 9cb715a9d4c the codec has a hardcoded dependency to spi. Add this dependency to Kconfig. It fixes buildbot compilation failure: sound/built-in.o: In function `spi_write': >> rt5677-spi.c:(.text+0x8265f): undefined reference to `spi_sync' sound/built-in.o: In function `rt5677_spi_driver_init': >> rt5677-spi.c:(.init.text+0x17db): undefined reference to `spi_register_driver' Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..2c7482e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -85,7 +85,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5670 if I2C - select SND_SOC_RT5677 if I2C + select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC -- cgit v1.1 From 7f6d75d77683c8f9c18836a2fea2a1e76efc3a9a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 10:50:56 -0300 Subject: ASoC: sgtl5000: Cleanup the comments Fix grammar and typos. Besides that, also fix the comment about the range of SYS_MCLK, which is from 8 to 27 MHz. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..10160e7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -530,16 +530,16 @@ static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * set clock according to i2s frame clock, - * sgtl5000 provide 2 clock sources. - * 1. sys_mclk. sample freq can only configure to + * sgtl5000 provides 2 clock sources: + * 1. sys_mclk: sample freq can only be configured to * 1/256, 1/384, 1/512 of sys_mclk. - * 2. pll. can derive any audio clocks. + * 2. pll: can derive any audio clocks. * * clock setting rules: - * 1. in slave mode, only sys_mclk can use. - * 2. as constraint by sys_mclk, sample freq should - * set to 32k, 44.1k and above. - * 3. using sys_mclk prefer to pll to save power. + * 1. in slave mode, only sys_mclk can be used + * 2. as constraint by sys_mclk, sample freq should be set to 32 kHz, 44.1 kHz + * and above. + * 3. usage of sys_mclk is preferred over pll to save power. */ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) { @@ -549,8 +549,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * sample freq should be divided by frame clock, - * if frame clock lower than 44.1khz, sample feq should set to - * 32khz or 44.1khz. + * if frame clock is lower than 44.1 kHz, sample freq should be set to + * 32 kHz or 44.1 kHz. */ switch (frame_rate) { case 8000: @@ -603,7 +603,8 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) /* * calculate the divider of mclk/sample_freq, - * factor of freq =96k can only be 256, since mclk in range (12m,27m) + * factor of freq = 96 kHz can only be 256, since mclk is in the range + * of 8 MHz - 27 MHz */ switch (sgtl5000->sysclk / sys_fs) { case 256: @@ -619,7 +620,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) SGTL5000_MCLK_FREQ_SHIFT; break; default: - /* if mclk not satisify the divider, use pll */ + /* if mclk does not satisfy the divider, use pll */ if (sgtl5000->master) { clk_ctl |= SGTL5000_MCLK_FREQ_PLL << SGTL5000_MCLK_FREQ_SHIFT; @@ -795,7 +796,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) SGTL5000_LINEREG_D_POWERUP, SGTL5000_LINEREG_D_POWERUP); - /* when internal ldo enabled, simple digital power can be disabled */ + /* when internal ldo is enabled, simple digital power can be disabled */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINREG_SIMPLE_POWERUP, 0); @@ -1079,7 +1080,7 @@ static bool sgtl5000_readable(struct device *dev, unsigned int reg) /* * sgtl5000 has 3 internal power supplies: * 1. VAG, normally set to vdda/2 - * 2. chargepump, set to different value + * 2. charge pump, set to different value * according to voltage of vdda and vddio * 3. line out VAG, normally set to vddio/2 * -- cgit v1.1 From bd0593f5f6add279257334b4a76aecd3ee8d31dc Mon Sep 17 00:00:00 2001 From: Jean-Michel Hautbois Date: Tue, 14 Oct 2014 08:43:11 +0200 Subject: ASoC: sgtl5000: Add MicBias resistor support in DT Some systems may require a different resistor than the default one (4K). This adds a property in sgtl5000 codec. It keeps the default of 4K when nothing is specified so it does not break existing code. Signed-off-by: Jean-Michel Hautbois Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sgtl5000.txt | 8 ++++ sound/soc/codecs/sgtl5000.c | 55 ++++++++++++++++++++-- 2 files changed, 59 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 955df60..d6ec927 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,10 +7,18 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs + The resistor can take values of 2k, 4k or 8k. + If set to 0 it will be off. + If this node is not mentioned or if the value is unknown, then + micbias resistor is set to 4K. + + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; + sgtl5000-micbias-resistor = <1>; }; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 10160e7..c417b4a 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -121,6 +122,13 @@ struct ldo_regulator { bool enabled; }; +enum sgtl5000_micbias_resistor { + SGTL5000_MICBIAS_OFF = 0, + SGTL5000_MICBIAS_2K = 2, + SGTL5000_MICBIAS_4K = 4, + SGTL5000_MICBIAS_8K = 8, +}; + /* sgtl5000 private structure in codec */ struct sgtl5000_priv { int sysclk; /* sysclk rate */ @@ -131,6 +139,7 @@ struct sgtl5000_priv { struct regmap *regmap; struct clk *mclk; int revision; + u8 micbias_resistor; }; /* @@ -145,12 +154,14 @@ struct sgtl5000_priv { static int mic_bias_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: - /* change mic bias resistor to 4Kohm */ + /* change mic bias resistor */ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); break; case SND_SOC_DAPM_PRE_PMD: @@ -1327,7 +1338,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); /* * disable DAP @@ -1419,6 +1432,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, struct sgtl5000_priv *sgtl5000; int ret, reg, rev; unsigned int mclk; + struct device_node *np = client->dev.of_node; + u32 value; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1471,6 +1486,38 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); sgtl5000->revision = rev; + if (np) { + if (!of_property_read_u32(np, + "micbias-resistor-k-ohms", &value)) { + switch (value) { + case SGTL5000_MICBIAS_OFF: + sgtl5000->micbias_resistor = 0; + break; + case SGTL5000_MICBIAS_2K: + sgtl5000->micbias_resistor = 1; + break; + case SGTL5000_MICBIAS_4K: + sgtl5000->micbias_resistor = 2; + break; + case SGTL5000_MICBIAS_8K: + sgtl5000->micbias_resistor = 3; + break; + default: + sgtl5000->micbias_resistor = 2; + dev_err(&client->dev, + "Unsuitable MicBias resistor\n"); + } + } else { + /* default is 4Kohms */ + sgtl5000->micbias_resistor = 2; + } + dev_err(&client->dev, + "Unsuitable MicBias resistor\n"); + } + } else { + } + } + i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- cgit v1.1 From 8735779774b8bbe14456c9e6ba4525eefc67a228 Mon Sep 17 00:00:00 2001 From: Jean-Michel Hautbois Date: Tue, 14 Oct 2014 08:43:12 +0200 Subject: ASoC: sgtl5000: Add MicBias voltage support Some systems may require to specify a bias different than default (1.25V). This adds a property in sgtl5000 codec. The property is specified in milli-volts so that it is coherent with datasheet. Signed-off-by: Jean-Michel Hautbois Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 7 ++++++- sound/soc/codecs/sgtl5000.c | 13 +++++++++++++ 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index d6ec927..1aab403 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -13,6 +13,10 @@ Required properties: If this node is not mentioned or if the value is unknown, then micbias resistor is set to 4K. +- micbias-voltage-m-volts : the bias voltage to be used in mVolts + The voltage can take values from 1.25V to 3V by 250mV steps + If this node is not mentionned or the value is unknown, then + the value is set to 1.25V. Example: @@ -20,5 +24,6 @@ codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 150>; - sgtl5000-micbias-resistor = <1>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; }; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c417b4a..59336f6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -140,6 +140,7 @@ struct sgtl5000_priv { struct clk *mclk; int revision; u8 micbias_resistor; + u8 micbias_voltage; }; /* @@ -1342,6 +1343,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_BIAS_R_MASK, sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); + snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_MASK, + sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); /* * disable DAP * TODO: @@ -1511,10 +1515,19 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* default is 4Kohms */ sgtl5000->micbias_resistor = 2; } + if (!of_property_read_u32(np, + "micbias-voltage-m-volts", &value)) { + /* 1250mV => 0 */ + /* steps of 250mV */ + if ((value >= 1250) && (value <= 3000)) + sgtl5000->micbias_voltage = (value / 250) - 5; + else { + sgtl5000->micbias_voltage = 0; dev_err(&client->dev, "Unsuitable MicBias resistor\n"); } } else { + sgtl5000->micbias_voltage = 0; } } -- cgit v1.1 From 84d4cbe9a60a1fdd35b4dd69951f31c518b467d8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Oct 2014 15:29:58 -0300 Subject: ASoC: simple-card: Delete unneeded test before of_node_put of_node_put() supports NULL as its argument, so the initial test is not necessary. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d1b7293..4f192ee 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -368,12 +368,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_link->cpu_dai_name = NULL; dai_link_of_err: - if (np) - of_node_put(np); - if (bitclkmaster) - of_node_put(bitclkmaster); - if (framemaster) - of_node_put(framemaster); + of_node_put(np); + of_node_put(bitclkmaster); + of_node_put(framemaster); return ret; } @@ -464,11 +461,9 @@ static int asoc_simple_card_unref(struct platform_device *pdev) num_links < card->num_links; num_links++, dai_link++) { np = (struct device_node *) dai_link->cpu_of_node; - if (np) - of_node_put(np); + of_node_put(np); np = (struct device_node *) dai_link->codec_of_node; - if (np) - of_node_put(np); + of_node_put(np); } return 0; } -- cgit v1.1 From 03ad6a8c93b6df2d65c305b5b5f9474068b45bfb Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Oct 2014 15:33:45 +0200 Subject: ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs In the scenario where there is one "Line Out", one "Speaker" and one "Headphone", and there are only two DACs, two outputs will share a DAC. Currently any mixer on such a DAC will get the "PCM" name, which is misleading. Instead use "Headphone+LO" or "Speaker+LO" to better specify what the volume actually controls. [fixed missing slave string additions by tiwai] Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 64220c0..dc13cce 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1038,6 +1038,19 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, break; *index = ch; return "Headphone"; + case AUTO_PIN_LINE_OUT: + /* This deals with the case where we have two DACs and + * one LO, one HP and one Speaker */ + if (!ch && cfg->speaker_outs && cfg->hp_outs) { + bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); + bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + if (hp_lo_shared && spk_lo_shared) + return spec->vmaster_mute.hook ? "PCM" : "Master"; + if (hp_lo_shared) + return "Headphone+LO"; + if (spk_lo_shared) + return "Speaker+LO"; + } } /* for a single channel output, we don't have to name the channel */ @@ -4524,7 +4537,7 @@ static const char * const slave_pfxs[] = { "CLFE", "Bass Speaker", "PCM", "Speaker Front", "Speaker Surround", "Speaker CLFE", "Speaker Side", "Headphone Front", "Headphone Surround", "Headphone CLFE", - "Headphone Side", + "Headphone Side", "Headphone+LO", "Speaker+LO", NULL, }; -- cgit v1.1 From 3abb4f4d0e7aaad0d12004b5057f4486a688752b Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Oct 2014 15:33:46 +0200 Subject: ALSA: hda - Use "Line Out" name instead of "PCM" when there are other outputs In case there are speakers or headphones as well, anything that only covers the line out should not be labelled "PCM". Let's name it "Line Out" instead for clarity. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dc13cce..06d7210 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1055,7 +1055,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, /* for a single channel output, we don't have to name the channel */ if (cfg->line_outs == 1 && !spec->multi_ios) - return "PCM"; + return "Line Out"; if (ch >= ARRAY_SIZE(channel_name)) { snd_BUG(); -- cgit v1.1 From 8e648204194cb51df8562d1e2a64b7dc6b0aead3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Oct 2014 14:38:09 +0200 Subject: ALSA: Update control names documentation This document was not really up-to-date. Add recent additions to this standard - based on what the HDA driver currently does, which is some kind of a de facto standard. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ControlNames.txt | 32 ++++++++++++++++++++++++++----- 1 file changed, 27 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index fea65bb..79a6127 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -1,6 +1,6 @@ This document describes standard names of mixer controls. -Syntax: SOURCE [DIRECTION] FUNCTION +Syntax: [LOCATION] SOURCE [CHANNEL] [DIRECTION] FUNCTION DIRECTION: (both directions) @@ -14,12 +14,29 @@ FUNCTION: Volume Route (route control, hardware specific) +CHANNEL: + (channel independent, or applies to all channels) + Front + Surround (rear left/right in 4.0/5.1 surround) + CLFE + Center + LFE + Side (side left/right for 7.1 surround) + +LOCATION: (physical location of source) + Front + Rear + Dock (docking station) + Internal + SOURCE: Master Master Mono Hardware Master Speaker (internal speaker) + Bass Speaker (internal LFE speaker) Headphone + Line Out Beep (beep generator) Phone Phone Input @@ -27,14 +44,14 @@ SOURCE: Synth FM Mic - Line + Headset Mic (mic part of combined headset jack - 4-pin headphone + mic) + Headphone Mic (mic part of either/or - 3-pin headphone or mic) + Line (input only, use "Line Out" for output) CD Video Zoom Video Aux PCM - PCM Front - PCM Rear PCM Pan Loopback Analog Loopback (D/A -> A/D loopback) @@ -47,8 +64,13 @@ SOURCE: Music I2S IEC958 + HDMI + SPDIF (output only) + SPDIF In + Digital In + HDMI/DP (either HDMI or DisplayPort) -Exceptions: +Exceptions (deprecated): [Digital] Capture Source [Digital] Capture Switch (aka input gain switch) [Digital] Capture Volume (aka input gain volume) -- cgit v1.1 From be93709cb13a1947fec3493267d04cd87baf497e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:06:04 +0200 Subject: ALSA: doc: Recommend the use of snd_ctl_elem_info() Instead of the open code for the info call back of enum elements, recommend the use of snd_ctl_elem_info(), which will reduce lots of codes. Signed-off-by: Takashi Iwai --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 784793d..84ef6a9 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -3658,6 +3658,29 @@ struct _snd_pcm_runtime { + The above callback can be simplified with a helper function, + snd_ctl_enum_info. The final code + looks like below. + (You can pass ARRAY_SIZE(texts) instead of 4 in the third + argument; it's a matter of taste.) + + + + + + + + + Some common info callbacks are available for your convenience: snd_ctl_boolean_mono_info() and snd_ctl_boolean_stereo_info(). -- cgit v1.1 From df803e1389716bcdf11932fff47d7f1fc198bc8a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:07:21 +0200 Subject: ALSA: control: Warn if too long string is passed to snd_ctl_enum_info() This allows us to catch the bugs in drivers easily. Signed-off-by: Takashi Iwai --- sound/core/control.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index b961134..f95df84 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1747,6 +1747,9 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, info->value.enumerated.items = items; if (info->value.enumerated.item >= items) info->value.enumerated.item = items - 1; + WARN(strlen(names[info->value.enumerated.item]) >= sizeof(info->value.enumerated.name), + "ALSA: too long item name '%s'\n", + names[info->value.enumerated.item]); strlcpy(info->value.enumerated.name, names[info->value.enumerated.item], sizeof(info->value.enumerated.name)); -- cgit v1.1 From a7e6fb99150ebb2852ebd0e7bad9ce37cc9a79dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:08:50 +0200 Subject: ALSA: control: Allow to pass items zero to snd_ctl_enum_info() Although this is weird, some drivers want to allow empty control elements intentionally, e.g. the number of items may change depending on the firmware status. Let the function simply returning in such a case. Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index f95df84..5c35bba 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1745,6 +1745,8 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels, info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = channels; info->value.enumerated.items = items; + if (!items) + return 0; if (info->value.enumerated.item >= items) info->value.enumerated.item = items - 1; WARN(strlen(names[info->value.enumerated.item]) >= sizeof(info->value.enumerated.name), -- cgit v1.1 From 04eeb606a8383b306f4bc6991da8231b5f3924b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:10:39 +0200 Subject: ALSA: aoa: Use snd_ctl_elem_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/aoa/codecs/onyx.c | 8 +------- sound/aoa/codecs/tas.c | 10 ++-------- 2 files changed, 3 insertions(+), 15 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 401107b..23c371ec 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -243,13 +243,7 @@ static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol, { static const char * const texts[] = { "Line-In", "Microphone" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol, diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index cf3c630..364c7c4 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -478,15 +478,9 @@ static struct snd_kcontrol_new drc_switch_control = { static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Line-In", "Microphone" }; + static const char * const texts[] = { "Line-In", "Microphone" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 6d416f594bf9a290406d267e2627c5286f51ea59 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:11:28 +0200 Subject: ALSA: mts64: Use snd_ctl_elem_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/drivers/mts64.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index f5fd448..0388fbb 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -604,21 +604,11 @@ static struct snd_kcontrol_new mts64_ctl_smpte_time_frames = { static int snd_mts64_ctl_smpte_fps_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { "24", - "25", - "29.97", - "30D", - "30" }; + static const char * const texts[5] = { + "24", "25", "29.97", "30D", "30" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item > 4) - uinfo->value.enumerated.item = 4; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } static int snd_mts64_ctl_smpte_fps_get(struct snd_kcontrol *kctl, -- cgit v1.1 From 7f471fd40742a5d87d887375430bf40331cbbcf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:11:58 +0200 Subject: ALSA: vx: Use snd_ctl_elem_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_mixer.c | 35 ++++++++--------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index 3b6823f..be9477e 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -471,30 +471,18 @@ static struct snd_kcontrol_new vx_control_output_level = { */ static int vx_audio_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts_mic[3] = { + static const char * const texts_mic[3] = { "Digital", "Line", "Mic" }; - static char *texts_vx2[2] = { + static const char * const texts_vx2[2] = { "Digital", "Analog" }; struct vx_core *chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - if (chip->type >= VX_TYPE_VXPOCKET) { - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts_mic[uinfo->value.enumerated.item]); - } else { - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts_vx2[uinfo->value.enumerated.item]); - } - return 0; + if (chip->type >= VX_TYPE_VXPOCKET) + return snd_ctl_enum_info(uinfo, 1, 3, texts_mic); + else + return snd_ctl_enum_info(uinfo, 1, 2, texts_vx2); } static int vx_audio_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -539,18 +527,11 @@ static struct snd_kcontrol_new vx_control_audio_src = { */ static int vx_clock_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char * const texts[3] = { "Auto", "Internal", "External" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int vx_clock_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From e7ced4137d859c576130ce7605e5fdd13221793d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Oct 2014 08:26:10 +0200 Subject: ALSA: bebob: More constify text arrays Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob.h | 4 ++-- sound/firewire/bebob/bebob_focusrite.c | 10 +++++----- sound/firewire/bebob/bebob_maudio.c | 18 +++++++++--------- sound/firewire/bebob/bebob_terratec.c | 4 ++-- sound/firewire/bebob/bebob_yamaha.c | 2 +- 5 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index e13eef9..dfbcd23 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -52,7 +52,7 @@ extern const unsigned int snd_bebob_rate_table[SND_BEBOB_STRM_FMT_ENTRIES]; #define SND_BEBOB_CLOCK_INTERNAL "Internal" struct snd_bebob_clock_spec { unsigned int num; - char *const *labels; + const char *const *labels; int (*get)(struct snd_bebob *bebob, unsigned int *id); }; struct snd_bebob_rate_spec { @@ -61,7 +61,7 @@ struct snd_bebob_rate_spec { }; struct snd_bebob_meter_spec { unsigned int num; - char *const *labels; + const char *const *labels; int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size); }; struct snd_bebob_spec { diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index 45a0eed..a45a869 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -101,11 +101,11 @@ saffire_write_quad(struct snd_bebob *bebob, u64 offset, u32 value) &data, sizeof(__be32), 0); } -static char *const saffirepro_26_clk_src_labels[] = { +static const char *const saffirepro_26_clk_src_labels[] = { SND_BEBOB_CLOCK_INTERNAL, "S/PDIF", "ADAT1", "ADAT2", "Word Clock" }; -static char *const saffirepro_10_clk_src_labels[] = { +static const char *const saffirepro_10_clk_src_labels[] = { SND_BEBOB_CLOCK_INTERNAL, "S/PDIF", "Word Clock" }; static int @@ -161,7 +161,7 @@ end: } struct snd_bebob_spec saffire_le_spec; -static char *const saffire_both_clk_src_labels[] = { +static const char *const saffire_both_clk_src_labels[] = { SND_BEBOB_CLOCK_INTERNAL, "S/PDIF" }; static int @@ -176,12 +176,12 @@ saffire_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) return err; }; -static char *const saffire_le_meter_labels[] = { +static const char *const saffire_le_meter_labels[] = { ANA_IN, ANA_IN, DIG_IN, ANA_OUT, ANA_OUT, ANA_OUT, ANA_OUT, STM_IN, STM_IN }; -static char *const saffire_meter_labels[] = { +static const char *const saffire_meter_labels[] = { ANA_IN, ANA_IN, STM_IN, STM_IN, STM_IN, STM_IN, STM_IN, }; diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 70faa3a..34cba80 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -340,7 +340,7 @@ end: } /* Clock source control for special firmware */ -static char *const special_clk_labels[] = { +static const char *const special_clk_labels[] = { SND_BEBOB_CLOCK_INTERNAL " with Digital Mute", "Digital", "Word Clock", SND_BEBOB_CLOCK_INTERNAL}; static int special_clk_get(struct snd_bebob *bebob, unsigned int *id) @@ -438,7 +438,7 @@ static struct snd_kcontrol_new special_sync_ctl = { }; /* Digital input interface control for special firmware */ -static char *const special_dig_in_iface_labels[] = { +static const char *const special_dig_in_iface_labels[] = { "S/PDIF Optical", "S/PDIF Coaxial", "ADAT Optical" }; static int special_dig_in_iface_ctl_info(struct snd_kcontrol *kctl, @@ -539,7 +539,7 @@ static struct snd_kcontrol_new special_dig_in_iface_ctl = { }; /* Digital output interface control for special firmware */ -static char *const special_dig_out_iface_labels[] = { +static const char *const special_dig_out_iface_labels[] = { "S/PDIF Optical and Coaxial", "ADAT Optical" }; static int special_dig_out_iface_ctl_info(struct snd_kcontrol *kctl, @@ -631,7 +631,7 @@ end: } /* Hardware metering for special firmware */ -static char *const special_meter_labels[] = { +static const char *const special_meter_labels[] = { ANA_IN, ANA_IN, ANA_IN, ANA_IN, SPDIF_IN, ADAT_IN, ADAT_IN, ADAT_IN, ADAT_IN, @@ -671,30 +671,30 @@ end: } /* last 4 bytes are omitted because it's clock info. */ -static char *const fw410_meter_labels[] = { +static const char *const fw410_meter_labels[] = { ANA_IN, DIG_IN, ANA_OUT, ANA_OUT, ANA_OUT, ANA_OUT, DIG_OUT, HP_OUT }; -static char *const audiophile_meter_labels[] = { +static const char *const audiophile_meter_labels[] = { ANA_IN, DIG_IN, ANA_OUT, ANA_OUT, DIG_OUT, HP_OUT, AUX_OUT, }; -static char *const solo_meter_labels[] = { +static const char *const solo_meter_labels[] = { ANA_IN, DIG_IN, STRM_IN, STRM_IN, ANA_OUT, DIG_OUT }; /* no clock info */ -static char *const ozonic_meter_labels[] = { +static const char *const ozonic_meter_labels[] = { ANA_IN, ANA_IN, STRM_IN, STRM_IN, ANA_OUT, ANA_OUT }; /* TODO: need testers. these positions are based on authour's assumption */ -static char *const nrv10_meter_labels[] = { +static const char *const nrv10_meter_labels[] = { ANA_IN, ANA_IN, ANA_IN, ANA_IN, DIG_IN, ANA_OUT, ANA_OUT, ANA_OUT, ANA_OUT, diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 0e4c0bf..83b6772 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -8,7 +8,7 @@ #include "./bebob.h" -static char *const phase88_rack_clk_src_labels[] = { +static const char *const phase88_rack_clk_src_labels[] = { SND_BEBOB_CLOCK_INTERNAL, "Digital In", "Word Clock" }; static int @@ -29,7 +29,7 @@ end: return err; } -static char *const phase24_series_clk_src_labels[] = { +static const char *const phase24_series_clk_src_labels[] = { SND_BEBOB_CLOCK_INTERNAL, "Digital In" }; static int diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c index 9b7e798..ef1fe38 100644 --- a/sound/firewire/bebob/bebob_yamaha.c +++ b/sound/firewire/bebob/bebob_yamaha.c @@ -28,7 +28,7 @@ * reccomend users to close ffado-mixer at 192.0kHz if mixer is needless. */ -static char *const clk_src_labels[] = {SND_BEBOB_CLOCK_INTERNAL, "SPDIF"}; +static const char *const clk_src_labels[] = {SND_BEBOB_CLOCK_INTERNAL, "SPDIF"}; static int clk_src_get(struct snd_bebob *bebob, unsigned int *id) { -- cgit v1.1 From 41be5164ea09c92d551e8007d2543418e40f847a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:12:26 +0200 Subject: ALSA: bebob: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_maudio.c | 41 ++++++++----------------------------- 1 file changed, 8 insertions(+), 33 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index 34cba80..a422aaa 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -352,17 +352,8 @@ static int special_clk_get(struct snd_bebob *bebob, unsigned int *id) static int special_clk_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *einf) { - einf->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - einf->count = 1; - einf->value.enumerated.items = ARRAY_SIZE(special_clk_labels); - - if (einf->value.enumerated.item >= einf->value.enumerated.items) - einf->value.enumerated.item = einf->value.enumerated.items - 1; - - strcpy(einf->value.enumerated.name, - special_clk_labels[einf->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(einf, 1, ARRAY_SIZE(special_clk_labels), + special_clk_labels); } static int special_clk_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *uval) @@ -444,17 +435,9 @@ static const char *const special_dig_in_iface_labels[] = { static int special_dig_in_iface_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *einf) { - einf->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - einf->count = 1; - einf->value.enumerated.items = ARRAY_SIZE(special_dig_in_iface_labels); - - if (einf->value.enumerated.item >= einf->value.enumerated.items) - einf->value.enumerated.item = einf->value.enumerated.items - 1; - - strcpy(einf->value.enumerated.name, - special_dig_in_iface_labels[einf->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(einf, 1, + ARRAY_SIZE(special_dig_in_iface_labels), + special_dig_in_iface_labels); } static int special_dig_in_iface_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *uval) @@ -545,17 +528,9 @@ static const char *const special_dig_out_iface_labels[] = { static int special_dig_out_iface_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *einf) { - einf->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - einf->count = 1; - einf->value.enumerated.items = ARRAY_SIZE(special_dig_out_iface_labels); - - if (einf->value.enumerated.item >= einf->value.enumerated.items) - einf->value.enumerated.item = einf->value.enumerated.items - 1; - - strcpy(einf->value.enumerated.name, - special_dig_out_iface_labels[einf->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(einf, 1, + ARRAY_SIZE(special_dig_out_iface_labels), + special_dig_out_iface_labels); } static int special_dig_out_iface_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *uval) -- cgit v1.1 From 609e478b40aceaa07d14f1bada02a3874bac2c45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:13:01 +0200 Subject: ALSA: ak4xxx-adda: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4xxx-adda.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index f3735e6..67dbfde 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -465,17 +465,10 @@ static int snd_akm4xxx_stereo_volume_put(struct snd_kcontrol *kcontrol, static int snd_akm4xxx_deemphasis_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { + static const char * const texts[4] = { "44.1kHz", "Off", "48kHz", "32kHz", }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_akm4xxx_deemphasis_get(struct snd_kcontrol *kcontrol, @@ -570,22 +563,13 @@ static int ak4xxx_capture_source_info(struct snd_kcontrol *kcontrol, { struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); int mixer_ch = AK_GET_SHIFT(kcontrol->private_value); - const char **input_names; - unsigned int num_names, idx; + unsigned int num_names; num_names = ak4xxx_capture_num_inputs(ak, mixer_ch); if (!num_names) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_names; - idx = uinfo->value.enumerated.item; - if (idx >= num_names) - return -EINVAL; - input_names = ak->adc_info[mixer_ch].input_names; - strlcpy(uinfo->value.enumerated.name, input_names[idx], - sizeof(uinfo->value.enumerated.name)); - return 0; + return snd_ctl_enum_info(uinfo, 1, num_names, + ak->adc_info[mixer_ch].input_names); } static int ak4xxx_capture_source_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 1da0c47779840d038583ea6fa4e3497939d6ea21 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:13:39 +0200 Subject: ALSA: ad1816a: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a_lib.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index f0fd98e..01a0798 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -731,18 +731,12 @@ int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, static int snd_ad1816a_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[8] = { + static const char * const texts[8] = { "Line", "Mix", "CD", "Synth", "Video", "Mic", "Phone", }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 7; - if (uinfo->value.enumerated.item > 6) - uinfo->value.enumerated.item = 6; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 2, 7, texts); } static int snd_ad1816a_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 69b0c762cfd4c5e86a5b1fc0074889881b859c4a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:14:07 +0200 Subject: ALSA: es1688: Use snd_ctl_enum_info() ... and reduce the open codes. Also correct the array size and add missing const. Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688_lib.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index b3b4f15..de810e4 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -762,18 +762,12 @@ int snd_es1688_pcm(struct snd_card *card, struct snd_es1688 *chip, static int snd_es1688_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[9] = { + static const char * const texts[8] = { "Mic", "Mic Master", "CD", "AOUT", "Mic1", "Mix", "Line", "Master" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 8, texts); } static int snd_es1688_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 4e28350a8c54a8b873b610311eb384a619c4045b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:14:24 +0200 Subject: ALSA: es18xx: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..63e7323 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -964,44 +964,28 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts5Source[5] = { + static const char * const texts5Source[5] = { "Mic", "CD", "Line", "Master", "Mix" }; - static char *texts8Source[8] = { + static const char * const texts8Source[8] = { "Mic", "Mic Master", "CD", "AOUT", "Mic1", "Mix", "Line", "Master" }; struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; switch (chip->version) { case 0x1868: case 0x1878: - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts5Source[uinfo->value.enumerated.item]); - break; + return snd_ctl_enum_info(uinfo, 1, 4, texts5Source); case 0x1887: case 0x1888: - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item > 4) - uinfo->value.enumerated.item = 4; - strcpy(uinfo->value.enumerated.name, texts5Source[uinfo->value.enumerated.item]); - break; + return snd_ctl_enum_info(uinfo, 1, 5, texts5Source); case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */ case 0x1879: - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts8Source[uinfo->value.enumerated.item]); - break; + return snd_ctl_enum_info(uinfo, 1, 8, texts8Source); default: return -EINVAL; } - return 0; } static int snd_es18xx_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 8edd2d120b6821d6eba9fedd0b17b7a47fbe9181 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:14:41 +0200 Subject: ALSA: msnd: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle_mixer.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 031dc69..17e49a0 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -55,20 +55,13 @@ static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char * const texts[3] = { "Analog", "MASS", "SPDIF", }; struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = items; - if (uinfo->value.enumerated.item >= items) - uinfo->value.enumerated.item = items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, items, texts); } static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 6b9e1288a506b4d10c48a67058532c52c5c46240 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:14:51 +0200 Subject: ALSA: sb16: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/isa/sb/sb16_main.c | 10 ++-------- sound/isa/sb/sb_mixer.c | 31 ++++++------------------------- 2 files changed, 8 insertions(+), 33 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 0bbcd47..72b10f4 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -702,17 +702,11 @@ static int snd_sb16_get_dma_mode(struct snd_sb *chip) static int snd_sb16_dma_control_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char * const texts[3] = { "Auto", "Playback", "Capture" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_sb16_dma_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 1ff78ec..e403334 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -182,17 +182,11 @@ static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[5] = { + static const char * const texts[5] = { "CD", "Mic", "Line", "Synth", "Master" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item > 4) - uinfo->value.enumerated.item = 4; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } static int snd_dt019x_input_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -275,18 +269,11 @@ static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl static int snd_als4k_mono_capture_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[3] = { + static const char * const texts[3] = { "L chan only", "R chan only", "L ch/2 + R ch/2" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_als4k_mono_capture_route_get(struct snd_kcontrol *kcontrol, @@ -335,17 +322,11 @@ static int snd_als4k_mono_capture_route_put(struct snd_kcontrol *kcontrol, static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[3] = { + static const char * const texts[3] = { "Mic", "CD", "Line" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } -- cgit v1.1 From 0773efa53260a57d6e3bc38469c216635700537e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:15:06 +0200 Subject: ALSA: wss: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 360b08b..347bb1b 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1993,25 +1993,20 @@ EXPORT_SYMBOL(snd_wss_timer); static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { + static const char * const texts[4] = { "Line", "Aux", "Mic", "Mix" }; - static char *opl3sa_texts[4] = { + static const char * const opl3sa_texts[4] = { "Line", "CD", "Mic", "Mix" }; - static char *gusmax_texts[4] = { + static const char * const gusmax_texts[4] = { "Line", "Synth", "Mic", "Mix" }; - char **ptexts = texts; + const char * const *ptexts = texts; struct snd_wss *chip = snd_kcontrol_chip(kcontrol); if (snd_BUG_ON(!chip->card)) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; if (!strcmp(chip->card->driver, "GUS MAX")) ptexts = gusmax_texts; switch (chip->hardware) { @@ -2023,8 +2018,7 @@ static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, ptexts = opl3sa_texts; break; } - strcpy(uinfo->value.enumerated.name, ptexts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 2, 4, ptexts); } static int snd_wss_get_mux(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 2a2085ab198439c8f08be2f6b9b5cbc9e93877b7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:15:16 +0200 Subject: ALSA: mips: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 04bb06c..33b08fc 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -201,17 +201,10 @@ static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[3] = { + static const char * const texts[3] = { "Cam Mic", "Mic", "Line" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 3c6a73cc6b6ccd9188b3405c744365c0874b9274 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:15:26 +0200 Subject: ALSA: parisc: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/parisc/harmony.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 4b20be7..29604a2 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -776,15 +776,9 @@ static int snd_harmony_captureroute_info(struct snd_kcontrol *kc, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "Line", "Mic" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + static const char * const texts[2] = { "Line", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int -- cgit v1.1 From 3b7a00dc9e4277d6fcad68dd1db35f77264ede5f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:15:36 +0200 Subject: ALSA: ac97: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 10 +-- sound/pci/ac97/ac97_patch.c | 176 +++++++++++++------------------------------- 2 files changed, 53 insertions(+), 133 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 14ad54b..5ee2f17 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -463,14 +463,8 @@ static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, { struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->mask; - - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; - strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, + e->mask, e->texts); } static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 9917622..50f420d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -81,22 +81,11 @@ static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsi /* * shared line-in/mic controls */ -static int ac97_enum_text_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo, - const char **texts, unsigned int nums) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = nums; - if (uinfo->value.enumerated.item > nums - 1) - uinfo->value.enumerated.item = nums - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - static int ac97_surround_jack_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[] = { "Shared", "Independent" }; - return ac97_enum_text_info(kcontrol, uinfo, texts, 2); + static const char * const texts[] = { "Shared", "Independent" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int ac97_surround_jack_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -123,9 +112,9 @@ static int ac97_surround_jack_mode_put(struct snd_kcontrol *kcontrol, struct snd static int ac97_channel_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[] = { "2ch", "4ch", "6ch", "8ch" }; - return ac97_enum_text_info(kcontrol, uinfo, texts, - kcontrol->private_value); + static const char * const texts[] = { "2ch", "4ch", "6ch", "8ch" }; + + return snd_ctl_enum_info(uinfo, 1, kcontrol->private_value, texts); } static int ac97_channel_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -240,17 +229,11 @@ static inline int alc850_is_aux_back_surround(struct snd_ac97 *ac97) static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char * const texts[3] = { "Standard", "Small", "Smaller" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol, @@ -293,15 +276,9 @@ static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker = static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "AC-Link", "A/D Converter" }; + static const char * const texts[2] = { "AC-Link", "A/D Converter" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol, @@ -401,15 +378,9 @@ static int patch_yamaha_ymf743(struct snd_ac97 *ac97) There is also a bit to mute S/PDIF output in a vendor-specific register. */ static int snd_ac97_ymf753_spdif_output_pin_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { "Disabled", "Pin 43", "Pin 48" }; + static const char * const texts[3] = { "Disabled", "Pin 43", "Pin 48" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_ac97_ymf753_spdif_output_pin_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1103,16 +1074,11 @@ static int patch_sigmatel_stac9756(struct snd_ac97 * ac97) static int snd_ac97_stac9758_output_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { "Input/Disabled", "Front Output", + static const char * const texts[5] = { + "Input/Disabled", "Front Output", "Rear Output", "Center/LFE Output", "Mixer Output" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item > 4) - uinfo->value.enumerated.item = 4; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } static int snd_ac97_stac9758_output_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1147,16 +1113,11 @@ static int snd_ac97_stac9758_output_jack_put(struct snd_kcontrol *kcontrol, stru static int snd_ac97_stac9758_input_jack_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[7] = { "Mic2 Jack", "Mic1 Jack", "Line In Jack", + static const char * const texts[7] = { + "Mic2 Jack", "Mic1 Jack", "Line In Jack", "Front Jack", "Rear Jack", "Center/LFE Jack", "Mute" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 7; - if (uinfo->value.enumerated.item > 6) - uinfo->value.enumerated.item = 6; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 7, texts); } static int snd_ac97_stac9758_input_jack_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1181,15 +1142,11 @@ static int snd_ac97_stac9758_input_jack_put(struct snd_kcontrol *kcontrol, struc static int snd_ac97_stac9758_phonesel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { "None", "Front Jack", "Rear Jack" }; + static const char * const texts[3] = { + "None", "Front Jack", "Rear Jack" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_ac97_stac9758_phonesel_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1804,15 +1761,9 @@ static int patch_ad1886(struct snd_ac97 * ac97) static int snd_ac97_ad198x_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "AC-Link", "A/D Converter" }; + static const char * const texts[2] = { "AC-Link", "A/D Converter" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ac97_ad198x_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1994,15 +1945,9 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd static int snd_ac97_ad1888_downmix_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = {"Off", "6 -> 4", "6 -> 2"}; + static const char * const texts[3] = {"Off", "6 -> 4", "6 -> 2"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_ac97_ad1888_downmix_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2153,16 +2098,11 @@ static int patch_ad1980(struct snd_ac97 * ac97) static int snd_ac97_ad1985_vrefout_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = {"High-Z", "3.7 V", "2.25 V", "0 V"}; + static const char * const texts[4] = { + "High-Z", "3.7 V", "2.25 V", "0 V" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_ac97_ad1985_vrefout_get(struct snd_kcontrol *kcontrol, @@ -2756,20 +2696,18 @@ static const struct snd_kcontrol_new snd_ac97_controls_alc655[] = { static int alc655_iec958_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts_655[3] = { "PCM", "Analog In", "IEC958 In" }; - static char *texts_658[4] = { "PCM", "Analog1 In", "Analog2 In", "IEC958 In" }; + static const char * const texts_655[3] = { + "PCM", "Analog In", "IEC958 In" + }; + static const char * const texts_658[4] = { + "PCM", "Analog1 In", "Analog2 In", "IEC958 In" + }; struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ac97->spec.dev_flags ? 4 : 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - ac97->spec.dev_flags ? - texts_658[uinfo->value.enumerated.item] : - texts_655[uinfo->value.enumerated.item]); - return 0; + if (ac97->spec.dev_flags) + return snd_ctl_enum_info(uinfo, 1, 4, texts_658); + else + return snd_ctl_enum_info(uinfo, 1, 3, texts_655); } static int alc655_iec958_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -3055,15 +2993,9 @@ static int patch_cm9738(struct snd_ac97 * ac97) static int snd_ac97_cmedia_spdif_playback_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Analog", "Digital" }; + static const char * const texts[] = { "Analog", "Digital" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ac97_cmedia_spdif_playback_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -3235,15 +3167,9 @@ static const struct snd_kcontrol_new snd_ac97_cm9761_controls[] = { static int cm9761_spdif_out_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "AC-Link", "ADC", "SPDIF-In" }; + static const char * const texts[] = { "AC-Link", "ADC", "SPDIF-In" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int cm9761_spdif_out_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -3552,11 +3478,12 @@ static int snd_ac97_vt1617a_smart51_info(struct snd_kcontrol *kcontrol, * is SM51EN *AND* it's Bit14, not Bit15 so the table is very * counter-intuitive */ - static const char* texts[] = { "LineIn Mic1", "LineIn Mic1 Mic3", + static const char * const texts[] = {"LineIn Mic1", "LineIn Mic1 Mic3", "Surr LFE/C Mic3", "LineIn LFE/C Mic3", "LineIn Mic2", "LineIn Mic2 Mic1", "Surr LFE Mic1", "Surr LFE Mic1 Mic2"}; - return ac97_enum_text_info(kcontrol, uinfo, texts, 8); + + return snd_ctl_enum_info(uinfo, 1, 8, texts); } static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, @@ -3720,9 +3647,8 @@ static struct vt1618_uaj_item vt1618_uaj[3] = { static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - return ac97_enum_text_info(kcontrol, uinfo, - vt1618_uaj[kcontrol->private_value].items, - 4); + return snd_ctl_enum_info(uinfo, 1, 4, + vt1618_uaj[kcontrol->private_value].items); } /* All of the vt1618 Universal Audio Jack twiddlers are on @@ -3767,9 +3693,9 @@ static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *txt_aux[] = {"Aux In", "Back Surr Out"}; + static const char * const txt_aux[] = {"Aux In", "Back Surr Out"}; - return ac97_enum_text_info(kcontrol, uinfo, txt_aux, 2); + return snd_ctl_enum_info(uinfo, 1, 2, txt_aux); } static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 1bc10bb68d348078af0eb8b64292ec542dcd7634 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Oct 2014 08:51:45 +0200 Subject: ALSA: ac97: Constify more text arrays Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 22 ++++++++++++++-------- sound/pci/ac97/ac97_patch.h | 2 +- 2 files changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 50f420d..ceaac1c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -33,7 +33,8 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, const char *name); static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves); + const unsigned int *tlv, + const char * const *slaves); /* * Chip specific initialization @@ -3196,7 +3197,9 @@ static int cm9761_spdif_out_source_put(struct snd_kcontrol *kcontrol, struct snd ucontrol->value.enumerated.item[0] == 1 ? 0x2 : 0); } -static const char *cm9761_dac_clock[] = { "AC-Link", "SPDIF-In", "Both" }; +static const char * const cm9761_dac_clock[] = { + "AC-Link", "SPDIF-In", "Both" +}; static const struct ac97_enum cm9761_dac_clock_enum = AC97_ENUM_SINGLE(AC97_CM9761_SPDIF_CTRL, 9, 3, cm9761_dac_clock); @@ -3310,7 +3313,9 @@ static int patch_cm9761(struct snd_ac97 *ac97) #define AC97_CM9780_MULTI_CHAN 0x66 #define AC97_CM9780_SPDIF 0x6c -static const char *cm9780_ch_select[] = { "Front", "Side", "Center/LFE", "Rear" }; +static const char * const cm9780_ch_select[] = { + "Front", "Side", "Center/LFE", "Rear" +}; static const struct ac97_enum cm9780_ch_select_enum = AC97_ENUM_SINGLE(AC97_CM9780_MULTI_CHAN, 6, 4, cm9780_ch_select); static const struct snd_kcontrol_new cm9780_controls[] = { @@ -3356,7 +3361,7 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0), AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0), }; -static const char *slave_vols_vt1616[] = { +static const char * const slave_vols_vt1616[] = { "Front Playback Volume", "Surround Playback Volume", "Center Playback Volume", @@ -3364,7 +3369,7 @@ static const char *slave_vols_vt1616[] = { NULL }; -static const char *slave_sws_vt1616[] = { +static const char * const slave_sws_vt1616[] = { "Front Playback Switch", "Surround Playback Switch", "Center Playback Switch", @@ -3385,10 +3390,11 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, /* create a virtual master control and add slaves */ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, - const unsigned int *tlv, const char **slaves) + const unsigned int *tlv, + const char * const *slaves) { struct snd_kcontrol *kctl; - const char **s; + const char * const *s; int err; kctl = snd_ctl_make_virtual_master(name, tlv); @@ -3612,7 +3618,7 @@ static int patch_vt1617a(struct snd_ac97 * ac97) struct vt1618_uaj_item { unsigned short mask; unsigned short shift; - const char *items[4]; + const char * const items[4]; }; /* This list reflects the vt1618 docs for Vendor Defined Register 0x60. */ diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index 47bf8df..d1ce151 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -49,7 +49,7 @@ struct ac97_enum { unsigned char shift_l; unsigned char shift_r; unsigned short mask; - const char **texts; + const char * const *texts; }; #define AC97_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ -- cgit v1.1 From 30d0ae425ab1c9bb0003c3798de78fbf30ddebdc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:15:50 +0200 Subject: ALSA: asihpi: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 46 +++++----------------------------------------- 1 file changed, 5 insertions(+), 41 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 5017176..ac66b32 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1625,18 +1625,7 @@ static const char * const asihpi_aesebu_format_names[] = { static int snd_asihpi_aesebu_format_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - asihpi_aesebu_format_names[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, asihpi_aesebu_format_names); } static int snd_asihpi_aesebu_format_get(struct snd_kcontrol *kcontrol, @@ -1863,22 +1852,7 @@ static int snd_asihpi_tuner_band_info(struct snd_kcontrol *kcontrol, if (num_bands < 0) return num_bands; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_bands; - - if (num_bands > 0) { - if (uinfo->value.enumerated.item >= - uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - asihpi_tuner_band_names[ - tuner_bands[uinfo->value.enumerated.item]]); - - } - return 0; + return snd_ctl_enum_info(uinfo, 1, num_bands, asihpi_tuner_band_names); } static int snd_asihpi_tuner_band_get(struct snd_kcontrol *kcontrol, @@ -2253,7 +2227,7 @@ static int snd_asihpi_cmode_info(struct snd_kcontrol *kcontrol, u32 h_control = kcontrol->private_value; u16 mode; int i; - u16 mode_map[6]; + const char *mapped_names[6]; int valid_modes = 0; /* HPI channel mode values can be from 1 to 6 @@ -2262,24 +2236,14 @@ static int snd_asihpi_cmode_info(struct snd_kcontrol *kcontrol, for (i = 0; i < HPI_CHANNEL_MODE_LAST; i++) if (!hpi_channel_mode_query_mode( h_control, i, &mode)) { - mode_map[valid_modes] = mode; + mapped_names[valid_modes] = mode_names[mode]; valid_modes++; } if (!valid_modes) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = valid_modes; - - if (uinfo->value.enumerated.item >= valid_modes) - uinfo->value.enumerated.item = valid_modes - 1; - - strcpy(uinfo->value.enumerated.name, - mode_names[mode_map[uinfo->value.enumerated.item]]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, valid_modes, mapped_names); } static int snd_asihpi_cmode_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 4d765e48c5edb2090b82e97680b2d1ddf6d18c31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:16:01 +0200 Subject: ALSA: aw2: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/pci/aw2/aw2-alsa.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 3878cf5..e1cf019 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -725,19 +725,10 @@ static int snd_aw2_new_pcm(struct aw2 *chip) static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { + static const char * const texts[2] = { "Analog", "Digital" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) { - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - } - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 9b311a0ad9ec0df9f010bcadd19193b1cee593f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:16:13 +0200 Subject: ALSA: azt3328: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5a69e26..fdbb9c0 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1034,11 +1034,6 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, const char * const *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = (reg.reg == IDX_MIXER_REC_SELECT) ? 2 : 1; - uinfo->value.enumerated.items = reg.enum_c; - if (uinfo->value.enumerated.item > reg.enum_c - 1U) - uinfo->value.enumerated.item = reg.enum_c - 1U; if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ @@ -1051,12 +1046,12 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, p = texts4; break; } - } else - if (reg.reg == IDX_MIXER_REC_SELECT) + } else if (reg.reg == IDX_MIXER_REC_SELECT) p = texts3; - strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, + (reg.reg == IDX_MIXER_REC_SELECT) ? 2 : 1, + reg.enum_c, p); } static int -- cgit v1.1 From de95eae25a2744ba5f9bd3c862bb43a1b177ad58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:16:26 +0200 Subject: ALSA: ca0106: Use snd_ctl_enum_info() ... and reduce the open codes. Also correct the array size and add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 40 ++++++++-------------------------------- 1 file changed, 8 insertions(+), 32 deletions(-) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 27de0de..68c0eb0 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -185,17 +185,11 @@ static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol, static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[6] = { + static const char * const texts[6] = { "IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 6; - if (uinfo->value.enumerated.item > 5) - uinfo->value.enumerated.item = 5; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 6, texts); } static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol, @@ -228,17 +222,11 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[6] = { + static const char * const texts[4] = { "Phone", "Mic", "Line in", "Aux" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol, @@ -273,29 +261,17 @@ static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "Side out", "Line in" }; + static const char * const texts[2] = { "Side out", "Line in" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { "Line in", "Mic in" }; + static const char * const texts[2] = { "Line in", "Mic in" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From c69a4f3046ee5a28ab09a1786a73d04bd6177445 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:16:38 +0200 Subject: ALSA: echoaudio: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 33 ++++++--------------------------- 1 file changed, 6 insertions(+), 27 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 631aaa4..d82321f 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1416,21 +1416,14 @@ static struct snd_kcontrol_new snd_echo_vmixer = { static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *names[4] = { + static const char * const names[4] = { "S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical", "S/PDIF Cdrom" }; struct echoaudio *chip; chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = chip->num_digital_modes; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= chip->num_digital_modes) - uinfo->value.enumerated.item = chip->num_digital_modes - 1; - strcpy(uinfo->value.enumerated.name, names[ - chip->digital_mode_list[uinfo->value.enumerated.item]]); - return 0; + return snd_ctl_enum_info(uinfo, 1, chip->num_digital_modes, names); } static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol, @@ -1509,16 +1502,9 @@ static struct snd_kcontrol_new snd_echo_digital_mode_switch = { static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *names[2] = {"Consumer", "Professional"}; + static const char * const names[2] = {"Consumer", "Professional"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 2; - uinfo->count = 1; - if (uinfo->value.enumerated.item) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - names[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, names); } static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol, @@ -1566,21 +1552,14 @@ static struct snd_kcontrol_new snd_echo_spdif_mode_switch = { static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *names[8] = { + static const char * const names[8] = { "Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync", "ESync96", "MTC" }; struct echoaudio *chip; chip = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = chip->num_clock_sources; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= chip->num_clock_sources) - uinfo->value.enumerated.item = chip->num_clock_sources - 1; - strcpy(uinfo->value.enumerated.name, names[ - chip->clock_source_list[uinfo->value.enumerated.item]]); - return 0; + return snd_ctl_enum_info(uinfo, 1, chip->num_clock_sources, names); } static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 1541c66d3bb78c8a388025b074c75658c790b72f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:16:48 +0200 Subject: ALSA: emu10k1: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emumixer.c | 58 ++++++++++---------------------------------- sound/pci/emu10k1/p16v.c | 20 +++------------ 2 files changed, 17 insertions(+), 61 deletions(-) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index c5ae2a2..1de3302 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -83,7 +83,7 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol, * Items labels in enum mixer controls assigning source data to * each destination */ -static char *emu1010_src_texts[] = { +static const char * const emu1010_src_texts[] = { "Silence", "Dock Mic A", "Dock Mic B", @@ -141,7 +141,7 @@ static char *emu1010_src_texts[] = { /* 1616(m) cardbus */ -static char *emu1616_src_texts[] = { +static const char * const emu1616_src_texts[] = { "Silence", "Dock Mic A", "Dock Mic B", @@ -393,23 +393,11 @@ static int snd_emu1010_input_output_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - char **items; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { - uinfo->value.enumerated.items = 49; - items = emu1616_src_texts; - } else { - uinfo->value.enumerated.items = 53; - items = emu1010_src_texts; - } - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - items[uinfo->value.enumerated.item]); - return 0; + if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) + return snd_ctl_enum_info(uinfo, 1, 49, emu1616_src_texts); + else + return snd_ctl_enum_info(uinfo, 1, 53, emu1010_src_texts); } static int snd_emu1010_output_source_get(struct snd_kcontrol *kcontrol, @@ -699,19 +687,11 @@ static struct snd_kcontrol_new snd_emu1010_dac_pads[] = { static int snd_emu1010_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { + static const char * const texts[4] = { "44100", "48000", "SPDIF", "ADAT" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; - - + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_emu1010_internal_clock_get(struct snd_kcontrol *kcontrol, @@ -830,21 +810,15 @@ static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { #if 0 - static char *texts[4] = { + static const char * const texts[4] = { "Unknown1", "Unknown2", "Mic", "Line" }; #endif - static char *texts[2] = { + static const char * const texts[2] = { "Mic", "Line" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_audigy_i2c_capture_source_get(struct snd_kcontrol *kcontrol, @@ -997,15 +971,9 @@ static struct snd_kcontrol_new snd_audigy_i2c_volume_ctls[] = { #if 0 static int snd_audigy_spdif_output_rate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"44100", "48000", "96000"}; + static const char * const texts[] = {"44100", "48000", "96000"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_audigy_spdif_output_rate_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index a4fe7f0..7ef3898 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -757,18 +757,12 @@ static int snd_p16v_volume_put(struct snd_kcontrol *kcontrol, static int snd_p16v_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[8] = { + static const char * const texts[8] = { "SPDIF", "I2S", "SRC48", "SRCMulti_SPDIF", "SRCMulti_I2S", "CDIF", "FX", "AC97" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 8, texts); } static int snd_p16v_capture_source_get(struct snd_kcontrol *kcontrol, @@ -805,15 +799,9 @@ static int snd_p16v_capture_source_put(struct snd_kcontrol *kcontrol, static int snd_p16v_capture_channel_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { "0", "1", "2", "3", }; + static const char * const texts[4] = { "0", "1", "2", "3", }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_p16v_capture_channel_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 6b6b295e8053dd5a005aaa089b5bed4b4a65c632 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:17:08 +0200 Subject: ALSA: es1938: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/pci/es1938.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 6399624..0fc46eb 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1045,18 +1045,12 @@ static int snd_es1938_new_pcm(struct es1938 *chip, int device) static int snd_es1938_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[8] = { + static const char * const texts[8] = { "Mic", "Mic Master", "CD", "AOUT", "Mic1", "Mix", "Line", "Master" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 8, texts); } static int snd_es1938_get_mux(struct snd_kcontrol *kcontrol, -- cgit v1.1 From ca776a28ae10bb06807f23e807f0f459dab78318 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:17:17 +0200 Subject: ALSA: fm801: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index c503830..d167aff 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -958,17 +958,11 @@ static int snd_fm801_put_double(struct snd_kcontrol *kcontrol, static int snd_fm801_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { + static const char * const texts[5] = { "AC97 Primary", "FM", "I2S", "PCM", "AC97 Secondary" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item > 4) - uinfo->value.enumerated.item = 4; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } static int snd_fm801_get_mux(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 3ff72219320f616489bf0d98ddac12899da4a9ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:17:28 +0200 Subject: ALSA: hda: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 ++----------------- 1 file changed, 2 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 15e0089..259fbea 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2927,16 +2927,8 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, static const char * const texts[] = { "On", "Off", "Follow Master" }; - unsigned int index; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - index = uinfo->value.enumerated.item; - if (index >= 3) - index = 2; - strcpy(uinfo->value.enumerated.name, texts[index]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol, @@ -5195,14 +5187,7 @@ int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol, texts = texts_default; } - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_items; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, num_items, texts); } EXPORT_SYMBOL_GPL(snd_hda_enum_helper_info); -- cgit v1.1 From c4fa251f6f3ed00d59d0d8ee63bf346e6dd6b664 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:18:33 +0200 Subject: ALSA: ice1712: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ews.c | 16 ++-------------- sound/pci/ice1712/ice1712.c | 27 ++++----------------------- 2 files changed, 6 insertions(+), 37 deletions(-) diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c index 817a1bc..5cb587c 100644 --- a/sound/pci/ice1712/ews.c +++ b/sound/pci/ice1712/ews.c @@ -580,13 +580,7 @@ static int snd_ice1712_ewx_io_sense_info(struct snd_kcontrol *kcontrol, struct s static const char * const texts[2] = { "+4dBu", "-10dBV", }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= 2) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_ice1712_ewx_io_sense_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -903,13 +897,7 @@ static int snd_ice1712_6fire_select_input_info(struct snd_kcontrol *kcontrol, st static const char * const texts[4] = { "Internal", "Front Input", "Rear Input", "Wave Table" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_ice1712_6fire_select_input_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 206ed2c..48a0c33 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1839,13 +1839,7 @@ static int snd_ice1712_pro_internal_clock_info(struct snd_kcontrol *kcontrol, "96000", /* 12: 7 */ "IEC958 Input", /* 13: -- */ }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 14; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 14, texts); } static int snd_ice1712_pro_internal_clock_get(struct snd_kcontrol *kcontrol, @@ -1930,13 +1924,7 @@ static int snd_ice1712_pro_internal_clock_default_info(struct snd_kcontrol *kcon "96000", /* 12: 7 */ /* "IEC958 Input", 13: -- */ }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 13; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 13, texts); } static int snd_ice1712_pro_internal_clock_default_get(struct snd_kcontrol *kcontrol, @@ -2057,15 +2045,8 @@ static int snd_ice1712_pro_route_info(struct snd_kcontrol *kcontrol, "IEC958 In L", "IEC958 In R", /* 9-10 */ "Digital Mixer", /* 11 - optional */ }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = - snd_ctl_get_ioffidx(kcontrol, &uinfo->id) < 2 ? 12 : 11; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + int num_items = snd_ctl_get_ioffidx(kcontrol, &uinfo->id) < 2 ? 12 : 11; + return snd_ctl_enum_info(uinfo, 1, num_items, texts); } static int snd_ice1712_pro_route_analog_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 597da2e4dfa04c8ee66b09fce931ab6825bc3e75 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:18:50 +0200 Subject: ALSA: ice1724: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 46 +++++++--------------------------------- sound/pci/ice1712/ice1724.c | 8 +------ sound/pci/ice1712/maya44.c | 20 ++--------------- sound/pci/ice1712/phase.c | 12 +---------- sound/pci/ice1712/pontis.c | 8 +------ sound/pci/ice1712/prodigy192.c | 18 ++-------------- sound/pci/ice1712/prodigy_hifi.c | 11 ++-------- sound/pci/ice1712/quartet.c | 27 ++++------------------- sound/pci/ice1712/se.c | 9 +------- 9 files changed, 22 insertions(+), 137 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 3b3cf4a..c9411df 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -205,13 +205,7 @@ static int aureon_universe_inmux_info(struct snd_kcontrol *kcontrol, static const char * const texts[3] = {"Internal Aux", "Wavetable", "Rear Line-In"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int aureon_universe_inmux_get(struct snd_kcontrol *kcontrol, @@ -1106,20 +1100,10 @@ static int wm_adc_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_in }; struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) { - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, universe_texts[uinfo->value.enumerated.item]); - } else { - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - } - return 0; + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON71_UNIVERSE) + return snd_ctl_enum_info(uinfo, 2, 8, universe_texts); + else + return snd_ctl_enum_info(uinfo, 2, 5, texts); } static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1167,16 +1151,10 @@ static int aureon_cs8415_mux_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ "CD", "Coax" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71) - strcpy(uinfo->value.enumerated.name, prodigy_texts[uinfo->value.enumerated.item]); + return snd_ctl_enum_info(uinfo, 1, 2, prodigy_texts); else - strcpy(uinfo->value.enumerated.name, aureon_texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, aureon_texts); } static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1392,15 +1370,7 @@ static int aureon_oversampling_info(struct snd_kcontrol *k, struct snd_ctl_elem_ { static const char * const texts[2] = { "128x", "64x" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int aureon_oversampling_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 08cb08a..f633e3b 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2049,13 +2049,7 @@ static int snd_vt1724_pro_route_info(struct snd_kcontrol *kcontrol, "IEC958 In L", "IEC958 In R", /* 3-4 */ }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } static inline int analog_route_shift(int idx) diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c index 63aa39f..7de25c4 100644 --- a/sound/pci/ice1712/maya44.c +++ b/sound/pci/ice1712/maya44.c @@ -359,15 +359,7 @@ static int maya_rec_src_info(struct snd_kcontrol *kcontrol, { static const char * const texts[] = { "Line", "Mic" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ARRAY_SIZE(texts); - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } static int maya_rec_src_get(struct snd_kcontrol *kcontrol, @@ -411,15 +403,7 @@ static int maya_pb_route_info(struct snd_kcontrol *kcontrol, "Input 1", "Input 2", "Input 3", "Input 4" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ARRAY_SIZE(texts); - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } static int maya_pb_route_shift(int idx) diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index 0011e04..e9ca89c 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -723,17 +723,7 @@ static int phase28_oversampling_info(struct snd_kcontrol *k, { static const char * const texts[2] = { "128x", "64x" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int phase28_oversampling_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 5555eb4..5101f40 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -417,13 +417,7 @@ static int cs_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_inf "Optical", /* RXP1 */ "CD", /* RXP2 */ }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int cs_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index f3b491a..1eb151aa 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -284,15 +284,7 @@ static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, { static const char * const texts[2] = { "Line In", "Mic" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } @@ -563,13 +555,7 @@ static int ak4114_input_sw_info(struct snd_kcontrol *kcontrol, { static const char * const texts[2] = { "Toslink", "Coax" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2261d1e..2697402 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -537,7 +537,7 @@ static int wm_master_vol_put(struct snd_kcontrol *kcontrol, static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char* texts[32] = { + static const char * const texts[32] = { "NULL", WM_AIN1, WM_AIN2, WM_AIN1 "+" WM_AIN2, WM_AIN3, WM_AIN1 "+" WM_AIN3, WM_AIN2 "+" WM_AIN3, WM_AIN1 "+" WM_AIN2 "+" WM_AIN3, @@ -560,14 +560,7 @@ static int wm_adc_mux_enum_info(struct snd_kcontrol *kcontrol, WM_AIN1 "+" WM_AIN2 "+" WM_AIN3 "+" WM_AIN4 "+" WM_AIN5 }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 32; - if (uinfo->value.enumerated.item > 31) - uinfo->value.enumerated.item = 31; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 32, texts); } static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 2c2df4b..d4caf9d 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -46,7 +46,7 @@ struct qtet_kcontrol_private { unsigned int bit; void (*set_register)(struct snd_ice1712 *ice, unsigned int val); unsigned int (*get_register)(struct snd_ice1712 *ice); - unsigned char * const texts[2]; + const char * const texts[2]; }; enum { @@ -554,17 +554,7 @@ static int qtet_ain12_enum_info(struct snd_kcontrol *kcontrol, { static const char * const texts[3] = {"Line In 1/2", "Mic", "Mic + Low-cut"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ARRAY_SIZE(texts); - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } static int qtet_ain12_sw_get(struct snd_kcontrol *kcontrol, @@ -706,17 +696,8 @@ static int qtet_enum_info(struct snd_kcontrol *kcontrol, { struct qtet_kcontrol_private private = qtet_privates[kcontrol->private_value]; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ARRAY_SIZE(private.texts); - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - private.texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(private.texts), + private.texts); } static int qtet_sw_get(struct snd_kcontrol *kcontrol, diff --git a/sound/pci/ice1712/se.c b/sound/pci/ice1712/se.c index ffd894b..1c5d5b2 100644 --- a/sound/pci/ice1712/se.c +++ b/sound/pci/ice1712/se.c @@ -452,14 +452,7 @@ static int se200pci_cont_enum_info(struct snd_kcontrol *kc, c = se200pci_get_enum_count(n); if (!c) return -EINVAL; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = c; - if (uinfo->value.enumerated.item >= c) - uinfo->value.enumerated.item = c - 1; - strcpy(uinfo->value.enumerated.name, - se200pci_cont[n].member[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, c, se200pci_cont[n].member); } static int se200pci_cont_volume_get(struct snd_kcontrol *kc, -- cgit v1.1 From f861237c80a07449abd351c04a6ba397418dc0ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:21:58 +0200 Subject: ALSA: korg1212: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/korg1212/korg1212.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 9fe549b..59d21c9 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -444,9 +444,9 @@ static char *stateName[] = { "Invalid" }; -static char *clockSourceTypeName[] = { "ADAT", "S/PDIF", "local" }; +static const char * const clockSourceTypeName[] = { "ADAT", "S/PDIF", "local" }; -static char *clockSourceName[] = { +static const char * const clockSourceName[] = { "ADAT at 44.1 kHz", "ADAT at 48 kHz", "S/PDIF at 44.1 kHz", @@ -455,7 +455,7 @@ static char *clockSourceName[] = { "local clock at 48 kHz" }; -static char *channelName[] = { +static const char * const channelName[] = { "ADAT-1", "ADAT-2", "ADAT-3", @@ -1844,14 +1844,9 @@ static int snd_korg1212_control_volume_put(struct snd_kcontrol *kcontrol, static int snd_korg1212_control_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = (kcontrol->private_value >= 8) ? 2 : 1; - uinfo->value.enumerated.items = kAudioChannels; - if (uinfo->value.enumerated.item > kAudioChannels-1) { - uinfo->value.enumerated.item = kAudioChannels-1; - } - strcpy(uinfo->value.enumerated.name, channelName[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, + (kcontrol->private_value >= 8) ? 2 : 1, + kAudioChannels, channelName); } static int snd_korg1212_control_route_get(struct snd_kcontrol *kcontrol, @@ -1961,14 +1956,7 @@ static int snd_korg1212_control_put(struct snd_kcontrol *kcontrol, static int snd_korg1212_control_sync_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) { - uinfo->value.enumerated.item = 2; - } - strcpy(uinfo->value.enumerated.name, clockSourceTypeName[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, clockSourceTypeName); } static int snd_korg1212_control_sync_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 08455ace3cafd9b0b5c35db3d89c4388f6d3a6fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:00 +0200 Subject: ALSA: pcxhr: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_mixer.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 95c9571..63136c4 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -660,14 +660,7 @@ static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol, if (chip->mgr->board_has_mic) i = 5; /* Mic and MicroMix available */ } - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = i; - if (uinfo->value.enumerated.item > (i-1)) - uinfo->value.enumerated.item = i-1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, i, texts); } static int pcxhr_audio_src_get(struct snd_kcontrol *kcontrol, @@ -756,14 +749,7 @@ static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol, texts = textsPCXHR; snd_BUG_ON(clock_items > (PCXHR_CLOCK_TYPE_MAX+1)); } - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = clock_items; - if (uinfo->value.enumerated.item >= clock_items) - uinfo->value.enumerated.item = clock_items-1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, clock_items, texts); } static int pcxhr_clock_type_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 11c6ef7c8d439ef2bc3c95e5a4dcea449ab1f90f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:12 +0200 Subject: ALSA: rme32: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/rme32.c | 34 ++++++++++------------------------ 1 file changed, 10 insertions(+), 24 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 4afd3ca..6c60dcd 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1608,30 +1608,24 @@ snd_rme32_info_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct rme32 *rme32 = snd_kcontrol_chip(kcontrol); - static char *texts[4] = { "Optical", "Coaxial", "Internal", "XLR" }; + static const char * const texts[4] = { + "Optical", "Coaxial", "Internal", "XLR" + }; + int num_items; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; switch (rme32->pci->device) { case PCI_DEVICE_ID_RME_DIGI32: case PCI_DEVICE_ID_RME_DIGI32_8: - uinfo->value.enumerated.items = 3; + num_items = 3; break; case PCI_DEVICE_ID_RME_DIGI32_PRO: - uinfo->value.enumerated.items = 4; + num_items = 4; break; default: snd_BUG(); - break; - } - if (uinfo->value.enumerated.item > - uinfo->value.enumerated.items - 1) { - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; + return -EINVAL; } - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, num_items, texts); } static int snd_rme32_get_inputtype_control(struct snd_kcontrol *kcontrol, @@ -1695,20 +1689,12 @@ static int snd_rme32_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { "AutoSync", + static const char * const texts[4] = { "AutoSync", "Internal 32.0kHz", "Internal 44.1kHz", "Internal 48.0kHz" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) { - uinfo->value.enumerated.item = 3; - } - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_rme32_get_clockmode_control(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 9c30d46a0fb3b294faf1226025071d6e802a8c36 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:23 +0200 Subject: ALSA: rme96: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 62 +++++++++++++++++++------------------------------------ 1 file changed, 21 insertions(+), 41 deletions(-) diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 5a395c8..2f1a851 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1884,39 +1884,38 @@ snd_rme96_put_loopback_control(struct snd_kcontrol *kcontrol, struct snd_ctl_ele static int snd_rme96_info_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *_texts[5] = { "Optical", "Coaxial", "Internal", "XLR", "Analog" }; + static const char * const _texts[5] = { + "Optical", "Coaxial", "Internal", "XLR", "Analog" + }; struct rme96 *rme96 = snd_kcontrol_chip(kcontrol); - char *texts[5] = { _texts[0], _texts[1], _texts[2], _texts[3], _texts[4] }; + const char *texts[5] = { + _texts[0], _texts[1], _texts[2], _texts[3], _texts[4] + }; + int num_items; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: case PCI_DEVICE_ID_RME_DIGI96_8: - uinfo->value.enumerated.items = 3; + num_items = 3; break; case PCI_DEVICE_ID_RME_DIGI96_8_PRO: - uinfo->value.enumerated.items = 4; + num_items = 4; break; case PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST: if (rme96->rev > 4) { /* PST */ - uinfo->value.enumerated.items = 4; + num_items = 4; texts[3] = _texts[4]; /* Analog instead of XLR */ } else { /* PAD */ - uinfo->value.enumerated.items = 5; + num_items = 5; } break; default: snd_BUG(); - break; - } - if (uinfo->value.enumerated.item > uinfo->value.enumerated.items - 1) { - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + return -EINVAL; } - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, num_items, texts); } static int snd_rme96_get_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2002,16 +2001,9 @@ snd_rme96_put_inputtype_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el static int snd_rme96_info_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { "AutoSync", "Internal", "Word" }; + static const char * const texts[3] = { "AutoSync", "Internal", "Word" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) { - uinfo->value.enumerated.item = 2; - } - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_rme96_get_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2041,16 +2033,11 @@ snd_rme96_put_clockmode_control(struct snd_kcontrol *kcontrol, struct snd_ctl_el static int snd_rme96_info_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { "0 dB", "-6 dB", "-12 dB", "-18 dB" }; + static const char * const texts[4] = { + "0 dB", "-6 dB", "-12 dB", "-18 dB" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) { - uinfo->value.enumerated.item = 3; - } - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_rme96_get_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2081,16 +2068,9 @@ snd_rme96_put_attenuation_control(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_rme96_info_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { "1+2", "3+4", "5+6", "7+8" }; + static const char * const texts[4] = { "1+2", "3+4", "5+6", "7+8" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) { - uinfo->value.enumerated.item = 3; - } - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_rme96_get_montracks_control(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 8d678da9f0afbb951778369510c09b99de608c24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:34 +0200 Subject: ALSA: hdsp: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 175 +++++++++++++++-------------------------------- 1 file changed, 57 insertions(+), 118 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7646ba1..2eb8baf 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1680,16 +1680,13 @@ static int hdsp_set_spdif_input(struct hdsp *hdsp, int in) static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = {"Optical", "Coaxial", "Internal", "AES"}; + static const char * const texts[4] = { + "Optical", "Coaxial", "Internal", "AES" + }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ((hdsp->io_type == H9632) ? 4 : 3); - if (uinfo->value.enumerated.item > ((hdsp->io_type == H9632) ? 3 : 2)) - uinfo->value.enumerated.item = ((hdsp->io_type == H9632) ? 3 : 2); - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 4 : 3, + texts); } static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1786,16 +1783,14 @@ static int snd_hdsp_put_toggle_setting(struct snd_kcontrol *kcontrol, static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static const char * const texts[] = { + "32000", "44100", "48000", "64000", "88200", "96000", + "None", "128000", "176400", "192000" + }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7, + texts); } static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1872,14 +1867,13 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + static const char * const texts[] = { + "32000", "44100", "48000", "64000", "88200", "96000", + "None", "128000", "176400", "192000" + }; + + return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7, + texts); } static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1940,15 +1934,9 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"Master", "Slave" }; + static const char * const texts[] = {"Master", "Slave" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2049,19 +2037,16 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; + static const char * const texts[] = { + "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", + "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", + "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", + "Internal 192.0 KHz" + }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - if (hdsp->io_type == H9632) - uinfo->value.enumerated.items = 10; - else - uinfo->value.enumerated.items = 7; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, (hdsp->io_type == H9632) ? 10 : 7, + texts); } static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2165,15 +2150,9 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; + static const char * const texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2250,15 +2229,9 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; + static const char * const texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2335,15 +2308,9 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; + static const char * const texts[] = {"0 dB", "-6 dB", "-12 dB"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2439,31 +2406,28 @@ static int hdsp_set_pref_sync_ref(struct hdsp *hdsp, int pref) static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; + static const char * const texts[] = { + "Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" + }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; + int num_items; switch (hdsp->io_type) { case Digiface: case H9652: - uinfo->value.enumerated.items = 6; + num_items = 6; break; case Multiface: - uinfo->value.enumerated.items = 4; + num_items = 4; break; case H9632: - uinfo->value.enumerated.items = 3; + num_items = 3; break; default: return -EINVAL; } - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, num_items, texts); } static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2543,15 +2507,11 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; + static const char * const texts[] = { + "Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 7; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 7, texts); } static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2738,14 +2698,9 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + static const char * const texts[] = {"No Lock", "Lock", "Sync" }; + + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int hdsp_wc_sync_check(struct hdsp *hdsp) @@ -3101,15 +3056,11 @@ static int snd_hdsp_put_rpm_input12(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_info_rpm_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB"}; + static const char * const texts[] = { + "Phono +6dB", "Phono 0dB", "Phono -6dB", "Line 0dB", "Line -6dB" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 5, texts); } @@ -3234,15 +3185,9 @@ static int snd_hdsp_put_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl static int snd_hdsp_info_rpm_bypass(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"On", "Off"}; + static const char * const texts[] = {"On", "Off"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } @@ -3291,15 +3236,9 @@ static int snd_hdsp_put_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_info_rpm_disconnect(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"On", "Off"}; + static const char * const texts[] = {"On", "Off"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static struct snd_kcontrol_new snd_hdsp_rpm_controls[] = { -- cgit v1.1 From c69a637b4df37fc5a011a89e422636ea393af5b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:41 +0200 Subject: ALSA: hdspm: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 52d86af..7f7277b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2645,18 +2645,7 @@ static int hdspm_set_clock_source(struct hdspm * hdspm, int mode) static int snd_hdspm_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 9; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - texts_freq[uinfo->value.enumerated.item+1]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, 9, texts_freq + 1); } static int snd_hdspm_get_clock_source(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 7298ece7a26753b073a9ce5f979a4942d3904d44 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:19:53 +0200 Subject: ALSA: rme9652: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/rme9652.c | 58 ++++++++++++++------------------------------- 1 file changed, 18 insertions(+), 40 deletions(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index fa9a2a8..6521521 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -920,15 +920,9 @@ static int rme9652_set_adat1_input(struct snd_rme9652 *rme9652, int internal) static int snd_rme9652_info_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = {"ADAT1", "Internal"}; + static const char * const texts[2] = {"ADAT1", "Internal"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_rme9652_get_adat1_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -991,15 +985,9 @@ static int rme9652_set_spdif_input(struct snd_rme9652 *rme9652, int in) static int snd_rme9652_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = {"ADAT1", "Coaxial", "Internal"}; + static const char * const texts[3] = {"ADAT1", "Coaxial", "Internal"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_rme9652_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1140,15 +1128,11 @@ static int rme9652_set_sync_mode(struct snd_rme9652 *rme9652, int mode) static int snd_rme9652_info_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = {"AutoSync", "Master", "Word Clock"}; + static const char * const texts[3] = { + "AutoSync", "Master", "Word Clock" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 3, texts); } static int snd_rme9652_get_sync_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1231,16 +1215,14 @@ static int rme9652_set_sync_pref(struct snd_rme9652 *rme9652, int pref) static int snd_rme9652_info_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = {"IEC958 In", "ADAT1 In", "ADAT2 In", "ADAT3 In"}; + static const char * const texts[4] = { + "IEC958 In", "ADAT1 In", "ADAT2 In", "ADAT3 In" + }; struct snd_rme9652 *rme9652 = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, + rme9652->ss_channels == RME9652_NCHANNELS ? 4 : 3, + texts); } static int snd_rme9652_get_sync_pref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1392,15 +1374,11 @@ static int snd_rme9652_get_spdif_rate(struct snd_kcontrol *kcontrol, struct snd_ static int snd_rme9652_info_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = {"No Lock", "Lock", "No Lock Sync", "Lock Sync"}; + static const char * const texts[4] = { + "No Lock", "Lock", "No Lock Sync", "Lock Sync" + }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 4, texts); } static int snd_rme9652_get_adat_sync(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 3e4bc5b78e5516585941c7888287ed50a5090bf4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:20:07 +0200 Subject: ALSA: sonicvibes: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/pci/sonicvibes.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 5b0d317..313a732 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -918,17 +918,11 @@ static int snd_sonicvibes_pcm(struct sonicvibes *sonic, int device, static int snd_sonicvibes_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[7] = { + static const char * const texts[7] = { "CD", "PCM", "Aux1", "Line", "Aux0", "Mic", "Mix" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 7; - if (uinfo->value.enumerated.item >= 7) - uinfo->value.enumerated.item = 6; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 2, 7, texts); } static int snd_sonicvibes_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.1 From 9883ab91e3ba5229bfe2d6e7f6ff497a2d03d4d2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:20:18 +0200 Subject: ALSA: via82xx: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index ecedf4d..e088467 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1610,16 +1610,10 @@ static int snd_via8233_capture_source_info(struct snd_kcontrol *kcontrol, /* formerly they were "Line" and "Mic", but it looks like that they * have nothing to do with the actual physical connections... */ - static char *texts[2] = { + static const char * const texts[2] = { "Input1", "Input2" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item >= 2) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snd_via8233_capture_source_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 9502272163ace71d77d809937216fd669c02364b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:20:28 +0200 Subject: ALSA: ppc: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/ppc/tumbler.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index b9ffc17..24c8766 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -795,16 +795,11 @@ static int snapper_set_capture_source(struct pmac_tumbler *mix) static int snapper_info_capture_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2] = { + static const char * const texts[2] = { "Line", "Mic" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); } static int snapper_get_capture_source(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 5fe0b0e3ea1c8cb704677ef7e85345bb683f9182 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:21:15 +0200 Subject: ALSA: sparc: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to the text array. Signed-off-by: Takashi Iwai --- sound/sparc/cs4231.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 4e91bca..06606f9 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1285,19 +1285,11 @@ static int snd_cs4231_timer(struct snd_card *card) static int snd_cs4231_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[4] = { + static const char * const texts[4] = { "Line", "CD", "Mic", "Mix" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 2, 4, texts); } static int snd_cs4231_get_mux(struct snd_kcontrol *kcontrol, -- cgit v1.1 From c8dd33fc80cd344d28a1f6a7e1f0af1cf7a2ffd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:21:27 +0200 Subject: ALSA: 6fire: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 22 ++++------------------ 1 file changed, 4 insertions(+), 18 deletions(-) diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index 184e398..54656ee 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -25,8 +25,8 @@ #include "comm.h" #include "chip.h" -static char *opt_coax_texts[2] = { "Optical", "Coax" }; -static char *line_phono_texts[2] = { "Line", "Phono" }; +static const char * const opt_coax_texts[2] = { "Optical", "Coax" }; +static const char * const line_phono_texts[2] = { "Line", "Phono" }; /* * data that needs to be sent to device. sets up card internal stuff. @@ -327,14 +327,7 @@ static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - line_phono_texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, line_phono_texts); } static int usb6fire_control_line_phono_put(struct snd_kcontrol *kcontrol, @@ -361,14 +354,7 @@ static int usb6fire_control_line_phono_get(struct snd_kcontrol *kcontrol, static int usb6fire_control_opt_coax_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - opt_coax_texts[uinfo->value.enumerated.item]); - return 0; + return snd_ctl_enum_info(uinfo, 1, 2, opt_coax_texts); } static int usb6fire_control_opt_coax_put(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 7bbd03e0143b562ff7d96f7e71c016104020b550 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Oct 2014 18:21:42 +0200 Subject: ALSA: usb-audio: Use snd_ctl_enum_info() ... and reduce the open codes. Also add missing const to text arrays. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 35 ++++++----------------------------- 1 file changed, 6 insertions(+), 29 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f119a41..3980bf5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -437,19 +437,9 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, static int snd_emu0204_ch_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[2] = {"1/2", - "3/4" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + static const char * const texts[2] = {"1/2", "3/4"}; - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } static int snd_emu0204_ch_switch_get(struct snd_kcontrol *kcontrol, @@ -735,25 +725,12 @@ struct snd_ftu_eff_switch_priv_val { static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char *texts[8] = {"Room 1", - "Room 2", - "Room 3", - "Hall 1", - "Hall 2", - "Plate", - "Delay", - "Echo" + static const char *const texts[8] = { + "Room 1", "Room 2", "Room 3", "Hall 1", + "Hall 2", "Plate", "Delay", "Echo" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, -- cgit v1.1 From cf6814f2b5014ed5bbdef764a42e4abaa09b3a2e Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Tue, 21 Oct 2014 16:28:47 +0530 Subject: ALSA: ctxfi: remove unused variable As of now the pointer to struct dai is not being used anywhere in the function. So it is safe to remove the variable. If we are ever doing anything with the container_of(daio, struct dai, daio), then at that time we can again add the variable. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 4546590..632e843 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1145,7 +1145,6 @@ static int atc_release_resources(struct ct_atc *atc) int i; struct daio_mgr *daio_mgr = NULL; struct dao *dao = NULL; - struct dai *dai = NULL; struct daio *daio = NULL; struct sum_mgr *sum_mgr = NULL; struct src_mgr *src_mgr = NULL; @@ -1172,9 +1171,6 @@ static int atc_release_resources(struct ct_atc *atc) dao = container_of(daio, struct dao, daio); dao->ops->clear_left_input(dao); dao->ops->clear_right_input(dao); - } else { - dai = container_of(daio, struct dai, daio); - /* some thing to do for dai ... */ } daio_mgr->put_daio(daio_mgr, daio); } -- cgit v1.1 From 469cda294dac1c4128e9adcb0a94636bc0cb280e Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Tue, 21 Oct 2014 17:33:29 +0200 Subject: ASoC: tegra: Read and use the GPIO flags of the headphone detect The headphone detect was hardcoded to low-active, use the flags from DT to allow high-active as well. Signed-off-by: Alban Bedel Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5640.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index a689883..4ebe387 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -44,6 +44,7 @@ struct tegra_rt5640 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + enum of_gpio_flags gpio_hp_det_flags; }; static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream, @@ -119,6 +120,8 @@ static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; + tegra_rt5640_hp_jack_gpio.invert = + !!(machine->gpio_hp_det_flags & OF_GPIO_ACTIVE_LOW); snd_soc_jack_add_gpios(&tegra_rt5640_hp_jack, 1, &tegra_rt5640_hp_jack_gpio); @@ -180,7 +183,8 @@ static int tegra_rt5640_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + machine->gpio_hp_det = of_get_named_gpio_flags( + np, "nvidia,hp-det-gpios", 0, &machine->gpio_hp_det_flags); if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- cgit v1.1 From 98ad73c995ed4886c36a1fcfcda53fbff484f666 Mon Sep 17 00:00:00 2001 From: Rasmus Villemoes Date: Tue, 21 Oct 2014 17:01:15 +0200 Subject: ASoC: dapm: Remove redundant cast Both path->name and e->texts[i] have type const char*, so the cast is slightly confusing and certainly unnecessary. Signed-off-by: Rasmus Villemoes Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..39f992b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -496,7 +496,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = (char*)e->texts[i]; + path->name = e->texts[i]; if (i == item) path->connect = 1; else -- cgit v1.1 From b3baaa47cc49fab3ecffbbaee660ce003d17d1f7 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 20 Oct 2014 20:54:30 +0530 Subject: ASoC: intel: use __iowrite32_copy for 32 bit copy The driver was using own method to do 32bit copy, turns out we have a kernel API so use that instead Tested-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_loader.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index b6d27c1..00f60c1 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,14 +39,12 @@ #include "sst.h" #include "../sst-dsp.h" -static void memcpy32_toio(void __iomem *dst, const void *src, int count) +static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) { - int i; - const u32 *src_32 = src; - u32 *dst_32 = dst; - - for (i = 0; i < count/sizeof(u32); i++) - writel(*src_32++, dst_32++); + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); } /** -- cgit v1.1 From 790b4075b3a6845543d02ab29c81dc450e7b6794 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 20 Oct 2014 20:54:31 +0530 Subject: ASoC: intel: log an error on double free the stream context should be freed only once on stream cleanup. If we ever hit a chance that stream context is getting double freed, though not an cause of panic as memory allocator can deal with this, we should still log this to help in finding issues and debugging Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index aadb0db..3a5e920 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -55,6 +55,8 @@ int free_stream_context(struct intel_sst_drv *ctx, unsigned int str_id) if (ret) sst_clean_stream(&ctx->streams[str_id]); return ret; + } else { + dev_err(ctx->dev, "we tried to free stream context %d which was freed!!!\n", str_id); } return ret; } -- cgit v1.1 From dee2ce696ea8b37a26eef9f2a3fcc64df7179d98 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 20 Oct 2014 20:54:32 +0530 Subject: ASoC: intel: fix the kernldoc comment copypaste error on function sst_get_num_channel caused the comment to be wrong, so fix it here Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 3a5e920..183b1eb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -94,7 +94,7 @@ int sst_get_sfreq(struct snd_sst_params *str_param) } /* - * sst_get_sfreq - this function returns the frequency of the stream + * sst_get_num_channel - get number of channels for the stream * * @str_param : stream params */ -- cgit v1.1 From 33c1256f1ce30a94f4b590bb30baf787e17f64aa Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 20 Oct 2014 20:54:33 +0530 Subject: ASoC: intel: explain why block not found isn't error always The IPC blocking can be error when we don't find block or a short message, explain that by adding a comment about this scenario Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_ipc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/sst/sst_ipc.c b/sound/soc/intel/sst/sst_ipc.c index 41a2b41..2126f5b 100644 --- a/sound/soc/intel/sst/sst_ipc.c +++ b/sound/soc/intel/sst/sst_ipc.c @@ -54,6 +54,21 @@ struct sst_block *sst_create_block(struct intel_sst_drv *ctx, return msg; } +/* + * while handling the interrupts, we need to check for message status and + * then if we are blocking for a message + * + * here we are unblocking the blocked ones, this is based on id we have + * passed and search that for block threads. + * We will not find block in two cases + * a) when its small message and block in not there, so silently ignore + * them + * b) when we are actually not able to find the block (bug perhaps) + * + * Since we have bit of small messages we can spam kernel log with err + * print on above so need to keep as debug prints which should be enabled + * via dynamic debug while debugging IPC issues + */ int sst_wake_up_block(struct intel_sst_drv *ctx, int result, u32 drv_id, u32 ipc, void *data, u32 size) { -- cgit v1.1 From 7f26680170e322730c7c7553f5625fb04de4f5b8 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 20 Oct 2014 20:54:34 +0530 Subject: ASoC: intel: use __iowrite32_copy for 32 bit copy The sst-firmware was also using own method to do 32bit copy, turns out we have a kernel API so use that instead [For BYT] Tested-by: Jarkko Nikula Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 3bb43da..cf3d199 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -32,13 +32,10 @@ static void block_module_remove(struct sst_module *module); -static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) +static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { - u32 i; - - /* copy one 32 bit word at a time as 64 bit access is not supported */ - for (i = 0; i < bytes; i += 4) - memcpy_toio(dest + i, src + i, 4); + /* __iowrite32_copy use 32bit size values so divide by 4 */ + __iowrite32_copy((void *)dest, src, bytes/4); } /* create new generic firmware object */ -- cgit v1.1 From c37de55efa1ccf018c27b876560725ff5e9f5701 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Oct 2014 12:01:13 +0200 Subject: ALSA: pcm: Remove arch-dependent mmap kludges Since we have consistently dma_mmap_coherent() for all architectures, the current ifdef and arch-specific codes in pcm core can be cleaned up gracefully. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 30 ------------------------------ 1 file changed, 30 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 815396d..aa6754d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,9 +35,6 @@ #include #include #include -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) -#include -#endif /* * Compatibility @@ -3251,20 +3248,6 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; -#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) - return virt_to_page(CAC_ADDR(vaddr)); -#endif -#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { - dma_addr_t addr = substream->runtime->dma_addr + ofs; - addr -= get_dma_offset(substream->dma_buffer.dev.dev); - /* assume dma_handle set via pfn_to_phys() in - * mm/dma-noncoherent.c - */ - return pfn_to_page(addr >> PAGE_SHIFT); - } -#endif return virt_to_page(vaddr); } @@ -3309,13 +3292,6 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .fault = snd_pcm_mmap_data_fault, }; -#ifndef ARCH_HAS_DMA_MMAP_COHERENT -/* This should be defined / handled globally! */ -#if defined(CONFIG_ARM) || defined(CONFIG_ARM64) -#define ARCH_HAS_DMA_MMAP_COHERENT -#endif -#endif - /* * mmap the DMA buffer on RAM */ @@ -3331,7 +3307,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ -#ifdef ARCH_HAS_DMA_MMAP_COHERENT if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3339,11 +3314,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); -#elif defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) - if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV && - !plat_device_is_coherent(substream->dma_buffer.dev.dev)) - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- cgit v1.1 From 78cb4d995b932d9014342ddc81ce0a5cc434ed98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Oct 2014 12:14:56 +0200 Subject: ASoC: core: Use snd_ctl_enum_info() ... and reduce the open codes. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..96ecdc3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2348,16 +2348,8 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, { struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->items; - - if (uinfo->value.enumerated.item >= e->items) - uinfo->value.enumerated.item = e->items - 1; - strlcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item], - sizeof(uinfo->value.enumerated.name)); - return 0; + return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, + e->items, e->texts); } EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); -- cgit v1.1 From 63825f3a879ea2be569471643bb6aac73d9261f0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Oct 2014 12:04:46 +0200 Subject: ALSA: pcm: Disable mmap for known broken archs Some architectures like PARISC is known not to support mmap properly with the DMA buffer, where dma_mmap_coherent() returns -EINVAL unconditionally. From the API POV, we should rather drop the mmap support there and expose it before the user-space tries to call mmap. The patch contains again ugly ifdef's, unfortunately, as there is no global flag indicating this. Once when such macro is defined, we can get rid of this instead. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index aa6754d..dc9a1355 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -192,6 +192,21 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, return err; } +static bool hw_support_mmap(struct snd_pcm_substream *substream) +{ + if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) + return false; + /* check architectures that return -EINVAL from dma_mmap_coherent() */ + /* FIXME: this should be some global flag */ +#if defined(CONFIG_C6X) || defined(CONFIG_FRV) || defined(CONFIG_MN10300) ||\ + defined(CONFIG_PARISC) || defined(CONFIG_XTENSA) + if (!substream->ops->mmap && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return false; +#endif + return true; +} + #undef RULES_DEBUG #ifdef RULES_DEBUG @@ -369,8 +384,12 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, } hw = &substream->runtime->hw; - if (!params->info) + if (!params->info) { params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + if (!hw_support_mmap(substream)) + params->info &= ~(SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID); + } if (!params->fifo_size) { m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); @@ -2069,7 +2088,7 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) mask |= 1 << SNDRV_PCM_ACCESS_RW_INTERLEAVED; if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) mask |= 1 << SNDRV_PCM_ACCESS_RW_NONINTERLEAVED; - if (hw->info & SNDRV_PCM_INFO_MMAP) { + if (hw_support_mmap(substream)) { if (hw->info & SNDRV_PCM_INFO_INTERLEAVED) mask |= 1 << SNDRV_PCM_ACCESS_MMAP_INTERLEAVED; if (hw->info & SNDRV_PCM_INFO_NONINTERLEAVED) -- cgit v1.1 From 9313484238ca49fe5c7513dfcb36aaddcea8c298 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:28 +0200 Subject: ASoC: ak4535: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 30e2978..eced46d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -373,36 +373,19 @@ static struct snd_soc_dai_driver ak4535_dai = { .ops = &ak4535_dai_ops, }; -static int ak4535_suspend(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int ak4535_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } static int ak4535_probe(struct snd_soc_codec *codec) { - /* power on device */ - ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } -/* power down chip */ -static int ak4535_remove(struct snd_soc_codec *codec) -{ - ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static const struct regmap_config ak4535_regmap = { .reg_bits = 8, .val_bits = 8, @@ -417,10 +400,10 @@ static const struct regmap_config ak4535_regmap = { static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .probe = ak4535_probe, - .remove = ak4535_remove, - .suspend = ak4535_suspend, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, -- cgit v1.1 From 4caab4194a99e58c08c70e7df846b9bda948f353 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:29 +0200 Subject: ASoC: ak4535: Use table based setup for controls Makes the code slightly shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index eced46d..9130d91 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -379,13 +379,6 @@ static int ak4535_resume(struct snd_soc_codec *codec) return 0; } -static int ak4535_probe(struct snd_soc_codec *codec) -{ - snd_soc_add_codec_controls(codec, ak4535_snd_controls, - ARRAY_SIZE(ak4535_snd_controls)); - return 0; -} - static const struct regmap_config ak4535_regmap = { .reg_bits = 8, .val_bits = 8, @@ -399,11 +392,12 @@ static const struct regmap_config ak4535_regmap = { }; static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { - .probe = ak4535_probe, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, .suspend_bias_off = true, + .controls = ak4535_snd_controls, + .num_controls = ARRAY_SIZE(ak4535_snd_controls), .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, -- cgit v1.1 From 0b0171e3ad22b5a3be01bbafddede4ebea1769bd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:30 +0200 Subject: ASoC: ak4641: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 33 +-------------------------------- 1 file changed, 1 insertion(+), 32 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7afe8f4..70861c7 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -505,39 +505,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = { }, }; -static int ak4641_suspend(struct snd_soc_codec *codec) -{ - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int ak4641_resume(struct snd_soc_codec *codec) -{ - ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - -static int ak4641_probe(struct snd_soc_codec *codec) -{ - /* power on device */ - ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int ak4641_remove(struct snd_soc_codec *codec) -{ - ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - - static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { - .probe = ak4641_probe, - .remove = ak4641_remove, - .suspend = ak4641_suspend, - .resume = ak4641_resume, .controls = ak4641_snd_controls, .num_controls = ARRAY_SIZE(ak4641_snd_controls), .dapm_widgets = ak4641_dapm_widgets, @@ -545,6 +513,7 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .dapm_routes = ak4641_audio_map, .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), .set_bias_level = ak4641_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config ak4641_regmap = { -- cgit v1.1 From 61ce9ee3aad2fc7a505a420957e8205c4050db69 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:31 +0200 Subject: ASoC: ak4642: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 0417125..dde8b49 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -491,23 +491,7 @@ static int ak4642_resume(struct snd_soc_codec *codec) return 0; } - -static int ak4642_probe(struct snd_soc_codec *codec) -{ - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int ak4642_remove(struct snd_soc_codec *codec) -{ - ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { - .probe = ak4642_probe, - .remove = ak4642_remove, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, .controls = ak4642_snd_controls, -- cgit v1.1 From e48d73c697b77b798a82e86c937fc41e597a1471 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:32 +0200 Subject: ASoC: ak4671: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 998fa0c..686cacb 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = { .ops = &ak4671_dai_ops, }; -static int ak4671_probe(struct snd_soc_codec *codec) -{ - return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int ak4671_remove(struct snd_soc_codec *codec) -{ - ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { - .probe = ak4671_probe, - .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, .controls = ak4671_snd_controls, .num_controls = ARRAY_SIZE(ak4671_snd_controls), -- cgit v1.1 From a613cc4063a315efe36f233006f424e958ef4e67 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:33 +0200 Subject: ASoC: max98088: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 27 ++------------------------- 1 file changed, 2 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 2cd3e54..bb892b3 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1887,25 +1887,6 @@ static void max98088_handle_pdata(struct snd_soc_codec *codec) max98088_handle_eq_pdata(codec); } -#ifdef CONFIG_PM -static int max98088_suspend(struct snd_soc_codec *codec) -{ - max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int max98088_resume(struct snd_soc_codec *codec) -{ - max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define max98088_suspend NULL -#define max98088_resume NULL -#endif - static int max98088_probe(struct snd_soc_codec *codec) { struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); @@ -1946,9 +1927,6 @@ static int max98088_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV); - /* initialize registers cache to hardware default */ - max98088_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_write(codec, M98088_REG_0F_IRQ_ENABLE, 0x00); snd_soc_write(codec, M98088_REG_22_MIX_DAC, @@ -1974,7 +1952,6 @@ static int max98088_remove(struct snd_soc_codec *codec) { struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); - max98088_set_bias_level(codec, SND_SOC_BIAS_OFF); kfree(max98088->eq_texts); return 0; @@ -1983,9 +1960,9 @@ static int max98088_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .probe = max98088_probe, .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, .set_bias_level = max98088_set_bias_level, + .suspend_bias_off = true, + .controls = max98088_snd_controls, .num_controls = ARRAY_SIZE(max98088_snd_controls), .dapm_widgets = max98088_dapm_widgets, -- cgit v1.1 From a8669f60321c8cb08af76438727b6460d1b591b6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:34 +0200 Subject: ASoC: max98095: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0ee6797..42103ca 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2317,9 +2317,6 @@ static int max98095_probe(struct snd_soc_codec *codec) snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV); - /* initialize registers cache to hardware default */ - max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_write(codec, M98095_048_MIX_DAC_LR, M98095_DAI1L_TO_DACL|M98095_DAI1R_TO_DACR); @@ -2359,8 +2356,6 @@ static int max98095_remove(struct snd_soc_codec *codec) struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); struct i2c_client *client = to_i2c_client(codec->dev); - max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (max98095->headphone_jack || max98095->mic_jack) max98095_jack_detect_disable(codec); -- cgit v1.1 From 46804120c59b1374f8beb2b8636ffe6b0a7c16c8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:35 +0200 Subject: ASoC: max9850: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 4fdf5aa..10f8e47 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -291,25 +291,6 @@ static struct snd_soc_dai_driver max9850_dai = { .ops = &max9850_dai_ops, }; -#ifdef CONFIG_PM -static int max9850_suspend(struct snd_soc_codec *codec) -{ - max9850_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int max9850_resume(struct snd_soc_codec *codec) -{ - max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define max9850_suspend NULL -#define max9850_resume NULL -#endif - static int max9850_probe(struct snd_soc_codec *codec) { /* enable zero-detect */ @@ -324,9 +305,8 @@ static int max9850_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .probe = max9850_probe, - .suspend = max9850_suspend, - .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, + .suspend_bias_off = true, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), -- cgit v1.1 From 815b776cf5983ab69d548146fb979adac5dec4de Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:36 +0200 Subject: ASoC: sta32x: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 4874085..7e18200 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -833,23 +833,6 @@ static struct snd_soc_dai_driver sta32x_dai = { .ops = &sta32x_dai_ops, }; -#ifdef CONFIG_PM -static int sta32x_suspend(struct snd_soc_codec *codec) -{ - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int sta32x_resume(struct snd_soc_codec *codec) -{ - sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define sta32x_suspend NULL -#define sta32x_resume NULL -#endif - static int sta32x_probe(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); @@ -936,7 +919,6 @@ static int sta32x_remove(struct snd_soc_codec *codec) struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); sta32x_watchdog_stop(sta32x); - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; @@ -955,9 +937,8 @@ static bool sta32x_reg_is_volatile(struct device *dev, unsigned int reg) static const struct snd_soc_codec_driver sta32x_codec = { .probe = sta32x_probe, .remove = sta32x_remove, - .suspend = sta32x_suspend, - .resume = sta32x_resume, .set_bias_level = sta32x_set_bias_level, + .suspend_bias_off = true, .controls = sta32x_snd_controls, .num_controls = ARRAY_SIZE(sta32x_snd_controls), .dapm_widgets = sta32x_dapm_widgets, -- cgit v1.1 From 2062c1ff3596e1ae8aafe8082460d03d9a420282 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:37 +0200 Subject: ASoC: sta350: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index cc97dd5..bda2ee1 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -912,23 +912,6 @@ static struct snd_soc_dai_driver sta350_dai = { .ops = &sta350_dai_ops, }; -#ifdef CONFIG_PM -static int sta350_suspend(struct snd_soc_codec *codec) -{ - sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int sta350_resume(struct snd_soc_codec *codec) -{ - sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define sta350_suspend NULL -#define sta350_resume NULL -#endif - static int sta350_probe(struct snd_soc_codec *codec) { struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); @@ -1065,7 +1048,6 @@ static int sta350_remove(struct snd_soc_codec *codec) { struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); - sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); return 0; @@ -1074,9 +1056,8 @@ static int sta350_remove(struct snd_soc_codec *codec) static const struct snd_soc_codec_driver sta350_codec = { .probe = sta350_probe, .remove = sta350_remove, - .suspend = sta350_suspend, - .resume = sta350_resume, .set_bias_level = sta350_set_bias_level, + .suspend_bias_off = true, .controls = sta350_snd_controls, .num_controls = ARRAY_SIZE(sta350_snd_controls), .dapm_widgets = sta350_dapm_widgets, -- cgit v1.1 From cfbb77ce368b8d4181e06f8982a440702567eb98 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:38 +0200 Subject: ASoC: sta529: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 35 ++--------------------------------- 1 file changed, 2 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 89c748d..b0f436d 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -319,41 +319,10 @@ static struct snd_soc_dai_driver sta529_dai = { .ops = &sta529_dai_ops, }; -static int sta529_probe(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* power down chip */ -static int sta529_remove(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sta529_suspend(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int sta529_resume(struct snd_soc_codec *codec) -{ - sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static const struct snd_soc_codec_driver sta529_codec_driver = { - .probe = sta529_probe, - .remove = sta529_remove, .set_bias_level = sta529_set_bias_level, - .suspend = sta529_suspend, - .resume = sta529_resume, + .suspend_bias_off = true, + .controls = sta529_snd_controls, .num_controls = ARRAY_SIZE(sta529_snd_controls), }; -- cgit v1.1 From 4c07a43d9691ab1f264337d683dc8655b1edca46 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 10:56:39 +0200 Subject: ASoC: stac9766: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 53b810d..9878534 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -254,12 +254,6 @@ static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) return 0; } -static int stac9766_codec_suspend(struct snd_soc_codec *codec) -{ - stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int stac9766_codec_resume(struct snd_soc_codec *codec) { u16 id, reset; @@ -278,7 +272,6 @@ reset: reset++; goto reset; } - stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -349,8 +342,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) goto codec_err; } - stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); @@ -371,9 +362,9 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { .write = stac9766_ac97_write, .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, + .suspend_bias_off = true, .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, - .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, .reg_cache_size = ARRAY_SIZE(stac9766_reg), .reg_word_size = sizeof(u16), -- cgit v1.1 From 15f6b09a00a6d12b594c439fb3a7e2d113a05592 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 19 Oct 2014 09:07:35 +0200 Subject: ASoC: soc-compress: consolidate two identical branches The actions taken in both branches are identical, so we can simplify the code. Spotted by Coverity. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index cecfab3c..590a82f 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -258,10 +258,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); - else - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; @@ -456,11 +453,7 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, if (ret < 0) goto out; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); - else - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); - + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- cgit v1.1 From 5e3363ad1b7b2e1f197a3f56b01e21cb155ad454 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 16 Oct 2014 11:24:26 -0700 Subject: ASoC: rt5677: add GPIO IRQ support This allows to enable Mic Jack detection feature Signed-off-by: Oder Chiou Modified-by: Anatol Pomozov Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5677.txt | 10 ++ include/sound/rt5677.h | 7 ++ sound/soc/codecs/rt5677.c | 134 +++++++++++++++++++++ sound/soc/codecs/rt5677.h | 49 ++++++++ 4 files changed, 200 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f82f0e9..740ff77 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -33,6 +33,15 @@ Optional properties: 1 - pull down 2 - pull up +- realtek,jd1-gpio + Configures GPIO Mic Jack detection 1. + Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively. + +- realtek,jd2-gpio +- realtek,jd3-gpio + Configures GPIO Mic Jack detection 2 and 3. + Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively. + Pins on the device (for linking into audio routes): * IN1P @@ -63,4 +72,5 @@ rt5677 { <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; realtek,in1-differential = "true"; realtek,gpio-config = /bits/ 8 <0 0 0 0 0 2>; /* pull up GPIO6 */ + realtek,jd2-gpio = <3>; /* Enables Jack detection for GPIO6 */ }; diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h index a56b429..d9eb7d8 100644 --- a/include/sound/rt5677.h +++ b/include/sound/rt5677.h @@ -30,6 +30,13 @@ struct rt5677_platform_data { /* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */ u8 gpio_config[6]; + + /* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */ + unsigned int jd1_gpio; + /* jd2 and jd3 can select 0 ~ 3 as + OFF, GPIO4, GPIO5 and GPIO6 respectively */ + unsigned int jd2_gpio; + unsigned int jd3_gpio; }; #endif diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d17d079..6c73dfd 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3614,6 +3614,46 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, } } +static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) +{ + struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct regmap_irq_chip_data *data = rt5677->irq_data; + int irq; + + if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) { + if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) || + (rt5677->pdata.jd1_gpio == 2 && + offset == RT5677_GPIO2) || + (rt5677->pdata.jd1_gpio == 3 && + offset == RT5677_GPIO3)) { + irq = RT5677_IRQ_JD1; + } else { + return -ENXIO; + } + } + + if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) { + if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) || + (rt5677->pdata.jd2_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd2_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD2; + } else if ((rt5677->pdata.jd3_gpio == 1 && + offset == RT5677_GPIO4) || + (rt5677->pdata.jd3_gpio == 2 && + offset == RT5677_GPIO5) || + (rt5677->pdata.jd3_gpio == 3 && + offset == RT5677_GPIO6)) { + irq = RT5677_IRQ_JD3; + } else { + return -ENXIO; + } + } + + return regmap_irq_get_virq(data, irq); +} + static struct gpio_chip rt5677_template_chip = { .label = "rt5677", .owner = THIS_MODULE, @@ -3621,6 +3661,7 @@ static struct gpio_chip rt5677_template_chip = { .set = rt5677_gpio_set, .direction_input = rt5677_gpio_direction_in, .get = rt5677_gpio_get, + .to_irq = rt5677_to_irq, .can_sleep = 1, }; @@ -3685,6 +3726,31 @@ static int rt5677_probe(struct snd_soc_codec *codec) for (i = 0; i < RT5677_GPIO_NUM; i++) rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]); + if (rt5677->irq_data) { + regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, 0x8000, + 0x8000); + regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x0018, + 0x0008); + + if (rt5677->pdata.jd1_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD1_MASK, + rt5677->pdata.jd1_gpio << + RT5677_SEL_GPIO_JD1_SFT); + + if (rt5677->pdata.jd2_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD2_MASK, + rt5677->pdata.jd2_gpio << + RT5677_SEL_GPIO_JD2_SFT); + + if (rt5677->pdata.jd3_gpio) + regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1, + RT5677_SEL_GPIO_JD3_MASK, + rt5677->pdata.jd3_gpio << + RT5677_SEL_GPIO_JD3_SFT); + } + mutex_init(&rt5677->dsp_cmd_lock); return 0; @@ -3915,9 +3981,74 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) of_property_read_u8_array(np, "realtek,gpio-config", rt5677->pdata.gpio_config, RT5677_GPIO_NUM); + of_property_read_u32(np, "realtek,jd1-gpio", &rt5677->pdata.jd1_gpio); + of_property_read_u32(np, "realtek,jd2-gpio", &rt5677->pdata.jd2_gpio); + of_property_read_u32(np, "realtek,jd3-gpio", &rt5677->pdata.jd3_gpio); + return 0; } +static struct regmap_irq rt5677_irqs[] = { + [RT5677_IRQ_JD1] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD1, + }, + [RT5677_IRQ_JD2] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD2, + }, + [RT5677_IRQ_JD3] = { + .reg_offset = 0, + .mask = RT5677_EN_IRQ_GPIO_JD3, + }, +}; + +static struct regmap_irq_chip rt5677_irq_chip = { + .name = "rt5677", + .irqs = rt5677_irqs, + .num_irqs = ARRAY_SIZE(rt5677_irqs), + + .num_regs = 1, + .status_base = RT5677_IRQ_CTRL1, + .mask_base = RT5677_IRQ_CTRL1, + .mask_invert = 1, +}; + +int rt5677_irq_init(struct i2c_client *i2c) +{ + int ret; + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (!rt5677->pdata.jd1_gpio && + !rt5677->pdata.jd2_gpio && + !rt5677->pdata.jd3_gpio) + return 0; + + if (!i2c->irq) { + dev_err(&i2c->dev, "No interrupt specified\n"); + return -EINVAL; + } + + ret = regmap_add_irq_chip(rt5677->regmap, i2c->irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, 0, + &rt5677_irq_chip, &rt5677->irq_data); + + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register IRQ chip: %d\n", ret); + return ret; + } + + return 0; +} + +void rt5677_irq_exit(struct i2c_client *i2c) +{ + struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); + + if (rt5677->irq_data) + regmap_del_irq_chip(i2c->irq, rt5677->irq_data); +} + static int rt5677_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -4015,6 +4146,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, } rt5677_init_gpio(i2c); + rt5677_irq_init(i2c); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); @@ -4022,6 +4154,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { + rt5677_irq_exit(i2c); + snd_soc_unregister_codec(&i2c->dev); rt5677_free_gpio(i2c); diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 20efa4a..d2c743c 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1368,6 +1368,48 @@ #define RT5677_SEL_SRC_IB01 (0x1 << 0) #define RT5677_SEL_SRC_IB01_SFT 0 +/* Jack Detect Control 1 (0xb5) */ +#define RT5677_SEL_GPIO_JD1_MASK (0x3 << 14) +#define RT5677_SEL_GPIO_JD1_SFT 14 +#define RT5677_SEL_GPIO_JD2_MASK (0x3 << 12) +#define RT5677_SEL_GPIO_JD2_SFT 12 +#define RT5677_SEL_GPIO_JD3_MASK (0x3 << 10) +#define RT5677_SEL_GPIO_JD3_SFT 10 + +/* IRQ Control 1 (0xbd) */ +#define RT5677_STA_GPIO_JD1 (0x1 << 15) +#define RT5677_STA_GPIO_JD1_SFT 15 +#define RT5677_EN_IRQ_GPIO_JD1 (0x1 << 14) +#define RT5677_EN_IRQ_GPIO_JD1_SFT 14 +#define RT5677_EN_GPIO_JD1_STICKY (0x1 << 13) +#define RT5677_EN_GPIO_JD1_STICKY_SFT 13 +#define RT5677_INV_GPIO_JD1 (0x1 << 12) +#define RT5677_INV_GPIO_JD1_SFT 12 +#define RT5677_STA_GPIO_JD2 (0x1 << 11) +#define RT5677_STA_GPIO_JD2_SFT 11 +#define RT5677_EN_IRQ_GPIO_JD2 (0x1 << 10) +#define RT5677_EN_IRQ_GPIO_JD2_SFT 10 +#define RT5677_EN_GPIO_JD2_STICKY (0x1 << 9) +#define RT5677_EN_GPIO_JD2_STICKY_SFT 9 +#define RT5677_INV_GPIO_JD2 (0x1 << 8) +#define RT5677_INV_GPIO_JD2_SFT 8 +#define RT5677_STA_MICBIAS1_OVCD (0x1 << 7) +#define RT5677_STA_MICBIAS1_OVCD_SFT 7 +#define RT5677_EN_IRQ_MICBIAS1_OVCD (0x1 << 6) +#define RT5677_EN_IRQ_MICBIAS1_OVCD_SFT 6 +#define RT5677_EN_MICBIAS1_OVCD_STICKY (0x1 << 5) +#define RT5677_EN_MICBIAS1_OVCD_STICKY_SFT 5 +#define RT5677_INV_MICBIAS1_OVCD (0x1 << 4) +#define RT5677_INV_MICBIAS1_OVCD_SFT 4 +#define RT5677_STA_GPIO_JD3 (0x1 << 3) +#define RT5677_STA_GPIO_JD3_SFT 3 +#define RT5677_EN_IRQ_GPIO_JD3 (0x1 << 2) +#define RT5677_EN_IRQ_GPIO_JD3_SFT 2 +#define RT5677_EN_GPIO_JD3_STICKY (0x1 << 1) +#define RT5677_EN_GPIO_JD3_STICKY_SFT 1 +#define RT5677_INV_GPIO_JD3 (0x1 << 0) +#define RT5677_INV_GPIO_JD3_SFT 0 + /* GPIO status (0xbf) */ #define RT5677_GPIO6_STATUS_MASK (0x1 << 5) #define RT5677_GPIO6_STATUS_SFT 5 @@ -1545,6 +1587,12 @@ enum { RT5677_GPIO_NUM, }; +enum { + RT5677_IRQ_JD1, + RT5677_IRQ_JD2, + RT5677_IRQ_JD3, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1565,6 +1613,7 @@ struct rt5677_priv { struct gpio_chip gpio_chip; #endif bool dsp_vad_en; + struct regmap_irq_chip_data *irq_data; }; #endif /* __RT5677_H__ */ -- cgit v1.1 From cdc4508b4d1c609e3b0e4f23697edbee0d23b86e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 19:36:33 +0200 Subject: ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() Each widget has a list of all the paths that it is connected to. There is no need to iterate over all paths when we are only interested in the paths of a specific widget. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 65 +++++++++++++++++++++++++++++++++------------------- 1 file changed, 42 insertions(+), 23 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 39f992b..2c4bfdb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3788,35 +3788,54 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); +/** + * dapm_is_external_path() - Checks if a path is a external path + * @card: The card the path belongs to + * @path: The path to check + * + * Returns true if the path is either between two different DAPM contexts or + * between two external pins of the same DAPM context. Otherwise returns + * false. + */ +static bool dapm_is_external_path(struct snd_soc_card *card, + struct snd_soc_dapm_path *path) +{ + dev_dbg(card->dev, + "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n", + path->source->name, path->source->id, path->source->dapm, + path->sink->name, path->sink->id, path->sink->dapm); + + /* Connection between two different DAPM contexts */ + if (path->source->dapm != path->sink->dapm) + return true; + + /* Loopback connection from external pin to external pin */ + if (path->sink->id == snd_soc_dapm_input) { + switch (path->source->id) { + case snd_soc_dapm_output: + case snd_soc_dapm_micbias: + return true; + default: + break; + } + } + + return false; +} + static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card, struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_path *p; - list_for_each_entry(p, &card->paths, list) { - if ((p->source == w) || (p->sink == w)) { - dev_dbg(card->dev, - "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n", - p->source->name, p->source->id, p->source->dapm, - p->sink->name, p->sink->id, p->sink->dapm); + list_for_each_entry(p, &w->sources, list_sink) { + if (dapm_is_external_path(card, p)) + return true; + } - /* Connected to something other than the codec */ - if (p->source->dapm != p->sink->dapm) - return true; - /* - * Loopback connection from codec external pin to - * codec external pin - */ - if (p->sink->id == snd_soc_dapm_input) { - switch (p->source->id) { - case snd_soc_dapm_output: - case snd_soc_dapm_micbias: - return true; - default: - break; - } - } - } + list_for_each_entry(p, &w->sinks, list_source) { + if (dapm_is_external_path(card, p)) + return true; } return false; -- cgit v1.1 From 7ddd4cd5c31ccaf32febe52462f9fdc915893212 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 19:36:34 +0200 Subject: ASoC: dapm: Remove always true path source/sink checks A path has always a valid source and a valid sink otherwise we wouldn't add it in the first place. Hence all tests that check if sink/source is non NULL always evaluate to true and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c4bfdb..28269f2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -909,7 +909,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, trace_snd_soc_dapm_output_path(widget, path); - if (path->sink && path->connect) { + if (path->connect) { path->walked = 1; path->walking = 1; @@ -1017,7 +1017,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, trace_snd_soc_dapm_input_path(widget, path); - if (path->source && path->connect) { + if (path->connect) { path->walked = 1; path->walking = 1; @@ -1219,9 +1219,6 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) !path->connected(path->source, path->sink)) continue; - if (!path->sink) - continue; - if (dapm_widget_power_check(path->sink)) return 1; } @@ -1636,12 +1633,9 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, /* If we changed our power state perhaps our neigbours changed * also. */ - list_for_each_entry(path, &w->sources, list_sink) { - if (path->source) { - dapm_widget_set_peer_power(path->source, power, - path->connect); - } - } + list_for_each_entry(path, &w->sources, list_sink) + dapm_widget_set_peer_power(path->source, power, path->connect); + switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: @@ -1650,12 +1644,9 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, /* Supplies can't affect their outputs, only their inputs */ break; default: - list_for_each_entry(path, &w->sinks, list_source) { - if (path->sink) { - dapm_widget_set_peer_power(path->sink, power, - path->connect); - } - } + list_for_each_entry(path, &w->sinks, list_source) + dapm_widget_set_peer_power(path->sink, power, + path->connect); break; } -- cgit v1.1 From cdef2ad3ae64cc1ab2daeff26335e0dde988eed7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 19:36:38 +0200 Subject: ASoC: dapm: Remove special DAI widget power check functions dapm_adc_check_power() checks if the widget is active, if yes it only checks whether there are any connected input paths. Otherwise it calls dapm_generic_check_power() which will check for both connected input and output paths. But the function that checks for connected output paths will return true if the widget is a active sink. Which means the generic power check function will work just fine and there is no need for a special power check function. The same applies for dapm_dac_check_power(), but with input and output paths reversed. This patch removes both dapm_adc_check_power() and dapm_dac_check_power() and replace their usage with dapm_generic_check_power(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 40 ++-------------------------------------- 1 file changed, 2 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 28269f2..219d73c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1169,38 +1169,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -/* Check to see if an ADC has power */ -static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) -{ - int in; - - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) { - in = is_connected_input_ep(w, NULL); - dapm_clear_walk_input(w->dapm, &w->sources); - return in != 0; - } else { - return dapm_generic_check_power(w); - } -} - -/* Check to see if a DAC has power */ -static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) -{ - int out; - - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) { - out = is_connected_output_ep(w, NULL); - dapm_clear_walk_output(w->dapm, &w->sinks); - return out != 0; - } else { - return dapm_generic_check_power(w); - } -} - /* Check to see if a power supply is needed */ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) { @@ -3086,12 +3054,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_mux: w->power_check = dapm_generic_check_power; break; - case snd_soc_dapm_dai_out: - w->power_check = dapm_adc_check_power; - break; - case snd_soc_dapm_dai_in: - w->power_check = dapm_dac_check_power; - break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: case snd_soc_dapm_dac: @@ -3106,6 +3068,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_mic: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_dai_out: + case snd_soc_dapm_dai_in: w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: -- cgit v1.1 From 130897ac5ac03adb4604d27497c378c64c7b22dd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 20 Oct 2014 19:36:39 +0200 Subject: ASoC: dapm: Remove path 'walked' flag The 'walked' flag was used to avoid walking paths that have already been walked. But since we started caching the number of inputs and outputs of a path we never actually get into a situation where we try to walk a path that has the 'walked' flag set. There are two cases in which we can end up walking a path multiple times within a single run of is_connected_output_ep() or is_connected_input_ep(). 1) If a path splits up and rejoins later: .--> C ---v A -> B E --> F '--> D ---^ When walking from A to F we'll end up at E twice, once via C and once via D. But since we do a depth first search we'll fully discover the path and initialize the number of outputs/inputs of the widget the first time we get there. The second time we get there we'll use the cached value and not bother to check any of the paths again. So we'll never see a path where 'walked' is set in this case. 2) If there is a circle: A --> B <-- C <-.--> F '--> D ---' When walking from A to F we'll end up twice at B. But since there is a circle the 'walking' flag will still be set on B once we get there the second time. This means we won't look at any of it's outgoing paths. So in this case we won't ever see a path where 'walked' is set either. So it is safe to remove the flag. This on one hand means we remove some always true checks from one of the hottest paths of the DAPM algorithm and on the other hand means we do not have to do the tedious clearing of the flag after checking the number inputs or outputs of a widget. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 - sound/soc/soc-dapm.c | 49 ++---------------------------------------------- 2 files changed, 2 insertions(+), 48 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..ebb93f2 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -508,7 +508,6 @@ struct snd_soc_dapm_path { /* status */ u32 connect:1; /* source and sink widgets are connected */ - u32 walked:1; /* path has been walked */ u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 219d73c..f03e0cf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,34 +754,6 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } -/* reset 'walked' bit for each dapm path */ -static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, - struct list_head *sink) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, sink, list_source) { - if (p->walked) { - p->walked = 0; - dapm_clear_walk_output(dapm, &p->sink->sinks); - } - } -} - -static void dapm_clear_walk_input(struct snd_soc_dapm_context *dapm, - struct list_head *source) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, source, list_sink) { - if (p->walked) { - p->walked = 0; - dapm_clear_walk_input(dapm, &p->source->sources); - } - } -} - - /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -904,13 +876,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->walking) return 1; - if (path->walked) - continue; - trace_snd_soc_dapm_output_path(widget, path); if (path->connect) { - path->walked = 1; path->walking = 1; /* do we need to add this widget to the list ? */ @@ -1012,13 +980,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->walking) return 1; - if (path->walked) - continue; - trace_snd_soc_dapm_input_path(widget, path); if (path->connect) { - path->walked = 1; path->walking = 1; /* do we need to add this widget to the list ? */ @@ -1066,15 +1030,10 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); dapm_reset(card); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (stream == SNDRV_PCM_STREAM_PLAYBACK) paths = is_connected_output_ep(dai->playback_widget, list); - dapm_clear_walk_output(&card->dapm, - &dai->playback_widget->sinks); - } else { + else paths = is_connected_input_ep(dai->capture_widget, list); - dapm_clear_walk_input(&card->dapm, - &dai->capture_widget->sources); - } trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); @@ -1163,9 +1122,7 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); in = is_connected_input_ep(w, NULL); - dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0 && in != 0; } @@ -1823,9 +1780,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, return -ENOMEM; in = is_connected_input_ep(w, NULL); - dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk_output(w->dapm, &w->sinks); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", -- cgit v1.1 From 2d27deb40db74c751c991e96ca91d675f966a0c5 Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Wed, 22 Oct 2014 20:04:08 +0800 Subject: ASoC: rt5677: rt5677_irq_init() can be static sound/soc/codecs/rt5677.c:4017:5: sparse: symbol 'rt5677_irq_init' was not declared. Should it be static? sound/soc/codecs/rt5677.c:4044:6: sparse: symbol 'rt5677_irq_exit' was not declared. Should it be static? Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 6c73dfd..413bccb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4014,7 +4014,7 @@ static struct regmap_irq_chip rt5677_irq_chip = { .mask_invert = 1, }; -int rt5677_irq_init(struct i2c_client *i2c) +static int rt5677_irq_init(struct i2c_client *i2c) { int ret; struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); @@ -4041,7 +4041,7 @@ int rt5677_irq_init(struct i2c_client *i2c) return 0; } -void rt5677_irq_exit(struct i2c_client *i2c) +static void rt5677_irq_exit(struct i2c_client *i2c) { struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); -- cgit v1.1 From ace0eb1e91a75b84b1be3d610b79509a5bd94df1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 21 Oct 2014 18:13:46 -0700 Subject: ASoC: rsnd: tidyup debug information when read/write b8c637864a6904a9ba8e0df556d5bdf9f26b2c54 (ASoC: rsnd: use regmap_mmio instead of original regmap bus) added regmap_mmio support on Renesas R-Car sound driver. Then, debug information of register read/write indicates regmap index, not register address. This is a little bit confusable information. This patch tidyup debug message, and added regmap debug hint on comment area. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index f95e7ab..61dee68 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -8,6 +8,17 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ + +/* + * #define DEBUG + * + * you can also add below in + * ${LINUX}/drivers/base/regmap/regmap.c + * for regmap debug + * + * #define LOG_DEVICE "xxxx.rcar_sound" + */ + #include "rsnd.h" struct rsnd_gen { @@ -67,9 +78,10 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + dev_dbg(dev, "r %s(%d) - %4d : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); - dev_dbg(dev, "r %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, val); + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); return val; } @@ -84,9 +96,10 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); + dev_dbg(dev, "w %s(%d) - %4d : %08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); - dev_dbg(dev, "w %s - 0x%04d : %08x\n", rsnd_mod_name(mod), reg, data); + regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); } void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, @@ -98,11 +111,11 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; + dev_dbg(dev, "b %s(%d) - %4d : %08x/%08x\n", + rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data, mask); + regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), mask, data); - - dev_dbg(dev, "b %s - 0x%04d : %08x/%08x\n", - rsnd_mod_name(mod), reg, data, mask); } #define rsnd_gen_regmap_init(priv, id_size, reg_id, conf) \ -- cgit v1.1 From 9960ce97432bdb1defc76ed80ac19e37e8778bc6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 21 Oct 2014 18:13:56 -0700 Subject: ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS RSND_DVC_VOLUME_NUM means DVC channel number. This patch tidyups this un-understandable naming Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 3f44393..b5f95ad4 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -12,7 +12,7 @@ #define RSND_DVC_NAME_SIZE 16 #define RSND_DVC_VOLUME_MAX 100 -#define RSND_DVC_VOLUME_NUM 2 +#define RSND_DVC_CHANNELS 2 #define DVC_NAME "dvc" @@ -20,8 +20,8 @@ struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - u8 volume[RSND_DVC_VOLUME_NUM]; - u8 mute[RSND_DVC_VOLUME_NUM]; + u8 volume[RSND_DVC_CHANNELS]; + u8 mute[RSND_DVC_CHANNELS]; }; #define rsnd_mod_to_dvc(_mod) \ @@ -37,11 +37,11 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) { struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); u32 max = (0x00800000 - 1); - u32 vol[RSND_DVC_VOLUME_NUM]; + u32 vol[RSND_DVC_CHANNELS]; u32 mute = 0; int i; - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { + for (i = 0; i < RSND_DVC_CHANNELS; i++) { vol[i] = max / RSND_DVC_VOLUME_MAX * dvc->volume[i]; mute |= (!!dvc->mute[i]) << i; } @@ -150,7 +150,7 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); u8 *val = (u8 *)kctrl->private_value; - uinfo->count = RSND_DVC_VOLUME_NUM; + uinfo->count = RSND_DVC_CHANNELS; uinfo->value.integer.min = 0; if (val == dvc->volume) { @@ -170,7 +170,7 @@ static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, u8 *val = (u8 *)kctrl->private_value; int i; - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) + for (i = 0; i < RSND_DVC_CHANNELS; i++) ucontrol->value.integer.value[i] = val[i]; return 0; @@ -183,7 +183,7 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, u8 *val = (u8 *)kctrl->private_value; int i, change = 0; - for (i = 0; i < RSND_DVC_VOLUME_NUM; i++) { + for (i = 0; i < RSND_DVC_CHANNELS; i++) { change |= (ucontrol->value.integer.value[i] != val[i]); val[i] = ucontrol->value.integer.value[i]; } -- cgit v1.1 From 92b9a6991b2e3a4ccf5ffc956730d36835d53a79 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 21 Oct 2014 18:14:14 -0700 Subject: ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings DVC can control Digital Volume / Mute / Volume Ramp etc, and these uses different max value. Current driver is using fixed max value for each settings. This patch adds new struct rsnd_dvc_cfg, and control these. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 51 ++++++++++++++++++++++++------------------------- 1 file changed, 25 insertions(+), 26 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index b5f95ad4..deaf0fa 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -11,17 +11,21 @@ #include "rsnd.h" #define RSND_DVC_NAME_SIZE 16 -#define RSND_DVC_VOLUME_MAX 100 #define RSND_DVC_CHANNELS 2 #define DVC_NAME "dvc" +struct rsnd_dvc_cfg { + unsigned int max; + u32 val[RSND_DVC_CHANNELS]; +}; + struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - u8 volume[RSND_DVC_CHANNELS]; - u8 mute[RSND_DVC_CHANNELS]; + struct rsnd_dvc_cfg volume; + struct rsnd_dvc_cfg mute; }; #define rsnd_mod_to_dvc(_mod) \ @@ -36,18 +40,15 @@ struct rsnd_dvc { static void rsnd_dvc_volume_update(struct rsnd_mod *mod) { struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); - u32 max = (0x00800000 - 1); - u32 vol[RSND_DVC_CHANNELS]; u32 mute = 0; int i; for (i = 0; i < RSND_DVC_CHANNELS; i++) { - vol[i] = max / RSND_DVC_VOLUME_MAX * dvc->volume[i]; - mute |= (!!dvc->mute[i]) << i; + mute |= (!!dvc->mute.val[i]) << i; } - rsnd_mod_write(mod, DVC_VOL0R, vol[0]); - rsnd_mod_write(mod, DVC_VOL1R, vol[1]); + rsnd_mod_write(mod, DVC_VOL0R, dvc->volume.val[0]); + rsnd_mod_write(mod, DVC_VOL1R, dvc->volume.val[1]); rsnd_mod_write(mod, DVC_ZCMCR, mute); } @@ -146,20 +147,16 @@ static int rsnd_dvc_stop(struct rsnd_mod *mod, static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, struct snd_ctl_elem_info *uinfo) { - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); - u8 *val = (u8 *)kctrl->private_value; + struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; uinfo->count = RSND_DVC_CHANNELS; uinfo->value.integer.min = 0; + uinfo->value.integer.max = cfg->max; - if (val == dvc->volume) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.max = RSND_DVC_VOLUME_MAX; - } else { + if (cfg->max == 1) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->value.integer.max = 1; - } + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; return 0; } @@ -167,11 +164,11 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { - u8 *val = (u8 *)kctrl->private_value; + struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; int i; for (i = 0; i < RSND_DVC_CHANNELS; i++) - ucontrol->value.integer.value[i] = val[i]; + ucontrol->value.integer.value[i] = cfg->val[i]; return 0; } @@ -180,12 +177,12 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, struct snd_ctl_elem_value *ucontrol) { struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - u8 *val = (u8 *)kctrl->private_value; + struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; int i, change = 0; for (i = 0; i < RSND_DVC_CHANNELS; i++) { - change |= (ucontrol->value.integer.value[i] != val[i]); - val[i] = ucontrol->value.integer.value[i]; + change |= (ucontrol->value.integer.value[i] != cfg->val[i]); + cfg->val[i] = ucontrol->value.integer.value[i]; } if (change) @@ -198,7 +195,7 @@ static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd, const unsigned char *name, - u8 *private) + struct rsnd_dvc_cfg *private) { struct snd_card *card = rtd->card->snd_card; struct snd_kcontrol *kctrl; @@ -232,18 +229,20 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int ret; /* Volume */ + dvc->volume.max = 0x00800000 - 1; ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Playback Volume" : "DVC In Capture Volume", - dvc->volume); + &dvc->volume); if (ret < 0) return ret; /* Mute */ + dvc->mute.max = 1; ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Mute Switch" : "DVC In Mute Switch", - dvc->mute); + &dvc->mute); if (ret < 0) return ret; -- cgit v1.1 From 00d647b081b5ef2193fd15910fbd103f483a5d15 Mon Sep 17 00:00:00 2001 From: Alexandre Courbot Date: Thu, 23 Oct 2014 17:15:18 +0900 Subject: ASoC: jack: update calls to gpiod_get*() Add the new flags argument to calls of (devm_)gpiod_get*() and remove any direction setting code afterwards. Currently both forms (with or without the flags argument) are valid thanks to transitional macros in . These macros will be removed once all consumers are updated and the flags argument will become compulsary. Signed-off-by: Alexandre Courbot Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea..f921d00 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -309,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, /* GPIO descriptor */ gpios[i].desc = gpiod_get_index(gpios[i].gpiod_dev, gpios[i].name, - gpios[i].idx); + gpios[i].idx, GPIOD_IN); if (IS_ERR(gpios[i].desc)) { ret = PTR_ERR(gpios[i].desc); dev_err(gpios[i].gpiod_dev, @@ -327,17 +327,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto undo; } - ret = gpio_request(gpios[i].gpio, gpios[i].name); + ret = gpio_request_one(gpios[i].gpio, GPIOF_IN, + gpios[i].name); if (ret) goto undo; gpios[i].desc = gpio_to_desc(gpios[i].gpio); } - ret = gpiod_direction_input(gpios[i].desc); - if (ret) - goto err; - INIT_DELAYED_WORK(&gpios[i].work, gpio_work); gpios[i].jack = jack; -- cgit v1.1 From e29bee098ea1cc9b8537628f3c1cdf60ead83514 Mon Sep 17 00:00:00 2001 From: Ben Zhang Date: Mon, 20 Oct 2014 20:30:13 -0700 Subject: ASoC: rt5677: fix rt5677 spi driver build Create a separate module for rt5677 spi driver. Without this patch, the build fails due to multiple defs of 'init_module' and 'cleanup_module'. module_spi_driver() defines its own module, so it can't be part of the rt5677 module. Signed-off-by: Ben Zhang Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++++ sound/soc/codecs/Makefile | 4 +++- sound/soc/codecs/rt5677-spi.c | 2 ++ 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2c7482e..6f21a76 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -504,6 +504,10 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate +config SND_SOC_RT5677_SPI + tristate + default SND_SOC_RT5677 + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4435f9f..3e57edc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -79,7 +79,8 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o rt5677-spi.o +snd-soc-rt5677-objs := rt5677.o +snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -256,6 +257,7 @@ obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o +obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 11c38f3..ef6348c 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -52,6 +52,7 @@ int rt5677_spi_write(u8 *txbuf, size_t len) return status; } +EXPORT_SYMBOL_GPL(rt5677_spi_write); /** * rt5677_spi_burst_write - Write data to SPI by rt5677 dsp memory address. @@ -107,6 +108,7 @@ int rt5677_spi_burst_write(u32 addr, const struct firmware *fw) return 0; } +EXPORT_SYMBOL_GPL(rt5677_spi_burst_write); static int rt5677_spi_probe(struct spi_device *spi) { -- cgit v1.1 From 49d776ffb50f2e428aafb6a6576e58e80f1e886c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Oct 2014 12:36:23 +0200 Subject: ALSA: pcm: Avoid mmap warnings on x86 On x86, using dma_mmap_coherent() for the pages allocated via dma_alloc_coherent() results in a warning like: aplay:32536 map pfn RAM range req uncached-minus for [mem 0x21d500000-0x21d51ffff], got write-back Until the issue is addressed in the core side, take back to the old good way in PCM code only for x86. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dc9a1355..03e1e92 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3326,6 +3326,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ +#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3333,6 +3334,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); +#endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- cgit v1.1 From 66797f36fd17e8975f4a3449aed895cda952c0ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Oct 2014 15:15:44 +0100 Subject: ALSA: hda - Pass printf argument directly to request_module() request_module() handles the printf style arguments, so we don't have to render strings in the caller side. Not only it reduces the unnecessary temporary string buffer, it's even safer from the security POV. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 259fbea..0025bf4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -966,14 +966,12 @@ find_codec_preset(struct hda_codec *codec) mutex_unlock(&preset_mutex); if (mod_requested < HDA_MODREQ_MAX_COUNT) { - char name[32]; if (!mod_requested) - snprintf(name, sizeof(name), "snd-hda-codec-id:%08x", - codec->vendor_id); + request_module("snd-hda-codec-id:%08x", + codec->vendor_id); else - snprintf(name, sizeof(name), "snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - request_module(name); + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); mod_requested++; goto again; } -- cgit v1.1 From 39581031a90d69e4b79cd044756169ff35ecab46 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 24 Oct 2014 13:49:46 +0530 Subject: ASoC: Intel: mrfld: Replace pci_id with unique device id In order to support both ACPI and PCI devices we need to use a genric device id in driver, so change all pci_id instances to device_id Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 10 +++++----- sound/soc/intel/sst/sst.h | 5 +++-- sound/soc/intel/sst/sst_pvt.c | 4 ++-- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index d88cdd9..fa34217 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -167,7 +167,7 @@ static struct intel_sst_ops mrfld_ops = { int sst_driver_ops(struct intel_sst_drv *sst) { - switch (sst->pci_id) { + switch (sst->dev_id) { case SST_MRFLD_PCI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; @@ -175,7 +175,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) default: dev_err(sst->dev, - "SST Driver capablities missing for pci_id: %x", sst->pci_id); + "SST Driver capablities missing for dev_id: %x", sst->dev_id); return -EINVAL; }; } @@ -210,7 +210,7 @@ static int intel_sst_probe(struct pci_dev *pci, return -ENOMEM; sst_drv_ctx->dev = &pci->dev; - sst_drv_ctx->pci_id = pci->device; + sst_drv_ctx->dev_id = pci->device; if (!sst_pdata) return -EINVAL; @@ -278,7 +278,7 @@ static int intel_sst_probe(struct pci_dev *pci, /* map registers */ /* DDR base */ - if (sst_drv_ctx->pci_id == SST_MRFLD_PCI_ID) { + if (sst_drv_ctx->dev_id == SST_MRFLD_PCI_ID) { sst_drv_ctx->ddr_base = pci_resource_start(pci, 0); /* check that the relocated IMR base matches with FW Binary */ ddr_base = relocate_imr_addr_mrfld(sst_drv_ctx->ddr_base); @@ -357,7 +357,7 @@ static int intel_sst_probe(struct pci_dev *pci, dev_dbg(sst_drv_ctx->dev, "Registered IRQ 0x%x\n", pci->irq); /* default intr are unmasked so set this as masked */ - if (sst_drv_ctx->pci_id == SST_MRFLD_PCI_ID) + if (sst_drv_ctx->dev_id == SST_MRFLD_PCI_ID) sst_shim_write64(sst_drv_ctx->shim, SST_IMRX, 0xFFFF0038); pci_set_drvdata(pci, sst_drv_ctx); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index bfcf51a..b65b9c0 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,7 +337,8 @@ struct sst_shim_regs64 { * struct intel_sst_drv - driver ops * * @sst_state : current sst device state - * @pci_id : PCI device id loaded + * @dev_id : device identifier, pci_id for pci devices and acpi_id for acpi + * devices * @shim : SST shim pointer * @mailbox : SST mailbox pointer * @iram : SST IRAM pointer @@ -371,7 +372,7 @@ struct sst_shim_regs64 { struct intel_sst_drv { int sst_state; int irq_num; - unsigned int pci_id; + unsigned int dev_id; void __iomem *ddr; void __iomem *shim; void __iomem *mailbox; diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 9e5f69b..1c2e081 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -115,7 +115,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) { unsigned long long val = 0; - switch (sst->pci_id) { + switch (sst->dev_id) { case SST_MRFLD_PCI_ID: val = sst_shim_read64(sst->shim, addr); break; @@ -126,7 +126,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) void write_shim_data(struct intel_sst_drv *sst, int addr, unsigned long long data) { - switch (sst->pci_id) { + switch (sst->dev_id) { case SST_MRFLD_PCI_ID: sst_shim_write64(sst->shim, addr, (u64) data); break; -- cgit v1.1 From 43c5e23197a187a6f4dade97f3bd23e35636ab1f Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 24 Oct 2014 13:49:47 +0530 Subject: ASoC: Intel: mrfld - Define ipc_info structure This will be used to abstract the differances in ipc offsets for different chips. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- arch/x86/include/asm/platform_sst_audio.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 268a96ae..6021dee 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -102,6 +102,11 @@ struct sst_lib_dnld_info { bool mod_ddr_dnld; }; +struct sst_ipc_info { + int ipc_offset; + unsigned int mbox_recv_off; +}; + struct sst_platform_info { const struct sst_info *probe_data; const struct sst_ipc_info *ipc_info; -- cgit v1.1 From 9a80e8f597f3bde0e1d4a4abb021d475520005a5 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Fri, 24 Oct 2014 13:49:48 +0530 Subject: ASoC: Intel: mrfld: Define sst_res_info for acpi To query resources in ACPI systems we need to define a data structure. This would be set as platform_info for the devices. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- arch/x86/include/asm/platform_sst_audio.h | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h index 6021dee..7249e6d 100644 --- a/arch/x86/include/asm/platform_sst_audio.h +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -102,6 +102,27 @@ struct sst_lib_dnld_info { bool mod_ddr_dnld; }; +struct sst_res_info { + unsigned int shim_offset; + unsigned int shim_size; + unsigned int shim_phy_addr; + unsigned int ssp0_offset; + unsigned int ssp0_size; + unsigned int dma0_offset; + unsigned int dma0_size; + unsigned int dma1_offset; + unsigned int dma1_size; + unsigned int iram_offset; + unsigned int iram_size; + unsigned int dram_offset; + unsigned int dram_size; + unsigned int mbox_offset; + unsigned int mbox_size; + unsigned int acpi_lpe_res_index; + unsigned int acpi_ddr_index; + unsigned int acpi_ipc_irq_index; +}; + struct sst_ipc_info { int ipc_offset; unsigned int mbox_recv_off; @@ -110,7 +131,9 @@ struct sst_ipc_info { struct sst_platform_info { const struct sst_info *probe_data; const struct sst_ipc_info *ipc_info; + const struct sst_res_info *res_info; const struct sst_lib_dnld_info *lib_info; + const char *platform; }; int add_sst_platform_device(void); #endif -- cgit v1.1 From 4a2019480bc5146eb54fc5f0b2ff57b95629a09a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:41:56 +0200 Subject: ASoC: dapm: Only mark paths dirty when the connection status changed Rework soc_dapm_{mixer,mux}_update_power() to only mark a path dirty if the connect state if the path has actually changed. This avoids unnecessary power state checks for the widgets involved. Also factor out the common code that is involved in this into a helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 39 ++++++++++++++++++++++++++------------- 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f03e0cf..116d443 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1925,12 +1925,31 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif +/* + * soc_dapm_connect_path() - Connects or disconnects a path + * @path: The path to update + * @connect: The new connect state of the path. True if the path is connected, + * false if it is disconneted. + * @reason: The reason why the path changed (for debugging only) + */ +static void soc_dapm_connect_path(struct snd_soc_dapm_path *path, + bool connect, const char *reason) +{ + if (path->connect == connect) + return; + + path->connect = connect; + dapm_mark_dirty(path->source, reason); + dapm_mark_dirty(path->sink, reason); +} + /* test and update the power status of a mux widget */ static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; + bool connect; lockdep_assert_held(&card->dapm_mutex); @@ -1941,16 +1960,12 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) { - path->connect = 1; /* new connection */ - dapm_mark_dirty(path->source, "mux connection"); - } else { - if (path->connect) - dapm_mark_dirty(path->source, - "mux disconnection"); - path->connect = 0; /* old connection must be powered down */ - } - dapm_mark_dirty(path->sink, "mux change"); + if (!(strcmp(path->name, e->texts[mux]))) + connect = true; + else + connect = false; + + soc_dapm_connect_path(path, connect, "mux update"); } if (found) @@ -1989,9 +2004,7 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; - path->connect = connect; - dapm_mark_dirty(path->source, "mixer connection"); - dapm_mark_dirty(path->sink, "mixer update"); + soc_dapm_connect_path(path, connect, "mixer update"); } if (found) -- cgit v1.1 From 98407efc1384b31cdcb1eeddc74ee35499d3418f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:41:57 +0200 Subject: ASoC: dapm: Do not add un-muxed paths to MUX control Paths that are directly connected to a MUX widget are not affected by changes to the MUX's control. Rather than checking if a path is directly connected each time the MUX is updated do it only once when MUX is created. We can also remove the check for e->texts[mux] != NULL, since if that condition was true the code would have had already crashed much earlier (And generally speaking if a enum's 'texts' entry is NULL it's a bug in the driver). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 116d443..1fed2207b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -738,8 +738,10 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) if (ret < 0) return ret; - list_for_each_entry(path, &w->sources, list_sink) - dapm_kcontrol_add_path(w->kcontrols[0], path); + list_for_each_entry(path, &w->sources, list_sink) { + if (path->name) + dapm_kcontrol_add_path(w->kcontrols[0], path); + } return 0; } @@ -1955,9 +1957,6 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { - if (!path->name || !e->texts[mux]) - continue; - found = 1; /* we now need to match the string in the enum to the path */ if (!(strcmp(path->name, e->texts[mux]))) -- cgit v1.1 From 5fe5b767dc6fb3df6fa6eaa8e05b727914f2bb4c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:41:58 +0200 Subject: ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets Controls on a path only have an effect if the sink on the path is either a mixer or mux widget. Currently we sort of silently ignore controls on other paths, but since they don't do anything having them on other paths does not make much sense and it is probably safe to assume that if we see such a path it is a mistake in the driver that registered the path. This patch modifies snd_soc_dapm_add_path() to report an error if a path with and control is encountered where we didn't expect a control. This also allows to simplify the code quite a bit. The patch also moves the connecting of the path lists out of dapm_connect_mux() and dapm_connect_mixer() into snd_soc_dapm_add_path(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 112 +++++++++++++++++---------------------------------- 1 file changed, 37 insertions(+), 75 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1fed2207b..c49df10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -469,10 +469,9 @@ out: /* connect mux widget to its interconnecting audio paths */ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, - struct snd_soc_dapm_path *path, const char *control_name, - const struct snd_kcontrol_new *kcontrol) + struct snd_soc_dapm_path *path, const char *control_name) { + const struct snd_kcontrol_new *kcontrol = &path->sink->kcontrol_news[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, item; int i; @@ -493,9 +492,6 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, for (i = 0; i < e->items; i++) { if (!(strcmp(control_name, e->texts[i]))) { - list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &dest->sources); - list_add(&path->list_source, &src->sinks); path->name = e->texts[i]; if (i == item) path->connect = 1; @@ -509,11 +505,10 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, } /* set up initial codec paths */ -static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, - struct snd_soc_dapm_path *p, int i) +static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i) { struct soc_mixer_control *mc = (struct soc_mixer_control *) - w->kcontrol_news[i].private_value; + p->sink->kcontrol_news[i].private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; unsigned int max = mc->max; @@ -522,7 +517,7 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, unsigned int val; if (reg != SND_SOC_NOPM) { - soc_dapm_read(w->dapm, reg, &val); + soc_dapm_read(p->sink->dapm, reg, &val); val = (val >> shift) & mask; if (invert) val = max - val; @@ -534,19 +529,15 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, /* connect mixer widget to its interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, struct snd_soc_dapm_path *path, const char *control_name) { int i; /* search for mixer kcontrol */ - for (i = 0; i < dest->num_kcontrols; i++) { - if (!strcmp(control_name, dest->kcontrol_news[i].name)) { - list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &dest->sources); - list_add(&path->list_source, &src->sinks); - path->name = dest->kcontrol_news[i].name; - dapm_set_mixer_path_status(dest, path, i); + for (i = 0; i < path->sink->num_kcontrols; i++) { + if (!strcmp(control_name, path->sink->kcontrol_news[i].name)) { + path->name = path->sink->kcontrol_news[i].name; + dapm_set_mixer_path_status(path, i); return 0; } } @@ -2272,69 +2263,40 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, wsource->ext = 1; } - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); - /* connect static paths */ if (control == NULL) { - list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &wsink->sources); - list_add(&path->list_source, &wsource->sinks); path->connect = 1; - return 0; - } - - /* connect dynamic paths */ - switch (wsink->id) { - case snd_soc_dapm_adc: - case snd_soc_dapm_dac: - case snd_soc_dapm_pga: - case snd_soc_dapm_out_drv: - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_siggen: - case snd_soc_dapm_micbias: - case snd_soc_dapm_vmid: - case snd_soc_dapm_pre: - case snd_soc_dapm_post: - case snd_soc_dapm_supply: - case snd_soc_dapm_regulator_supply: - case snd_soc_dapm_clock_supply: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai_in: - case snd_soc_dapm_dai_out: - case snd_soc_dapm_dai_link: - case snd_soc_dapm_kcontrol: - list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &wsink->sources); - list_add(&path->list_source, &wsource->sinks); - path->connect = 1; - return 0; - case snd_soc_dapm_mux: - ret = dapm_connect_mux(dapm, wsource, wsink, path, control, - &wsink->kcontrol_news[0]); - if (ret != 0) - goto err; - break; - case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: - case snd_soc_dapm_mixer_named_ctl: - ret = dapm_connect_mixer(dapm, wsource, wsink, path, control); - if (ret != 0) + } else { + /* connect dynamic paths */ + switch (wsink->id) { + case snd_soc_dapm_mux: + ret = dapm_connect_mux(dapm, path, control); + if (ret != 0) + goto err; + break; + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + ret = dapm_connect_mixer(dapm, path, control); + if (ret != 0) + goto err; + break; + default: + dev_err(dapm->dev, + "Control not supported for path %s -> [%s] -> %s\n", + wsource->name, control, wsink->name); + ret = -EINVAL; goto err; - break; - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_line: - case snd_soc_dapm_spk: - list_add(&path->list, &dapm->card->paths); - list_add(&path->list_sink, &wsink->sources); - list_add(&path->list_source, &wsource->sinks); - path->connect = 0; - return 0; + } } + list_add(&path->list, &dapm->card->paths); + list_add(&path->list_sink, &wsink->sources); + list_add(&path->list_source, &wsource->sinks); + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: kfree(path); -- cgit v1.1 From 6dd98b0a3e58b7b48a422802b5610b95ef5128eb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:41:59 +0200 Subject: ASoC: dapm: Introduce toplevel widget categories DAPM widgets can be classified into four categories: * supply: Supply widgets do not affect the power state of their non-supply widget neighbors and unlike other widgets a supply widget is not powered up when it is on an active path, but when at least on of its neighbors is powered up. * source: A source is a widget that receives data from outside the DAPM graph or generates data. This can for example be a microphone, the playback DMA or a signal generator. A source widget will be considered powered up if there is an active path to a sink widget. * sink: A sink is a widget that transmits data to somewhere outside of the DAPM graph. This can e.g. be a speaker or the capture DMA. A sink widget will be considered powered up if there is an active path from a source widget. * normal: Normal widgets are widgets not covered by the categories above. A normal widget will be considered powered up if it is on an active path between a source widget and a sink widget. The way the number of input and output paths for a widget is calculated depends on its category. There are a bunch of factors which decide which category a widget is. Currently there is no formal classification of these categories and we calculate the category of the widget based on these factors whenever we want to know it. This is at least once for every widget during each power update sequence. The factors which determine the category of the widgets are mostly static though and if at all change rather seldom. This patch introduces three new per widget flags, one for each of non-normal widgets categories. Instead of re-computing the category each time we want to know them the flags will be checked. For the majority of widgets the category is solely determined by the widget id, which means it never changes and only has to be set once when the widget is created. The only widgets with dynamic categories are: snd_soc_dapm_dai_out: Is considered a sink iff the capture stream is active, otherwise normal. snd_soc_dapm_dai_in: Is considered a source iff the playback stream is active, otherwise normal. snd_soc_dapm_input: Is considered a sink iff it has no outgoing paths, otherwise normal. snd_soc_dapm_output: Is considered a source iff it has no incoming paths, otherwise normal. snd_soc_dapm_line: Is considered a sink iff it has no outgoing paths and is considered a source iff it has no incoming paths, otherwise normal. For snd_soc_dapm_dai_out/snd_soc_dapm_dai_in widgets the category will be updated when a stream is started or stopped. For the other dynamic widgets the category will be updated when a path connecting to it is added or removed. Introducing those new widget categories allows to make is_connected_{output,input}_ep, which are among the hottest paths of the DAPM algorithm, more generic and significantly shorter. The before and after sizes for is_connected_{output,input}_ep are: On ARM (defconfig + CONFIG_SND_SOC): function old new delta is_connected_output_ep 480 340 -140 is_connected_input_ep 456 352 -104 On amd64 (defconfig + CONFIG_SND_SOC): function old new delta is_connected_output_ep 579 427 -152 is_connected_input_ep 563 427 -136 Which is about a 25%-30% decrease, other architectures are expected to have similar numbers. At the same time the size of the snd_soc_dapm_widget struct does not change since the new flags are stored in the same word as the existing flags. Note: that since the per widget 'ext' flag was only used to decide whether a snd_soc_dapm_input or snd_soc_dapm_output widget was a source or a sink it is now unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 +- sound/soc/soc-dapm.c | 210 +++++++++++++++++++++-------------------------- 2 files changed, 95 insertions(+), 119 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ebb93f2..8cf3aad 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -543,11 +543,13 @@ struct snd_soc_dapm_widget { unsigned char active:1; /* active stream on DAC, ADC's */ unsigned char connected:1; /* connected codec pin */ unsigned char new:1; /* cnew complete */ - unsigned char ext:1; /* has external widgets */ unsigned char force:1; /* force state */ unsigned char ignore_suspend:1; /* kept enabled over suspend */ unsigned char new_power:1; /* power from this run */ unsigned char power_checked:1; /* power checked this run */ + unsigned char is_supply:1; /* Widget is a supply type widget */ + unsigned char is_sink:1; /* Widget is a sink type widget */ + unsigned char is_source:1; /* Widget is a source type widget */ int subseq; /* sort within widget type */ int (*power_check)(struct snd_soc_dapm_widget *w); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c49df10..2cad5f7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,43 +821,12 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, DAPM_UPDATE_STAT(widget, path_checks); - switch (widget->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_regulator_supply: - case snd_soc_dapm_clock_supply: - case snd_soc_dapm_kcontrol: + if (widget->is_supply) return 0; - default: - break; - } - switch (widget->id) { - case snd_soc_dapm_adc: - case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai_out: - if (widget->active) { - widget->outputs = snd_soc_dapm_suspend_check(widget); - return widget->outputs; - } - default: - break; - } - - if (widget->connected) { - /* connected pin ? */ - if (widget->id == snd_soc_dapm_output && !widget->ext) { - widget->outputs = snd_soc_dapm_suspend_check(widget); - return widget->outputs; - } - - /* connected jack or spk ? */ - if (widget->id == snd_soc_dapm_hp || - widget->id == snd_soc_dapm_spk || - (widget->id == snd_soc_dapm_line && - !list_empty(&widget->sources))) { - widget->outputs = snd_soc_dapm_suspend_check(widget); - return widget->outputs; - } + if (widget->is_sink && widget->connected) { + widget->outputs = snd_soc_dapm_suspend_check(widget); + return widget->outputs; } list_for_each_entry(path, &widget->sinks, list_source) { @@ -913,55 +882,12 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, DAPM_UPDATE_STAT(widget, path_checks); - switch (widget->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_regulator_supply: - case snd_soc_dapm_clock_supply: - case snd_soc_dapm_kcontrol: + if (widget->is_supply) return 0; - default: - break; - } - - /* active stream ? */ - switch (widget->id) { - case snd_soc_dapm_dac: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai_in: - if (widget->active) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } - default: - break; - } - - if (widget->connected) { - /* connected pin ? */ - if (widget->id == snd_soc_dapm_input && !widget->ext) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } - /* connected VMID/Bias for lower pops */ - if (widget->id == snd_soc_dapm_vmid) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } - - /* connected jack ? */ - if (widget->id == snd_soc_dapm_mic || - (widget->id == snd_soc_dapm_line && - !list_empty(&widget->sinks))) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } - - /* signal generator */ - if (widget->id == snd_soc_dapm_siggen) { - widget->inputs = snd_soc_dapm_suspend_check(widget); - return widget->inputs; - } + if (widget->is_source && widget->connected) { + widget->inputs = snd_soc_dapm_suspend_check(widget); + return widget->inputs; } list_for_each_entry(path, &widget->sources, list_sink) { @@ -1554,18 +1480,11 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, list_for_each_entry(path, &w->sources, list_sink) dapm_widget_set_peer_power(path->source, power, path->connect); - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_regulator_supply: - case snd_soc_dapm_clock_supply: - case snd_soc_dapm_kcontrol: - /* Supplies can't affect their outputs, only their inputs */ - break; - default: + /* Supplies can't affect their outputs, only their inputs */ + if (!w->is_supply) { list_for_each_entry(path, &w->sinks, list_source) dapm_widget_set_peer_power(path->sink, power, path->connect); - break; } if (power) @@ -2226,6 +2145,53 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); +/* + * dapm_update_widget_flags() - Re-compute widget sink and source flags + * @w: The widget for which to update the flags + * + * Some widgets have a dynamic category which depends on which neighbors they + * are connected to. This function update the category for these widgets. + * + * This function must be called whenever a path is added or removed to a widget. + */ +static void dapm_update_widget_flags(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_path *p; + + switch (w->id) { + case snd_soc_dapm_input: + w->is_source = 1; + list_for_each_entry(p, &w->sources, list_sink) { + if (p->source->id == snd_soc_dapm_micbias || + p->source->id == snd_soc_dapm_mic || + p->source->id == snd_soc_dapm_line || + p->source->id == snd_soc_dapm_output) { + w->is_source = 0; + break; + } + } + break; + case snd_soc_dapm_output: + w->is_sink = 1; + list_for_each_entry(p, &w->sinks, list_source) { + if (p->sink->id == snd_soc_dapm_spk || + p->sink->id == snd_soc_dapm_hp || + p->sink->id == snd_soc_dapm_line || + p->sink->id == snd_soc_dapm_input) { + w->is_sink = 0; + break; + } + } + break; + case snd_soc_dapm_line: + w->is_sink = !list_empty(&w->sources); + w->is_source = !list_empty(&w->sinks); + break; + default: + break; + } +} + static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, const char *control, @@ -2247,22 +2213,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); - /* check for external widgets */ - if (wsink->id == snd_soc_dapm_input) { - if (wsource->id == snd_soc_dapm_micbias || - wsource->id == snd_soc_dapm_mic || - wsource->id == snd_soc_dapm_line || - wsource->id == snd_soc_dapm_output) - wsink->ext = 1; - } - if (wsource->id == snd_soc_dapm_output) { - if (wsink->id == snd_soc_dapm_spk || - wsink->id == snd_soc_dapm_hp || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_input) - wsource->ext = 1; - } - /* connect static paths */ if (control == NULL) { path->connect = 1; @@ -2294,6 +2244,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); + dapm_update_widget_flags(wsource); + dapm_update_widget_flags(wsink); + dapm_mark_dirty(wsource, "Route added"); dapm_mark_dirty(wsink, "Route added"); @@ -2377,6 +2330,7 @@ err: static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route) { + struct snd_soc_dapm_widget *wsource, *wsink; struct snd_soc_dapm_path *path, *p; const char *sink; const char *source; @@ -2414,10 +2368,17 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, } if (path) { - dapm_mark_dirty(path->source, "Route removed"); - dapm_mark_dirty(path->sink, "Route removed"); + wsource = path->source; + wsink = path->sink; + + dapm_mark_dirty(wsource, "Route removed"); + dapm_mark_dirty(wsink, "Route removed"); dapm_free_path(path); + + /* Update any path related flags */ + dapm_update_widget_flags(wsource); + dapm_update_widget_flags(wsink); } else { dev_warn(dapm->dev, "ASoC: Route %s->%s does not exist\n", source, sink); @@ -2975,26 +2936,33 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } switch (w->id) { - case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: - case snd_soc_dapm_mixer_named_ctl: + case snd_soc_dapm_mic: + case snd_soc_dapm_input: + w->is_source = 1; w->power_check = dapm_generic_check_power; break; - case snd_soc_dapm_mux: + case snd_soc_dapm_spk: + case snd_soc_dapm_hp: + case snd_soc_dapm_output: + w->is_sink = 1; w->power_check = dapm_generic_check_power; break; + case snd_soc_dapm_vmid: + case snd_soc_dapm_siggen: + w->is_source = 1; + w->power_check = dapm_always_on_check_power; + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: - case snd_soc_dapm_input: - case snd_soc_dapm_output: case snd_soc_dapm_micbias: - case snd_soc_dapm_spk: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: case snd_soc_dapm_dai_out: @@ -3005,6 +2973,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: case snd_soc_dapm_kcontrol: + w->is_supply = 1; w->power_check = dapm_supply_check_power; break; default: @@ -3368,6 +3337,11 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: break; } + + if (w->id == snd_soc_dapm_dai_in) + w->is_source = w->active; + else + w->is_sink = w->active; } } -- cgit v1.1 From c1862c8bae520a8986dd7c47ce33f16eb7c791c2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:42:00 +0200 Subject: ASoC: dapm: Add a flag to mark paths connected to supply widgets Supply widgets do not count towards the input and output widgets of their neighbors and for supply widgets themselves we do not care for the number of input or output paths. This means that a path that connects to a supply widget effectively behaves the same as a path that as the weak property set. This patch adds a new path flag that gets set to true when the path is connected to at least one supply widget. If a path with the flag set is encountered in is_connected_{input,output}_ep() is is skipped in the same way that weak paths are skipped. This slightly brings down the number of path checks. Since both the weak and the supply flag are implemented as bitfields which are stored in the same word there is no runtime overhead due to checking both rather than just one and also the size of the path struct is not increased by this patch. Another advantage is that we do not have to handle supply widgets in is_connected_{input,output}_ep() anymore since it will never be called for supply widgets. The only exception is from dapm_widget_power_read_file() where a check is added to special case supply widgets. Testing with the ADAU1761, which has a handful of supply widgets, shows the following changes in the DAPM stats for a playback stream start. Power Path Neighbour Before: 34 78 117 After: 34 48 117 Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 23 +++++++++++++---------- 2 files changed, 14 insertions(+), 10 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8cf3aad..e7ebeb7 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,7 @@ struct snd_soc_dapm_path { u32 connect:1; /* source and sink widgets are connected */ u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ + u32 is_supply:1; /* At least one of the connected widgets is a supply */ int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2cad5f7..d89be15 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,9 +821,6 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, DAPM_UPDATE_STAT(widget, path_checks); - if (widget->is_supply) - return 0; - if (widget->is_sink && widget->connected) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -832,7 +829,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, list_for_each_entry(path, &widget->sinks, list_source) { DAPM_UPDATE_STAT(widget, neighbour_checks); - if (path->weak) + if (path->weak || path->is_supply) continue; if (path->walking) @@ -882,9 +879,6 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, DAPM_UPDATE_STAT(widget, path_checks); - if (widget->is_supply) - return 0; - if (widget->is_source && widget->connected) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -893,7 +887,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, list_for_each_entry(path, &widget->sources, list_sink) { DAPM_UPDATE_STAT(widget, neighbour_checks); - if (path->weak) + if (path->weak || path->is_supply) continue; if (path->walking) @@ -1691,8 +1685,14 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + /* Supply widgets are not handled by is_connected_{input,output}_ep() */ + if (w->is_supply) { + in = 0; + out = 0; + } else { + in = is_connected_input_ep(w, NULL); + out = is_connected_output_ep(w, NULL); + } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", @@ -2213,6 +2213,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); + if (wsource->is_supply || wsink->is_supply) + path->is_supply = 1; + /* connect static paths */ if (control == NULL) { path->connect = 1; -- cgit v1.1 From 8be4da29cf5b8ec65e974c36e7ae4d90b381ac5e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:42:01 +0200 Subject: ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume The state of endpoint widgets is affected by that card's power state. Endpoint widgets that do no have the ignore_suspend flag set will be considered inactive during suspend. So they have to be re-checked and marked dirty after the card's power state changes. Currently the input and output widgets are marked dirty instead, this works most of the time since typically a path from one endpoint to another will go via a input or output widget. But marking the endpoints dirty is technically more correct and will also work for odd corner cases. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dapm.c | 15 ++++----------- 3 files changed, 9 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e7ebeb7..43ca165 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -435,7 +435,7 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card); unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol); /* Mostly internal - should not normally be used */ -void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); +void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..443be00 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -629,8 +629,8 @@ int snd_soc_suspend(struct device *dev) SND_SOC_DAPM_STREAM_SUSPEND); } - /* Recheck all analogue paths too */ - dapm_mark_io_dirty(&card->dapm); + /* Recheck all endpoints too, their state is affected by suspend */ + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); /* suspend all CODECs */ @@ -796,8 +796,8 @@ static void soc_resume_deferred(struct work_struct *work) /* userspace can access us now we are back as we were before */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); - /* Recheck all analogue paths too */ - dapm_mark_io_dirty(&card->dapm); + /* Recheck all endpoints too, their state is affected by suspend */ + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d89be15..ffcda7e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -159,27 +159,20 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } } -void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) +void dapm_mark_endpoints_dirty(struct snd_soc_card *card) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_input: - case snd_soc_dapm_output: - dapm_mark_dirty(w, "Rechecking inputs and outputs"); - break; - default: - break; - } + if (w->is_sink || w->is_source) + dapm_mark_dirty(w, "Rechecking endpoints"); } mutex_unlock(&card->dapm_mutex); } -EXPORT_SYMBOL_GPL(dapm_mark_io_dirty); +EXPORT_SYMBOL_GPL(dapm_mark_endpoints_dirty); /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( -- cgit v1.1 From e409dfbfccf9a49409197afc677a21e1c11ba015 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:42:02 +0200 Subject: ASoC: dapm: Add a few supply widget sanity checks Supply widgets are somewhat special and not all kinds of paths to or from supply widgets make sense. This patch adds a few sanity checks that errors out during the path instantiation for those invalid paths. This will prevent drivers to depend on weird behavior resulting from such paths as well as will allow the DAPM algorithms to assume that they never see such paths. This patch adds checks for the following three invalid types of paths: * A path with a non-supply widget as a source connected to a supply widget as a sink. Such a path has no effect on either of the two connected widgets. * Paths with a connected() callback that have a non-supply widget as the source. The DAPM algorithm only uses the conneceted() callback for supply widget power checks. And since it prevents caching of the DAPM state there is no intention to make it more generic as it has negative performance implications. * Paths which connect a supply to a mixer or mux via a control. Controls are only meant to affect the routing of audio data. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ffcda7e..8e26c2b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2194,6 +2194,27 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path; int ret; + if (wsink->is_supply && !wsource->is_supply) { + dev_err(dapm->dev, + "Connecting non-supply widget to supply widget is not supported (%s -> %s)\n", + wsource->name, wsink->name); + return -EINVAL; + } + + if (connected && !wsource->is_supply) { + dev_err(dapm->dev, + "connected() callback only supported for supply widgets (%s -> %s)\n", + wsource->name, wsink->name); + return -EINVAL; + } + + if (wsource->is_supply && control) { + dev_err(dapm->dev, + "Conditional paths are not supported for supply widgets (%s -> [%s] -> %s)\n", + wsource->name, control, wsink->name); + return -EINVAL; + } + path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); if (!path) return -ENOMEM; -- cgit v1.1 From 92a99ea439c4e27fc6e32eb6d51c5d091c6084bd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 17:42:03 +0200 Subject: ASoC: dapm: Use more aggressive caching Currently we cache the number of input and output paths going to/from a widget only within a power update sequence. But not in between power update sequences. But we know how changes to the DAPM graph affect the number of input (form a source) and output (to a sink) paths of a widget and only need to recalculate them if a operation has been performed that might have changed them. * Adding/removing or connecting/disconnecting a path means that the for the source of the path the number of output paths can change and for the sink the number of input paths can change. * Connecting/disconnecting a widget has the same effect has connecting/ disconnecting all paths of the widget. So for the widget itself the number of inputs and outputs can change, for all sinks of the widget the number of inputs can change and for all sources of the widget the number of outputs can change. * Activating/Deactivating a stream can either change the number of outputs on the sources of the widget associated with the stream or the number of inputs on the sinks. Instead of always invalidating all cached numbers of input and output paths for each power up or down sequence this patch restructures the code to only invalidate the cached numbers when a operation that might change them has been performed. This can greatly reduce the number of DAPM power checks for some very common operations. Since per DAPM operation typically only either change the number of inputs or outputs the number of path checks is reduced by at least 50%. The number of neighbor checks is also reduced about the same percentage, but since the number of neighbors encountered when walking from sink to source is not the same as when walking from source to sink the actual numbers will slightly vary from card to card (e.g. for a mixer we see 1 neighbor when walking from source to sink, but the number of inputs neighbors when walking from source to sink). Bigger improvements can be observed for widgets with multiple connected inputs and output (e.g. mixers probably being the most widespread form of this). Previously we had to re-calculate the number of inputs and outputs on all input and output paths. With this change we only have to re-calculate the number of outputs on the input path that got changed and the number of inputs on the output paths. E.g. imagine the following example: A --> B ----. v M --> N --> Z <-- S <-- R | v X Widget Z has multiple input paths, if any change was made that cause Z to be marked as dirty the power state of Z has to be re-computed. This requires to know the number of inputs and outputs of Z, which requires to know the number of inputs and outputs of all widgets on all paths from or to Z. Previously this meant re-computing all inputs and outputs of all the path going into or out of Z. With this patch in place only paths that actually have changed need to be re-computed. If the system is idle (or the part of the system affected by the changed path) the number of path checks drops to either 0 or 1, regardless of how large or complex the DAPM context is. 0 if there is no connected sink and no connected source. 1 if there is either a connected source or sink, but not both. The number of neighbor checks again will scale accordingly and will be a constant number that is the number of inputs or outputs of the widget for which we did the path check. When loading a state file or switching between different profiles typically multiple mixer and mux settings are changed, so we see the benefit of this patch multiplied for these kinds of operations. Testing with the ADAU1761 shows the following changes in DAPM stats for changing a single Mixer switch for a Mixer with 5 inputs while the DAPM context is idle. Power Path Neighbour Before: 2 12 30 After: 2 1 2 For the same switch, but with a active playback stream the stat changed are as follows. Power Path Neighbour Before: 10 20 54 After: 10 7 21 Cumulative numbers for switching the audio profile which changes 7 controls while the system is idle: Power Path Neighbour Before: 16 80 170 After: 16 7 23 Cumulative numbers for switching the audio profile which changes 7 controls while playback is active: Power Path Neighbour Before: 51 123 273 After: 51 29 109 Starting (or stopping) the playback stream: Power Path Neighbour Before: 34 34 117 After: 34 17 69 Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 161 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 154 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 43ca165..89823cf 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -569,6 +569,7 @@ struct snd_soc_dapm_widget { struct list_head sinks; /* used during DAPM updates */ + struct list_head work_list; struct list_head power_list; struct list_head dirty; int inputs; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8e26c2b..6bf2c97 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -159,6 +159,116 @@ static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } } +/* + * dapm_widget_invalidate_input_paths() - Invalidate the cached number of input + * paths + * @w: The widget for which to invalidate the cached number of input paths + * + * The function resets the cached number of inputs for the specified widget and + * all widgets that can be reached via outgoing paths from the widget. + * + * This function must be called if the number of input paths for a widget might + * have changed. E.g. if the source state of a widget changes or a path is added + * or activated with the widget as the sink. + */ +static void dapm_widget_invalidate_input_paths(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *sink; + struct snd_soc_dapm_path *p; + LIST_HEAD(list); + + dapm_assert_locked(w->dapm); + + if (w->inputs == -1) + return; + + w->inputs = -1; + list_add_tail(&w->work_list, &list); + + list_for_each_entry(w, &list, work_list) { + list_for_each_entry(p, &w->sinks, list_source) { + if (p->is_supply || p->weak || !p->connect) + continue; + sink = p->sink; + if (sink->inputs != -1) { + sink->inputs = -1; + list_add_tail(&sink->work_list, &list); + } + } + } +} + +/* + * dapm_widget_invalidate_output_paths() - Invalidate the cached number of + * output paths + * @w: The widget for which to invalidate the cached number of output paths + * + * Resets the cached number of outputs for the specified widget and all widgets + * that can be reached via incoming paths from the widget. + * + * This function must be called if the number of output paths for a widget might + * have changed. E.g. if the sink state of a widget changes or a path is added + * or activated with the widget as the source. + */ +static void dapm_widget_invalidate_output_paths(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget *source; + struct snd_soc_dapm_path *p; + LIST_HEAD(list); + + dapm_assert_locked(w->dapm); + + if (w->outputs == -1) + return; + + w->outputs = -1; + list_add_tail(&w->work_list, &list); + + list_for_each_entry(w, &list, work_list) { + list_for_each_entry(p, &w->sources, list_sink) { + if (p->is_supply || p->weak || !p->connect) + continue; + source = p->source; + if (source->outputs != -1) { + source->outputs = -1; + list_add_tail(&source->work_list, &list); + } + } + } +} + +/* + * dapm_path_invalidate() - Invalidates the cached number of inputs and outputs + * for the widgets connected to a path + * @p: The path to invalidate + * + * Resets the cached number of inputs for the sink of the path and the cached + * number of outputs for the source of the path. + * + * This function must be called when a path is added, removed or the connected + * state changes. + */ +static void dapm_path_invalidate(struct snd_soc_dapm_path *p) +{ + /* + * Weak paths or supply paths do not influence the number of input or + * output paths of their neighbors. + */ + if (p->weak || p->is_supply) + return; + + /* + * The number of connected endpoints is the sum of the number of + * connected endpoints of all neighbors. If a node with 0 connected + * endpoints is either connected or disconnected that sum won't change, + * so there is no need to re-check the path. + */ + if (p->source->inputs != 0) + dapm_widget_invalidate_input_paths(p->sink); + if (p->sink->outputs != 0) + dapm_widget_invalidate_output_paths(p->source); +} + void dapm_mark_endpoints_dirty(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; @@ -166,8 +276,13 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { - if (w->is_sink || w->is_source) + if (w->is_sink || w->is_source) { dapm_mark_dirty(w, "Rechecking endpoints"); + if (w->is_sink) + dapm_widget_invalidate_output_paths(w); + if (w->is_source) + dapm_widget_invalidate_input_paths(w); + } } mutex_unlock(&card->dapm_mutex); @@ -379,8 +494,6 @@ static void dapm_reset(struct snd_soc_card *card) list_for_each_entry(w, &card->widgets, list) { w->new_power = w->power; w->power_checked = false; - w->inputs = -1; - w->outputs = -1; } } @@ -931,10 +1044,19 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list) { struct snd_soc_card *card = dai->card; + struct snd_soc_dapm_widget *w; int paths; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_reset(card); + + /* + * For is_connected_{output,input}_ep fully discover the graph we need + * to reset the cached number of inputs and outputs. + */ + list_for_each_entry(w, &card->widgets, list) { + w->inputs = -1; + w->outputs = -1; + } if (stream == SNDRV_PCM_STREAM_PLAYBACK) paths = is_connected_output_ep(dai->playback_widget, list); @@ -1846,6 +1968,7 @@ static void soc_dapm_connect_path(struct snd_soc_dapm_path *path, path->connect = connect; dapm_mark_dirty(path->source, reason); dapm_mark_dirty(path->sink, reason); + dapm_path_invalidate(path); } /* test and update the power status of a mux widget */ @@ -2084,8 +2207,11 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return -EINVAL; } - if (w->connected != status) + if (w->connected != status) { dapm_mark_dirty(w, "pin configuration"); + dapm_widget_invalidate_input_paths(w); + dapm_widget_invalidate_output_paths(w); + } w->connected = status; if (status == 0) @@ -2267,6 +2393,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(wsource, "Route added"); dapm_mark_dirty(wsink, "Route added"); + if (dapm->card->instantiated && path->connect) + dapm_path_invalidate(path); + return 0; err: kfree(path); @@ -2390,6 +2519,8 @@ static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(wsource, "Route removed"); dapm_mark_dirty(wsink, "Route removed"); + if (path->connect) + dapm_path_invalidate(path); dapm_free_path(path); @@ -3007,6 +3138,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->dirty); list_add(&w->list, &dapm->card->widgets); + w->inputs = -1; + w->outputs = -1; + /* machine layer set ups unconnected pins and insertions */ w->connected = 1; return w; @@ -3355,10 +3489,13 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, break; } - if (w->id == snd_soc_dapm_dai_in) + if (w->id == snd_soc_dapm_dai_in) { w->is_source = w->active; - else + dapm_widget_invalidate_input_paths(w); + } else { w->is_sink = w->active; + dapm_widget_invalidate_output_paths(w); + } } } @@ -3485,7 +3622,15 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, } dev_dbg(w->dapm->dev, "ASoC: force enable pin %s\n", pin); - w->connected = 1; + if (!w->connected) { + /* + * w->force does not affect the number of input or output paths, + * so we only have to recheck if w->connected is changed + */ + dapm_widget_invalidate_input_paths(w); + dapm_widget_invalidate_output_paths(w); + w->connected = 1; + } w->force = 1; dapm_mark_dirty(w, "force enable"); -- cgit v1.1 From c1b4d1c7774189002bc08766ec10e339dfbc98d6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 25 Oct 2014 20:25:56 +0200 Subject: ASoC: Use generic control handlers for S8 control Commit f227b88f0fce ("ASoC: core: Add signed register volume control logic") added support for signed control to the generic volsw control handler. This makes it possible to use them for the S8 control as well, rather than having to use a custom control handler implementation. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 19 +++++------- sound/soc/soc-core.c | 87 ---------------------------------------------------- 2 files changed, 8 insertions(+), 98 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..ad47e96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -36,6 +36,11 @@ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ .invert = xinvert, .autodisable = xautodisable}) +#define SOC_DOUBLE_S_VALUE(xreg, shift_left, shift_right, xmin, xmax, xsign_bit, xinvert, xautodisable) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = shift_left, \ + .rshift = shift_right, .min = xmin, .max = xmax, .platform_max = xmax, \ + .sign_bit = xsign_bit, .invert = xinvert, .autodisable = xautodisable}) #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ @@ -171,11 +176,9 @@ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ - .put = snd_soc_put_volsw_s8, \ - .private_value = (unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .min = xmin, .max = xmax, \ - .platform_max = xmax} } + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ + .put = snd_soc_put_volsw, \ + .private_value = SOC_DOUBLE_S_VALUE(xreg, 0, 8, xmin, xmax, 7, 0, 0) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ @@ -545,12 +548,6 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96ecdc3..47c378a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2721,93 +2721,6 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); /** - * snd_soc_info_volsw_s8 - signed mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); - -/** - * snd_soc_get_volsw_s8 - signed mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int val; - int min = mc->min; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); - -/** - * snd_soc_put_volsw_sgn - signed mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a signed mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - int min = mc->min; - unsigned int val; - - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - - return snd_soc_component_update_bits(component, reg, 0xffff, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); - -/** * snd_soc_info_volsw_range - single mixer info callback with range. * @kcontrol: mixer control * @uinfo: control element information -- cgit v1.1 From 728384a10b673dbd9819b524852df23142ee2fdd Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Fri, 24 Oct 2014 21:50:04 +0400 Subject: ARM: pxa: spitz: register spitz-audio device Register spitz-audio device to be used by ASoC driver to bind ASoC platform driver. Currently old 'soc-audio' approach is used, which needs to be replaced with proper device. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- arch/arm/mach-pxa/spitz.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 840c3a4..962a7f3 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -924,6 +924,14 @@ static inline void spitz_i2c_init(void) {} #endif /****************************************************************************** + * Audio devices + ******************************************************************************/ +static inline void spitz_audio_init(void) +{ + platform_device_register_simple("spitz-audio", -1, NULL, 0); +} + +/****************************************************************************** * Machine init ******************************************************************************/ static void spitz_poweroff(void) @@ -970,6 +978,7 @@ static void __init spitz_init(void) spitz_nor_init(); spitz_nand_init(); spitz_i2c_init(); + spitz_audio_init(); } static void __init spitz_fixup(struct tag *tags, char **cmdline) -- cgit v1.1 From ecf0015161b2220879fcd41c030761b4eb561b95 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Fri, 24 Oct 2014 21:50:05 +0400 Subject: ASoC: pxa: Convert spitz to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 52 +++++++++++++++++++++++++++------------------------ 1 file changed, 28 insertions(+), 24 deletions(-) diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1373b01..d7d5fb2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -305,19 +305,15 @@ static struct snd_soc_card snd_soc_spitz = { .num_dapm_routes = ARRAY_SIZE(spitz_audio_map), }; -static struct platform_device *spitz_snd_device; - -static int __init spitz_init(void) +static int spitz_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_spitz; int ret; - if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) - return -ENODEV; - - if (machine_is_borzoi() || machine_is_spitz()) - spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; - else + if (machine_is_akita()) spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; + else + spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); if (ret) @@ -327,37 +323,45 @@ static int __init spitz_init(void) if (ret) goto err2; - spitz_snd_device = platform_device_alloc("soc-audio", -1); - if (!spitz_snd_device) { - ret = -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); goto err2; } - platform_set_drvdata(spitz_snd_device, &snd_soc_spitz); - - ret = platform_device_add(spitz_snd_device); - if (ret) - goto err3; - return 0; -err3: - platform_device_put(spitz_snd_device); err2: gpio_free(spitz_mic_gpio); err1: return ret; } -static void __exit spitz_exit(void) +static int spitz_remove(struct platform_device *pdev) { - platform_device_unregister(spitz_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); gpio_free(spitz_mic_gpio); + return 0; } -module_init(spitz_init); -module_exit(spitz_exit); +static struct platform_driver spitz_driver = { + .driver = { + .name = "spitz-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = spitz_probe, + .remove = spitz_remove, +}; + +module_platform_driver(spitz_driver); MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Spitz"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spitz-audio"); -- cgit v1.1 From 3f7256fe5fc64132a2dd19695255c990aa2188cf Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 24 Oct 2014 13:01:25 -0200 Subject: ASoC: sgtl5000: Use the preferred form for passing a size of a struct According to Documentation/CodingStyle - Chapter 14: "The preferred form for passing a size of a struct is the following: p = kmalloc(sizeof(*p), ...); The alternative form where struct name is spelled out hurts readability and introduces an opportunity for a bug when the pointer variable type is changed but the corresponding sizeof that is passed to a memory allocator is not." So do it as recommeded. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 59336f6..490404c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1439,8 +1439,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, struct device_node *np = client->dev.of_node; u32 value; - sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), - GFP_KERNEL); + sgtl5000 = devm_kzalloc(&client->dev, sizeof(*sgtl5000), GFP_KERNEL); if (!sgtl5000) return -ENOMEM; -- cgit v1.1 From 54ec2d5f3f751ddcbf07b0fc1e5f01e43015e8e0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 24 Oct 2014 13:01:26 -0200 Subject: ASoC: wm8962: Use the preferred form for passing a size of a struct According to Documentation/CodingStyle - Chapter 14: "The preferred form for passing a size of a struct is the following: p = kmalloc(sizeof(*p), ...); The alternative form where struct name is spelled out hurts readability and introduces an opportunity for a bug when the pointer variable type is changed but the corresponding sizeof that is passed to a memory allocator is not." So do it as recommeded. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 9077411..cfd3891 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3552,8 +3552,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i, irq_pol, trigger; - wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), - GFP_KERNEL); + wm8962 = devm_kzalloc(&i2c->dev, sizeof(*wm8962), GFP_KERNEL); if (wm8962 == NULL) return -ENOMEM; -- cgit v1.1 From cea82d8af3986508a05949cfbb6ad8e99ffc15eb Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 24 Oct 2014 13:01:27 -0200 Subject: ASoC: wm8731: Use the preferred form for passing a size of a struct According to Documentation/CodingStyle - Chapter 14: "The preferred form for passing a size of a struct is the following: p = kmalloc(sizeof(*p), ...); The alternative form where struct name is spelled out hurts readability and introduces an opportunity for a bug when the pointer variable type is changed but the corresponding sizeof that is passed to a memory allocator is not." So do it as recommeded. Cc: Charles Keepax Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index eebb328..2c9f2a7 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -680,8 +680,7 @@ static int wm8731_spi_probe(struct spi_device *spi) struct wm8731_priv *wm8731; int ret; - wm8731 = devm_kzalloc(&spi->dev, sizeof(struct wm8731_priv), - GFP_KERNEL); + wm8731 = devm_kzalloc(&spi->dev, sizeof(*wm8731), GFP_KERNEL); if (wm8731 == NULL) return -ENOMEM; -- cgit v1.1 From 326f0480b7f8504c4f594c4f36ab7874e17780bc Mon Sep 17 00:00:00 2001 From: Aya Mahfouz Date: Tue, 28 Oct 2014 14:27:44 +0200 Subject: ALSA: pcxhr: convert timeval to ktime_t This patch is concerned with migrating the time variables in the pcxhr module found in the sound driver. The changes are concerend with the y2038 problem where timeval will overflow in the year 2038. ktime_t was used instead of timeval to get the wall time. The difference is displayed now in nanoseconds instead of microseconds. Signed-off-by: Aya Mahfouz Reviewed-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 10 ++++++---- sound/pci/pcxhr/pcxhr_core.c | 10 ++++++---- 2 files changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b854fc5..7c33c97 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -711,8 +711,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) int playback_mask = 0; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif mutex_lock(&mgr->setup_mutex); @@ -823,9 +824,10 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***TRIGGER START*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif } diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index a584acb..181f772 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -910,8 +910,9 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int audio_mask; #ifdef CONFIG_SND_DEBUG_VERBOSE - struct timeval my_tv1, my_tv2; - do_gettimeofday(&my_tv1); + ktime_t start_time, stop_time, diff_time; + + start_time = ktime_get(); #endif audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); @@ -960,9 +961,10 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, return err; } #ifdef CONFIG_SND_DEBUG_VERBOSE - do_gettimeofday(&my_tv2); + stop_time = ktime_get(); + diff_time = ktime_sub(stop_time, start_time); dev_dbg(&mgr->pci->dev, "***SET PIPE STATE*** TIME = %ld (err = %x)\n", - (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); + (long)(ktime_to_ns(diff_time)), err); #endif return 0; } -- cgit v1.1 From 90446d0746c399e47246232f66ff2fa8616ba470 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Oct 2014 15:28:07 +0100 Subject: ALSA: doc: Add missing headers and compress stuff to alsa-driver-api.tmpl Some header files have kereldoc comments but are not referred properly. Let's add them. Signed-off-by: Takashi Iwai --- Documentation/DocBook/alsa-driver-api.tmpl | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 0230a96..13f8b24 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -57,6 +57,7 @@ !Esound/core/pcm.c !Esound/core/pcm_lib.c !Esound/core/pcm_native.c +!include/sound/pcm.h PCM Format Helpers !Esound/core/pcm_misc.c @@ -64,6 +65,10 @@ PCM Memory Management !Esound/core/pcm_memory.c + PCM DMA Engine API +!Esound/core/pcm_dmaengine.c +!Iinclude/sound/dmaengine_pcm.h + Control/Mixer API General Control Interface @@ -91,12 +96,19 @@ !Esound/core/info.c + Compress Offload + Compress Offload API +!Esound/core/compress_offload.c +!Iinclude/sound/compress_driver.h + + Miscellaneous Functions Hardware-Dependent Devices API !Esound/core/hwdep.c Jack Abstraction Layer API !Esound/core/jack.c +!Iinclude/sound/jack.h ISA DMA Helpers !Esound/core/isadma.c -- cgit v1.1 From 7b366d5f161c2a69eeafe525105a5a9277982472 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Oct 2014 15:30:00 +0100 Subject: ALSA: jack: Fix kerneldoc comments Signed-off-by: Takashi Iwai --- include/sound/jack.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 5891657..37e612e 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -28,8 +28,9 @@ struct input_dev; /** - * Jack types which can be reported. These values are used as a - * bitmask. + * enum snd_jack_types: Jack types which can be reported + * + * These values are used as a bitmask. * * Note that this must be kept in sync with the lookup table in * sound/core/jack.c. -- cgit v1.1 From 41282920aa35033a4fcf2bc68aeba42a037e6c4d Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 24 Oct 2014 16:48:11 -0700 Subject: ASoC: fsl-asoc-card: Don't bypass settings if cpu-dai is Master When cpu-dai is the DAI Master (CBM_CFx), it may need some configurations, set_sysclk() call for eample, for cpu-dai side in the hw_params(), even if the set_bias_level() has already taken care of the codec-dai side. So this patch just simply adds an additional condition. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 007c772..14572e6 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -125,7 +125,12 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); - if (priv->card.set_bias_level) + /* + * If codec-dai is DAI Master and all configurations are already in the + * set_bias_level(), bypass the remaining settings in hw_params(). + * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. + */ + if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) return 0; /* Specific configurations of DAIs starts from here */ -- cgit v1.1 From 1b5721b24306c2daf786f3b31f678b40ab9b2ce7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 27 Oct 2014 18:04:52 -0700 Subject: ASoC: simple-card: add asoc_simple_card_parse_daifmt() Current daifmt setting method in simple-card driver is placed to many places, and using un-readable/confusable method. This patch adds new asoc_simple_card_parse_daifmt() and tidyup code. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 134 +++++++++++++++++++--------------------- 1 file changed, 65 insertions(+), 69 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4f192ee..cac95d7 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -226,6 +226,53 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return 0; } +static int asoc_simple_card_parse_daifmt(struct device_node *node, + struct simple_card_data *priv, + struct device_node *cpu, + struct device_node *codec, + char *prefix, int idx) +{ + struct device *dev = simple_priv_to_dev(priv); + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; + struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); + struct asoc_simple_dai *cpu_dai = &dai_props->cpu_dai; + struct asoc_simple_dai *codec_dai = &dai_props->codec_dai; + unsigned int daifmt; + + daifmt = snd_soc_of_parse_daifmt(node, prefix, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + + if (strlen(prefix) && !bitclkmaster && !framemaster) { + /* + * No dai-link level and master setting was not found from + * sound node level, revert back to legacy DT parsing and + * take the settings from codec node. + */ + dev_dbg(dev, "Revert to legacy daifmt parsing\n"); + + cpu_dai->fmt = codec_dai->fmt = + snd_soc_of_parse_daifmt(codec, NULL, NULL, NULL) | + (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); + } else { + if (codec == bitclkmaster) + daifmt |= (codec == framemaster) ? + SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; + else + daifmt |= (codec == framemaster) ? + SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; + + cpu_dai->fmt = daifmt; + codec_dai->fmt = daifmt; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + + return 0; +} + static int asoc_simple_card_dai_link_of(struct device_node *node, struct simple_card_data *priv, int idx, @@ -234,10 +281,8 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); - struct device_node *np = NULL; - struct device_node *bitclkmaster = NULL; - struct device_node *framemaster = NULL; - unsigned int daifmt; + struct device_node *cpu = NULL; + struct device_node *codec = NULL; char *name; char prop[128]; char *prefix = ""; @@ -247,85 +292,36 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (is_top_level_node) prefix = "simple-audio-card,"; - daifmt = snd_soc_of_parse_daifmt(node, prefix, - &bitclkmaster, &framemaster); - daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - snprintf(prop, sizeof(prop), "%scpu", prefix); - np = of_get_child_by_name(node, prop); - if (!np) { + cpu = of_get_child_by_name(node, prop); + + snprintf(prop, sizeof(prop), "%scodec", prefix); + codec = of_get_child_by_name(node, prop); + + if (!cpu || !codec) { ret = -EINVAL; dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); goto dai_link_of_err; } - ret = asoc_simple_card_sub_parse_of(np, &dai_props->cpu_dai, + ret = asoc_simple_card_parse_daifmt(node, priv, + cpu, codec, prefix, idx); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_card_sub_parse_of(cpu, &dai_props->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name, &cpu_args); if (ret < 0) goto dai_link_of_err; - dai_props->cpu_dai.fmt = daifmt; - switch (((np == bitclkmaster) << 4) | (np == framemaster)) { - case 0x11: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - case 0x10: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - case 0x01: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - default: - dai_props->cpu_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - } - - of_node_put(np); - snprintf(prop, sizeof(prop), "%scodec", prefix); - np = of_get_child_by_name(node, prop); - if (!np) { - ret = -EINVAL; - dev_err(dev, "%s: Can't find %s DT node\n", __func__, prop); - goto dai_link_of_err; - } - - ret = asoc_simple_card_sub_parse_of(np, &dai_props->codec_dai, + ret = asoc_simple_card_sub_parse_of(codec, &dai_props->codec_dai, &dai_link->codec_of_node, &dai_link->codec_dai_name, NULL); if (ret < 0) goto dai_link_of_err; - if (strlen(prefix) && !bitclkmaster && !framemaster) { - /* - * No DAI link level and master setting was found - * from sound node level, revert back to legacy DT - * parsing and take the settings from codec node. - */ - dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n", - __func__); - dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = - snd_soc_of_parse_daifmt(np, NULL, NULL, NULL) | - (daifmt & ~SND_SOC_DAIFMT_CLOCK_MASK); - } else { - dai_props->codec_dai.fmt = daifmt; - switch (((np == bitclkmaster) << 4) | (np == framemaster)) { - case 0x11: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFM; - break; - case 0x10: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBM_CFS; - break; - case 0x01: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFM; - break; - default: - dai_props->codec_dai.fmt |= SND_SOC_DAIFMT_CBS_CFS; - break; - } - } - if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) { ret = -EINVAL; goto dai_link_of_err; @@ -368,9 +364,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_link->cpu_dai_name = NULL; dai_link_of_err: - of_node_put(np); - of_node_put(bitclkmaster); - of_node_put(framemaster); + of_node_put(cpu); + of_node_put(codec); + return ret; } -- cgit v1.1 From 859b2e374a86482004d1b8b94c1666269e1d7fd6 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 28 Oct 2014 21:25:13 +0530 Subject: ALSA: compress: fix documentation errors Some structure documentation was not right so fix it now Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index ae6c3b8..396e8f7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -42,12 +42,11 @@ struct snd_compr_ops; * @buffer_size: size of the above buffer * @fragment_size: size of buffer fragment in bytes * @fragments: number of such fragments - * @hw_pointer: offset of last location in buffer where DSP copied data - * @app_pointer: offset of last location in buffer where app wrote data * @total_bytes_available: cumulative number of bytes made available in * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @private_data: driver private data pointer */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -94,6 +93,8 @@ struct snd_compr_stream { * This can be called in during stream creation only to set codec params * and the stream properties * @get_params: retrieve the codec parameters, mandatory + * @set_metadata: Set the metadata values for a stream + * @get_metadata: retreives the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory -- cgit v1.1 From 2e6705c09065ecb357140e44d12dc32274b1a723 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Oct 2014 17:16:51 +0100 Subject: ALSA: ctxfi: Kill the rest snd_print*() Use the standard dev_*() instead. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctatc.c | 6 ++++-- sound/pci/ctxfi/cttimer.c | 4 ++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 632e843..977a598 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -438,7 +438,9 @@ atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm) position = src->ops->get_ca(src); if (position < apcm->vm_block->addr) { - snd_printdd("ctxfi: bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", position, apcm->vm_block->addr, apcm->vm_block->size); + dev_dbg(atc->card->dev, + "bad ca - ca=0x%08x, vba=0x%08x, vbs=0x%08x\n", + position, apcm->vm_block->addr, apcm->vm_block->size); position = apcm->vm_block->addr; } @@ -1295,7 +1297,7 @@ static int atc_identify_card(struct ct_atc *atc, unsigned int ssid) atc->model = CT20K2_UNKNOWN; } atc->model_name = ct_subsys_name[atc->model]; - snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n", + dev_info(atc->card->dev, "chip %s model %s (%04x:%04x) is found\n", atc->chip_name, atc->model_name, vendor_id, device_id); return 0; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 03fb909..a5d4604 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -421,12 +421,12 @@ struct ct_timer *ct_timer_new(struct ct_atc *atc) atimer->atc = atc; hw = atc->hw; if (!use_system_timer && hw->set_timer_irq) { - snd_printd(KERN_INFO "ctxfi: Use xfi-native timer\n"); + dev_info(atc->card->dev, "Use xfi-native timer\n"); atimer->ops = &ct_xfitimer_ops; hw->irq_callback_data = atimer; hw->irq_callback = ct_timer_interrupt; } else { - snd_printd(KERN_INFO "ctxfi: Use system timer\n"); + dev_info(atc->card->dev, "Use system timer\n"); atimer->ops = &ct_systimer_ops; } return atimer; -- cgit v1.1 From f48a6df28239f5bf35d80e43e580261d2298395a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Oct 2014 17:22:45 +0100 Subject: ALSA: pcxhr: Kill the rest snd_print*() Use the standard dev_*() instead. Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 42 +++++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 7c33c97..a602930 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -501,10 +501,10 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, /* * start or stop playback/capture substream */ -static int pcxhr_set_stream_state(struct pcxhr_stream *stream) +static int pcxhr_set_stream_state(struct snd_pcxhr *chip, + struct pcxhr_stream *stream) { int err; - struct snd_pcxhr *chip; struct pcxhr_rmh rmh; int stream_mask, start; @@ -512,8 +512,8 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) start = 1; else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { - snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state " - "CANNOT be stopped\n"); + dev_err(chip->card->dev, + "pcxhr_set_stream_state CANNOT be stopped\n"); return -EINVAL; } start = 0; @@ -560,6 +560,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) struct pcxhr_rmh rmh; unsigned int header; + chip = snd_pcm_substream_chip(stream->substream); switch (stream->format) { case SNDRV_PCM_FORMAT_U8: header = HEADER_FMT_BASE_LIN; @@ -582,11 +583,10 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL; break; default: - snd_printk(KERN_ERR - "error pcxhr_set_format() : unknown format\n"); + dev_err(chip->card->dev, + "error pcxhr_set_format() : unknown format\n"); return -EINVAL; } - chip = snd_pcm_substream_chip(stream->substream); sample_rate = chip->mgr->sample_rate; if (sample_rate <= 32000 && sample_rate !=0) { @@ -643,11 +643,11 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE); stream_num = is_capture ? 0 : subs->number; - snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : " - "addr(%p) bytes(%zx) subs(%d)\n", - is_capture ? 'c' : 'p', - chip->chip_idx, (void *)(long)subs->runtime->dma_addr, - subs->runtime->dma_bytes, subs->number); + dev_dbg(chip->card->dev, + "pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + is_capture ? 'c' : 'p', + chip->chip_idx, (void *)(long)subs->runtime->dma_addr, + subs->runtime->dma_bytes, subs->number); pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS); pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, @@ -687,7 +687,7 @@ static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, *sample_count = ((snd_pcm_uframes_t)rmh.stat[0]) << 24; *sample_count += (snd_pcm_uframes_t)rmh.stat[1]; } - snd_printdd("PIPE_SAMPLE_COUNT = %lx\n", *sample_count); + dev_dbg(chip->card->dev, "PIPE_SAMPLE_COUNT = %lx\n", *sample_count); return err; } #endif @@ -779,12 +779,12 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) for (j = 0; j < chip->nb_streams_capt; j++) { stream = &chip->capture_stream[j]; if (pcxhr_stream_scheduled_get_pipe(stream, &pipe)) - err = pcxhr_set_stream_state(stream); + err = pcxhr_set_stream_state(chip, stream); } for (j = 0; j < chip->nb_streams_play; j++) { stream = &chip->playback_stream[j]; if (pcxhr_stream_scheduled_get_pipe(stream, &pipe)) - err = pcxhr_set_stream_state(stream); + err = pcxhr_set_stream_state(chip, stream); } } @@ -839,12 +839,12 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd) { struct pcxhr_stream *stream; struct snd_pcm_substream *s; + struct snd_pcxhr *chip = snd_pcm_substream_chip(subs); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_printdd("SNDRV_PCM_TRIGGER_START\n"); + dev_dbg(chip->card->dev, "SNDRV_PCM_TRIGGER_START\n"); if (snd_pcm_stream_linked(subs)) { - struct snd_pcxhr *chip = snd_pcm_substream_chip(subs); snd_pcm_group_for_each_entry(s, subs) { if (snd_pcm_substream_chip(s) != chip) continue; @@ -856,7 +856,7 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd) pcxhr_start_linked_stream(chip->mgr); } else { stream = subs->runtime->private_data; - snd_printdd("Only one Substream %c %d\n", + dev_dbg(chip->card->dev, "Only one Substream %c %d\n", stream->pipe->is_capture ? 'C' : 'P', stream->pipe->first_audio); if (pcxhr_set_format(stream)) @@ -865,17 +865,17 @@ static int pcxhr_trigger(struct snd_pcm_substream *subs, int cmd) return -EINVAL; stream->status = PCXHR_STREAM_STATUS_SCHEDULE_RUN; - if (pcxhr_set_stream_state(stream)) + if (pcxhr_set_stream_state(chip, stream)) return -EINVAL; stream->status = PCXHR_STREAM_STATUS_RUNNING; } break; case SNDRV_PCM_TRIGGER_STOP: - snd_printdd("SNDRV_PCM_TRIGGER_STOP\n"); + dev_dbg(chip->card->dev, "SNDRV_PCM_TRIGGER_STOP\n"); snd_pcm_group_for_each_entry(s, subs) { stream = s->runtime->private_data; stream->status = PCXHR_STREAM_STATUS_SCHEDULE_STOP; - if (pcxhr_set_stream_state(stream)) + if (pcxhr_set_stream_state(chip, stream)) return -EINVAL; snd_pcm_trigger_done(s, subs); } -- cgit v1.1 From f994cb3a09a5f2018c286f854c10277132f4a9a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Oct 2014 17:36:50 +0100 Subject: ALSA: au88x0: Kill the rest snd_print*() Use the standard dev_*() instead. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index e9c3833..9963691 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -314,7 +314,7 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) if (snd_seq_device_new(card, 1, SNDRV_SEQ_DEV_ID_VORTEX_SYNTH, sizeof(snd_vortex_synth_arg_t), &wave) < 0 || wave == NULL) { - snd_printk(KERN_ERR "Can't initialize Aureal wavetable synth\n"); + dev_err(card->dev, "Can't initialize Aureal wavetable synth\n"); } else { snd_vortex_synth_arg_t *arg; -- cgit v1.1 From e9600bc166d529cf03862afae51fb2e3cf987d02 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 28 Oct 2014 17:37:12 +0000 Subject: ASoC: Intel: Make ADSP memory block allocation more generic Current block allocation is tied to block type and requestor type. Make the allocation more generic by removing the struct module parameter and adding a generic block allocator structure. Also pass in the list that the blocks have to be added too in order to remove dependence on block requestor type. ASoC: Intel: update scratch allocator to use generic block allocator Update the scratch allocator to use the generic block allocator and calculate total scratch buffer size. ASoC: Intel: Add call to calculate offsets internally within the DSP. A call to calculate internal DSP memory addresses used to allocate persistent and scartch buffers. ASoC: Intel: Add runtime module support. Add support for runtime module objects that can be created for every FW module that is parsed from the FW file. This gives a 1:N mapping between the FW module from file and the runtime instantiations of that module. We also need to make sure we remove every module and runtime module when we unload the FW. ASoC: Intel: Add DMA firmware loading support Add support for DMA to load firmware modules to the DSP memory blocks. Two DMA engines are supported, DesignWare and Intel MID. ASoC: Intel: Add runtime module lookup API call Add an API to allow quick lookup of runtime modules based on ID. ASoC: Intel: Provide streams with dynamic module information Remove the hard coded module paramaters and provide each module with dynamically generated buffer information for scratch and persistent buffers. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 133 ++++-- sound/soc/intel/sst-dsp.c | 8 + sound/soc/intel/sst-dsp.h | 7 + sound/soc/intel/sst-firmware.c | 929 ++++++++++++++++++++++++++++++------- sound/soc/intel/sst-haswell-dsp.c | 59 +-- sound/soc/intel/sst-haswell-ipc.c | 112 ++--- sound/soc/intel/sst-haswell-ipc.h | 11 +- sound/soc/intel/sst-haswell-pcm.c | 83 +++- 9 files changed, 1025 insertions(+), 341 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c index fc58876..5a9e567 100644 --- a/sound/soc/intel/sst-baytrail-dsp.c +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -67,17 +67,12 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; int count; memset(&template, 0, sizeof(template)); template.id = module->type; template.entry = module->entry_point; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -94,19 +89,19 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_BYT_IRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_BYT_DRAM: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + dsp->addr.dram_offset; - block_data.type = SST_MEM_DRAM; + mod->type = SST_MEM_DRAM; break; case SST_BYT_CACHE: - block_data.offset = block->ram_offset + + mod->offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); - block_data.type = SST_MEM_CACHE; + mod->type = SST_MEM_CACHE; break; default: dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", @@ -114,11 +109,10 @@ static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); - sst_module_insert_fixed_block(mod, &block_data); + sst_module_alloc_blocks(mod); block = (void *)block + sizeof(*block) + block->size; } diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index ffb308b..be81b86 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -26,6 +26,9 @@ struct sst_mem_block; struct sst_module; struct sst_fw; +/* do we need to remove or keep */ +#define DSP_DRAM_ADDR_OFFSET 0x400000 + /* * DSP Operations exported by platform Audio DSP driver. */ @@ -67,6 +70,8 @@ struct sst_addr { u32 shim_offset; u32 iram_offset; u32 dram_offset; + u32 dsp_iram_offset; + u32 dsp_dram_offset; void __iomem *lpe; void __iomem *shim; void __iomem *pci_cfg; @@ -84,15 +89,6 @@ struct sst_mailbox { }; /* - * Audio DSP Firmware data types. - */ -enum sst_data_type { - SST_DATA_M = 0, /* module block data */ - SST_DATA_P = 1, /* peristant data (text, data) */ - SST_DATA_S = 2, /* scratch data (usually buffers) */ -}; - -/* * Audio DSP memory block types. */ enum sst_mem_type { @@ -125,23 +121,6 @@ struct sst_fw { }; /* - * Audio DSP Generic Module data. - * - * This is used to dsecribe any sections of persistent (text and data) and - * scratch (buffers) of module data in ADSP memory space. - */ -struct sst_module_data { - - enum sst_mem_type type; /* destination memory type */ - enum sst_data_type data_type; /* type of module data */ - - u32 size; /* size in bytes */ - int32_t offset; /* offset in FW file */ - u32 data_offset; /* offset in ADSP memory space */ - void *data; /* module data */ -}; - -/* * Audio DSP Generic Module Template. * * Used to define and register a new FW module. This data is extracted from @@ -150,15 +129,52 @@ struct sst_module_data { struct sst_module_template { u32 id; u32 entry; /* entry point */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; + u32 persistent_size; +}; + +/* + * Block Allocator - Used to allocate blocks of DSP memory. + */ +struct sst_block_allocator { + u32 id; + u32 offset; + int size; + enum sst_mem_type type; +}; + +/* + * Runtime Module Instance - A module object can be instanciated multiple + * times within the DSP FW. + */ +struct sst_module_runtime { + struct sst_dsp *dsp; + int id; + struct sst_module *module; /* parent module we belong too */ + + u32 persistent_offset; /* private memory offset */ + void *private; + + struct list_head list; + struct list_head block_list; /* list of blocks used */ +}; + +/* + * Runtime Module Context - The runtime context must be manually stored by the + * driver prior to enter S3 and restored after leaving S3. This should really be + * part of the memory context saved by the enter D3 message IPC ??? + */ +struct sst_module_runtime_context { + dma_addr_t dma_buffer; + u32 *buffer; }; /* * Audio DSP Generic Module. * * Each Firmware file can consist of 1..N modules. A module can span multiple - * ADSP memory blocks. The simplest FW will be a file with 1 module. + * ADSP memory blocks. The simplest FW will be a file with 1 module. A module + * can be instanciated multiple times in the DSP. */ struct sst_module { struct sst_dsp *dsp; @@ -167,10 +183,13 @@ struct sst_module { /* module configuration */ u32 id; u32 entry; /* module entry point */ - u32 offset; /* module offset in firmware file */ + s32 offset; /* module offset in firmware file */ u32 size; /* module size */ - struct sst_module_data s; /* scratch data */ - struct sst_module_data p; /* peristant data */ + u32 scratch_size; /* global scratch memory required */ + u32 persistent_size; /* private memory required */ + enum sst_mem_type type; /* destination memory type */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ /* runtime */ u32 usage_count; /* can be unloaded if count == 0 */ @@ -180,6 +199,7 @@ struct sst_module { struct list_head block_list; /* Module list of blocks in use */ struct list_head list; /* DSP list of modules */ struct list_head list_fw; /* FW list of modules */ + struct list_head runtime_list; /* list of runtime module objects*/ }; /* @@ -208,7 +228,6 @@ struct sst_mem_block { struct sst_block_ops *ops; /* block operations, if any */ /* block status */ - enum sst_data_type data_type; /* data type held in this block */ u32 bytes_used; /* bytes in use by modules */ void *private; /* generic core does not touch this */ int users; /* number of modules using this block */ @@ -253,6 +272,11 @@ struct sst_dsp { struct list_head module_list; struct list_head fw_list; + /* scratch buffer */ + struct list_head scratch_block_list; + u32 scratch_offset; + u32 scratch_size; + /* platform data */ struct sst_pdata *pdata; @@ -290,18 +314,33 @@ void sst_fw_unload(struct sst_fw *sst_fw); /* Create/Free firmware modules */ struct sst_module *sst_module_new(struct sst_fw *sst_fw, struct sst_module_template *template, void *private); -void sst_module_free(struct sst_module *sst_module); -int sst_module_insert(struct sst_module *sst_module); -int sst_module_remove(struct sst_module *sst_module); -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data); +void sst_module_free(struct sst_module *module); struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); - -/* allocate/free pesistent/scratch memory regions managed by drv */ -struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); -void sst_mem_block_free_scratch(struct sst_dsp *dsp, - struct sst_module *scratch); -int sst_block_module_remove(struct sst_module *module); +int sst_module_alloc_blocks(struct sst_module *module); +int sst_module_free_blocks(struct sst_module *module); + +/* Create/Free firmware module runtime instances */ +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private); +void sst_module_runtime_free(struct sst_module_runtime *runtime); +struct sst_module_runtime *sst_module_runtime_get_from_id( + struct sst_module *module, u32 id); +int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, + int offset); +int sst_module_runtime_free_blocks(struct sst_module_runtime *runtime); +int sst_module_runtime_save(struct sst_module_runtime *runtime, + struct sst_module_runtime_context *context); +int sst_module_runtime_restore(struct sst_module_runtime *runtime, + struct sst_module_runtime_context *context); + +/* generic block allocation */ +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list); +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list); + +/* scratch allocation */ +int sst_block_alloc_scratch(struct sst_dsp *dsp); +void sst_block_free_scratch(struct sst_dsp *dsp); /* Register the DSPs memory blocks - would be nice to read from ACPI */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, @@ -309,4 +348,10 @@ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, void *private); void sst_mem_block_unregister_all(struct sst_dsp *dsp); +/* Create/Free DMA resources */ +int sst_dma_new(struct sst_dsp *sst); +void sst_dma_free(struct sst_dma *dma); + +u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, + enum sst_mem_type type); #endif diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index cd23060..d0fc685 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -352,6 +352,7 @@ struct sst_dsp *sst_dsp_new(struct device *dev, INIT_LIST_HEAD(&sst->free_block_list); INIT_LIST_HEAD(&sst->module_list); INIT_LIST_HEAD(&sst->fw_list); + INIT_LIST_HEAD(&sst->scratch_block_list); /* Initialise SST Audio DSP */ if (sst->ops->init) { @@ -366,6 +367,10 @@ struct sst_dsp *sst_dsp_new(struct device *dev, if (err) goto irq_err; + err = sst_dma_new(sst); + if (err) + dev_warn(dev, "sst_dma_new failed %d\n", err); + return sst; irq_err: @@ -381,6 +386,9 @@ void sst_dsp_free(struct sst_dsp *sst) free_irq(sst->irq, sst); if (sst->ops->free) sst->ops->free(sst); + + if (sst->dma) + sst_dma_free(sst->dma); } EXPORT_SYMBOL_GPL(sst_dsp_free); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 3165dfa..17ee923 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -245,6 +245,13 @@ void sst_memcpy_fromio_32(struct sst_dsp *sst, /* DSP reset & boot */ void sst_dsp_reset(struct sst_dsp *sst); int sst_dsp_boot(struct sst_dsp *sst); +/* DMA */ +int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id); +void sst_dsp_dma_put_channel(struct sst_dsp *dsp); +int sst_dsp_dma_copyfrom(struct sst_dsp *sst, dma_addr_t dest_addr, + dma_addr_t src_addr, size_t size); +int sst_dsp_dma_copyto(struct sst_dsp *sst, dma_addr_t dest_addr, + dma_addr_t src_addr, size_t size); /* Msg IO */ void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg); diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index cf3d199..692a6ae 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -23,6 +23,11 @@ #include #include #include +#include + +/* supported DMA engine drivers */ +#include +#include #include #include @@ -30,7 +35,20 @@ #include "sst-dsp.h" #include "sst-dsp-priv.h" -static void block_module_remove(struct sst_module *module); +#define SST_DMA_RESOURCES 2 +#define SST_DSP_DMA_MAX_BURST 0x3 +#define SST_HSW_BLOCK_ANY 0xffffffff + +#define SST_HSW_MASK_DMA_ADDR_DSP 0xfff00000 + +struct sst_dma { + struct sst_dsp *sst; + + struct dw_dma_chip *chip; + + struct dma_async_tx_descriptor *desc; + struct dma_chan *ch; +}; static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) { @@ -38,6 +56,281 @@ static inline void sst_memcpy32(volatile void __iomem *dest, void *src, u32 byte __iowrite32_copy((void *)dest, src, bytes/4); } +static void sst_dma_transfer_complete(void *arg) +{ + struct sst_dsp *sst = (struct sst_dsp *)arg; + + dev_dbg(sst->dev, "DMA: callback\n"); +} + +static int sst_dsp_dma_copy(struct sst_dsp *sst, dma_addr_t dest_addr, + dma_addr_t src_addr, size_t size) +{ + struct dma_async_tx_descriptor *desc; + struct sst_dma *dma = sst->dma; + + if (dma->ch == NULL) { + dev_err(sst->dev, "error: no DMA channel\n"); + return -ENODEV; + } + + dev_dbg(sst->dev, "DMA: src: 0x%lx dest 0x%lx size %zu\n", + (unsigned long)src_addr, (unsigned long)dest_addr, size); + + desc = dma->ch->device->device_prep_dma_memcpy(dma->ch, dest_addr, + src_addr, size, DMA_CTRL_ACK); + if (!desc){ + dev_err(sst->dev, "error: dma prep memcpy failed\n"); + return -EINVAL; + } + + desc->callback = sst_dma_transfer_complete; + desc->callback_param = sst; + + desc->tx_submit(desc); + dma_wait_for_async_tx(desc); + + return 0; +} + +/* copy to DSP */ +int sst_dsp_dma_copyto(struct sst_dsp *sst, dma_addr_t dest_addr, + dma_addr_t src_addr, size_t size) +{ + return sst_dsp_dma_copy(sst, dest_addr | SST_HSW_MASK_DMA_ADDR_DSP, + src_addr, size); +} +EXPORT_SYMBOL_GPL(sst_dsp_dma_copyto); + +/* copy from DSP */ +int sst_dsp_dma_copyfrom(struct sst_dsp *sst, dma_addr_t dest_addr, + dma_addr_t src_addr, size_t size) +{ + return sst_dsp_dma_copy(sst, dest_addr, + src_addr | SST_HSW_MASK_DMA_ADDR_DSP, size); +} +EXPORT_SYMBOL_GPL(sst_dsp_dma_copyfrom); + +/* remove module from memory - callers hold locks */ +static void block_list_remove(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block, *tmp; + int err; + + /* disable each block */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + dev_dbg(dsp->dev, "block freed %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_list_prepare(struct sst_dsp *dsp, + struct list_head *block_list) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for data */ + list_for_each_entry(block, block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + +struct dw_dma_platform_data dw_pdata = { + .is_private = 1, + .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, + .chan_priority = CHAN_PRIORITY_ASCENDING, +}; + +static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, + int irq) +{ + struct dw_dma_chip *chip; + int err; + + chip = devm_kzalloc(dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return ERR_PTR(-ENOMEM); + + chip->irq = irq; + chip->regs = devm_ioremap_resource(dev, mem); + if (IS_ERR(chip->regs)) + return ERR_CAST(chip->regs); + + err = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(31)); + if (err) + return ERR_PTR(err); + + chip->dev = dev; + err = dw_dma_probe(chip, &dw_pdata); + if (err) + return ERR_PTR(err); + + return chip; +} + +static void dw_remove(struct dw_dma_chip *chip) +{ + dw_dma_remove(chip); +} + +static bool dma_chan_filter(struct dma_chan *chan, void *param) +{ + struct sst_dsp *dsp = (struct sst_dsp *)param; + + return chan->device->dev == dsp->dma_dev; +} + +int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id) +{ + struct sst_dma *dma = dsp->dma; + struct dma_slave_config slave; + dma_cap_mask_t mask; + int ret; + + /* The Intel MID DMA engine driver needs the slave config set but + * Synopsis DMA engine driver safely ignores the slave config */ + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_MEMCPY, mask); + + dma->ch = dma_request_channel(mask, dma_chan_filter, dsp); + if (dma->ch == NULL) { + dev_err(dsp->dev, "error: DMA request channel failed\n"); + return -EIO; + } + + memset(&slave, 0, sizeof(slave)); + slave.direction = DMA_MEM_TO_DEV; + slave.src_addr_width = + slave.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave.src_maxburst = slave.dst_maxburst = SST_DSP_DMA_MAX_BURST; + + ret = dmaengine_slave_config(dma->ch, &slave); + if (ret) { + dev_err(dsp->dev, "error: unable to set DMA slave config %d\n", + ret); + dma_release_channel(dma->ch); + dma->ch = NULL; + } + + return ret; +} +EXPORT_SYMBOL_GPL(sst_dsp_dma_get_channel); + +void sst_dsp_dma_put_channel(struct sst_dsp *dsp) +{ + struct sst_dma *dma = dsp->dma; + + if (!dma->ch) + return; + + dma_release_channel(dma->ch); + dma->ch = NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_dma_put_channel); + +int sst_dma_new(struct sst_dsp *sst) +{ + struct sst_pdata *sst_pdata = sst->pdata; + struct sst_dma *dma; + struct resource mem; + const char *dma_dev_name; + int ret = 0; + + /* configure the correct platform data for whatever DMA engine + * is attached to the ADSP IP. */ + switch (sst->pdata->dma_engine) { + case SST_DMA_TYPE_DW: + dma_dev_name = "dw_dmac"; + break; + case SST_DMA_TYPE_MID: + dma_dev_name = "Intel MID DMA"; + break; + default: + dev_err(sst->dev, "error: invalid DMA engine %d\n", + sst->pdata->dma_engine); + return -EINVAL; + } + + dma = devm_kzalloc(sst->dev, sizeof(struct sst_dma), GFP_KERNEL); + if (!dma) + return -ENOMEM; + + dma->sst = sst; + + memset(&mem, 0, sizeof(mem)); + + mem.start = sst->addr.lpe_base + sst_pdata->dma_base; + mem.end = sst->addr.lpe_base + sst_pdata->dma_base + sst_pdata->dma_size - 1; + mem.flags = IORESOURCE_MEM; + + /* now register DMA engine device */ + dma->chip = dw_probe(sst->dma_dev, &mem, sst_pdata->irq); + if (IS_ERR(dma->chip)) { + dev_err(sst->dev, "error: DMA device register failed\n"); + ret = PTR_ERR(dma->chip); + goto err_dma_dev; + } + + sst->dma = dma; + sst->fw_use_dma = true; + return 0; + +err_dma_dev: + devm_kfree(sst->dev, dma); + return ret; +} +EXPORT_SYMBOL(sst_dma_new); + +void sst_dma_free(struct sst_dma *dma) +{ + + if (dma == NULL) + return; + + if (dma->ch) + dma_release_channel(dma->ch); + + if (dma->chip) + dw_remove(dma->chip); + +} +EXPORT_SYMBOL(sst_dma_free); + /* create new generic firmware object */ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, const struct firmware *fw, void *private) @@ -68,6 +361,12 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, /* copy FW data to DMA-able memory */ memcpy((void *)sst_fw->dma_buf, (void *)fw->data, fw->size); + if (dsp->fw_use_dma) { + err = sst_dsp_dma_get_channel(dsp, 0); + if (err < 0) + goto chan_err; + } + /* call core specific FW paser to load FW data into DSP */ err = dsp->ops->parse_fw(sst_fw); if (err < 0) { @@ -75,6 +374,9 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, goto parse_err; } + if (dsp->fw_use_dma) + sst_dsp_dma_put_channel(dsp); + mutex_lock(&dsp->mutex); list_add(&sst_fw->list, &dsp->fw_list); mutex_unlock(&dsp->mutex); @@ -82,9 +384,13 @@ struct sst_fw *sst_fw_new(struct sst_dsp *dsp, return sst_fw; parse_err: - dma_free_coherent(dsp->dev, sst_fw->size, + if (dsp->fw_use_dma) + sst_dsp_dma_put_channel(dsp); +chan_err: + dma_free_coherent(dsp->dma_dev, sst_fw->size, sst_fw->dma_buf, sst_fw->dmable_fw_paddr); + sst_fw->dma_buf = NULL; kfree(sst_fw); return NULL; } @@ -108,21 +414,37 @@ EXPORT_SYMBOL_GPL(sst_fw_reload); void sst_fw_unload(struct sst_fw *sst_fw) { - struct sst_dsp *dsp = sst_fw->dsp; - struct sst_module *module, *tmp; + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_module *module, *mtmp; + struct sst_module_runtime *runtime, *rtmp; + + dev_dbg(dsp->dev, "unloading firmware\n"); - dev_dbg(dsp->dev, "unloading firmware\n"); + mutex_lock(&dsp->mutex); + + /* check module by module */ + list_for_each_entry_safe(module, mtmp, &dsp->module_list, list) { + if (module->sst_fw == sst_fw) { + + /* remove runtime modules */ + list_for_each_entry_safe(runtime, rtmp, &module->runtime_list, list) { + + block_list_remove(dsp, &runtime->block_list); + list_del(&runtime->list); + kfree(runtime); + } + + /* now remove the module */ + block_list_remove(dsp, &module->block_list); + list_del(&module->list); + kfree(module); + } + } - mutex_lock(&dsp->mutex); - list_for_each_entry_safe(module, tmp, &dsp->module_list, list) { - if (module->sst_fw == sst_fw) { - block_module_remove(module); - list_del(&module->list); - kfree(module); - } - } + /* remove all scratch blocks */ + block_list_remove(dsp, &dsp->scratch_block_list); - mutex_unlock(&dsp->mutex); + mutex_unlock(&dsp->mutex); } EXPORT_SYMBOL_GPL(sst_fw_unload); @@ -135,7 +457,8 @@ void sst_fw_free(struct sst_fw *sst_fw) list_del(&sst_fw->list); mutex_unlock(&dsp->mutex); - dma_free_coherent(dsp->dma_dev, sst_fw->size, sst_fw->dma_buf, + if (sst_fw->dma_buf) + dma_free_coherent(dsp->dma_dev, sst_fw->size, sst_fw->dma_buf, sst_fw->dmable_fw_paddr); kfree(sst_fw); } @@ -172,11 +495,11 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->id = template->id; sst_module->dsp = dsp; sst_module->sst_fw = sst_fw; - - memcpy(&sst_module->s, &template->s, sizeof(struct sst_module_data)); - memcpy(&sst_module->p, &template->p, sizeof(struct sst_module_data)); + sst_module->scratch_size = template->scratch_size; + sst_module->persistent_size = template->persistent_size; INIT_LIST_HEAD(&sst_module->block_list); + INIT_LIST_HEAD(&sst_module->runtime_list); mutex_lock(&dsp->mutex); list_add(&sst_module->list, &dsp->module_list); @@ -199,73 +522,122 @@ void sst_module_free(struct sst_module *sst_module) } EXPORT_SYMBOL_GPL(sst_module_free); -static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, - u32 offset) +struct sst_module_runtime *sst_module_runtime_new(struct sst_module *module, + int id, void *private) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_module_runtime *runtime; + + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (runtime == NULL) + return NULL; + + runtime->id = id; + runtime->dsp = dsp; + runtime->module = module; + INIT_LIST_HEAD(&runtime->block_list); + + mutex_lock(&dsp->mutex); + list_add(&runtime->list, &module->runtime_list); + mutex_unlock(&dsp->mutex); + + return runtime; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_new); + +void sst_module_runtime_free(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + list_del(&runtime->list); + mutex_unlock(&dsp->mutex); + + kfree(runtime); +} +EXPORT_SYMBOL_GPL(sst_module_runtime_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, + struct sst_block_allocator *ba) { struct sst_mem_block *block; list_for_each_entry(block, &dsp->free_block_list, list) { - if (block->type == type && block->offset == offset) + if (block->type == ba->type && block->offset == ba->offset) return block; } return NULL; } -static int block_alloc_contiguous(struct sst_module *module, - struct sst_module_data *data, u32 offset, int size) +/* Block allocator must be on block boundary */ +static int block_alloc_contiguous(struct sst_dsp *dsp, + struct sst_block_allocator *ba, struct list_head *block_list) { struct list_head tmp = LIST_HEAD_INIT(tmp); - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block; + u32 block_start = SST_HSW_BLOCK_ANY; + int size = ba->size, offset = ba->offset; - while (size > 0) { - block = find_block(dsp, data->type, offset); + while (ba->size > 0) { + + block = find_block(dsp, ba); if (!block) { list_splice(&tmp, &dsp->free_block_list); + + ba->size = size; + ba->offset = offset; return -ENOMEM; } list_move_tail(&block->list, &tmp); - offset += block->size; - size -= block->size; + ba->offset += block->size; + ba->size -= block->size; } + ba->size = size; + ba->offset = offset; + + list_for_each_entry(block, &tmp, list) { + + if (block->offset < block_start) + block_start = block->offset; + + list_add(&block->module_list, block_list); - list_for_each_entry(block, &tmp, list) - list_add(&block->module_list, &module->block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + } list_splice(&tmp, &dsp->used_block_list); return 0; } -/* allocate free DSP blocks for module data - callers hold locks */ -static int block_alloc(struct sst_module *module, - struct sst_module_data *data) +/* allocate first free DSP blocks for data - callers hold locks */ +static int block_alloc(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; int ret = 0; - if (data->size == 0) + if (ba->size == 0) return 0; /* find first free whole blocks that can hold module */ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; - if (data->size > block->size) + if (ba->size > block->size) continue; - data->offset = block->offset; - block->data_type = data->data_type; - block->bytes_used = data->size % block->size; - list_add(&block->module_list, &module->block_list); + ba->offset = block->offset; + block->bytes_used = ba->size % block->size; + list_add(&block->module_list, block_list); list_move(&block->list, &dsp->used_block_list); - dev_dbg(dsp->dev, " *module %d added block %d:%d\n", - module->id, block->type, block->index); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } @@ -273,15 +645,19 @@ static int block_alloc(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { /* ignore blocks with wrong type */ - if (block->type != data->type) + if (block->type != ba->type) continue; /* do we span > 1 blocks */ - if (data->size > block->size) { - ret = block_alloc_contiguous(module, data, - block->offset, data->size); + if (ba->size > block->size) { + + /* align ba to block boundary */ + ba->offset = block->offset; + + ret = block_alloc_contiguous(dsp, ba, block_list); if (ret == 0) return ret; + } } @@ -289,93 +665,74 @@ static int block_alloc(struct sst_module *module, return -ENOMEM; } -/* remove module from memory - callers hold locks */ -static void block_module_remove(struct sst_module *module) +int sst_alloc_blocks(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_mem_block *block, *tmp; - struct sst_dsp *dsp = module->dsp; - int err; + int ret; - /* disable each block */ - list_for_each_entry(block, &module->block_list, module_list) { + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba->size, ba->offset, ba->type); - if (block->ops && block->ops->disable) { - err = block->ops->disable(block); - if (err < 0) - dev_err(dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - } - } + mutex_lock(&dsp->mutex); - /* mark each block as free */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { - list_del(&block->module_list); - list_move(&block->list, &dsp->free_block_list); + ret = block_alloc(dsp, ba, block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc blocks %d\n", ret); + goto out; } -} -/* prepare the memory block to receive data from host - callers hold locks */ -static int block_module_prepare(struct sst_module *module) -{ - struct sst_mem_block *block; - int ret = 0; - - /* enable each block so that's it'e ready for module P/S data */ - list_for_each_entry(block, &module->block_list, module_list) { + /* prepare DSP blocks for module usage */ + ret = block_list_prepare(dsp, block_list); + if (ret < 0) + dev_err(dsp->dev, "error: prepare failed\n"); - if (block->ops && block->ops->enable) { - ret = block->ops->enable(block); - if (ret < 0) { - dev_err(module->dsp->dev, - "error: cant disable block %d:%d\n", - block->type, block->index); - goto err; - } - } - } +out: + mutex_unlock(&dsp->mutex); return ret; +} +EXPORT_SYMBOL_GPL(sst_alloc_blocks); -err: - list_for_each_entry(block, &module->block_list, module_list) { - if (block->ops && block->ops->disable) - block->ops->disable(block); - } - return ret; +int sst_free_blocks(struct sst_dsp *dsp, struct list_head *block_list) +{ + mutex_lock(&dsp->mutex); + block_list_remove(dsp, block_list); + mutex_unlock(&dsp->mutex); + return 0; } +EXPORT_SYMBOL_GPL(sst_free_blocks); /* allocate memory blocks for static module addresses - callers hold locks */ -static int block_alloc_fixed(struct sst_module *module, - struct sst_module_data *data) +static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba, + struct list_head *block_list) { - struct sst_dsp *dsp = module->dsp; struct sst_mem_block *block, *tmp; - u32 end = data->offset + data->size, block_end; + u32 end = ba->offset + ba->size, block_end; int err; /* only IRAM/DRAM blocks are managed */ - if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + if (ba->type != SST_MEM_IRAM && ba->type != SST_MEM_DRAM) return 0; /* are blocks already attached to this module */ - list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_for_each_entry_safe(block, tmp, block_list, module_list) { - /* force compacting mem blocks of the same data_type */ - if (block->data_type != data->data_type) + /* ignore blocks with wrong type */ + if (block->type != ba->type) continue; block_end = block->offset + block->size; /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) + if (ba->offset >= block->offset && end <= block_end) return 0; /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset + block->size, - data->size - block->size); + /* align ba to block boundary */ + ba->size -= block_end - ba->offset; + ba->offset = block_end; + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; @@ -388,82 +745,270 @@ static int block_alloc_fixed(struct sst_module *module, list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { block_end = block->offset + block->size; + /* ignore blocks with wrong type */ + if (block->type != ba->type) + continue; + /* find block that holds section */ - if (data->offset >= block->offset && end < block_end) { + if (ba->offset >= block->offset && end <= block_end) { /* add block */ - block->data_type = data->data_type; list_move(&block->list, &dsp->used_block_list); - list_add(&block->module_list, &module->block_list); + list_add(&block->module_list, block_list); + dev_dbg(dsp->dev, "block allocated %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); return 0; } /* does block span more than 1 section */ - if (data->offset >= block->offset && data->offset < block_end) { + if (ba->offset >= block->offset && ba->offset < block_end) { - err = block_alloc_contiguous(module, data, - block->offset, data->size); + /* align ba to block boundary */ + ba->offset = block->offset; + + err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) return -ENOMEM; return 0; } - } return -ENOMEM; } /* Load fixed module data into DSP memory blocks */ -int sst_module_insert_fixed_block(struct sst_module *module, - struct sst_module_data *data) +int sst_module_alloc_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; + struct sst_fw *sst_fw = module->sst_fw; + struct sst_block_allocator ba; int ret; + ba.size = module->size; + ba.type = module->type; + ba.offset = module->offset; + + dev_dbg(dsp->dev, "block request 0x%x bytes at offset 0x%x type %d\n", + ba.size, ba.offset, ba.type); + mutex_lock(&dsp->mutex); /* alloc blocks that includes this section */ - ret = block_alloc_fixed(module, data); + ret = block_alloc_fixed(dsp, &ba, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: no free blocks for section at offset 0x%x size 0x%x\n", - data->offset, data->size); + module->offset, module->size); mutex_unlock(&dsp->mutex); return -ENOMEM; } /* prepare DSP blocks for module copy */ - ret = block_module_prepare(module); + ret = block_list_prepare(dsp, &module->block_list); if (ret < 0) { dev_err(dsp->dev, "error: fw module prepare failed\n"); goto err; } /* copy partial module data to blocks */ - sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); + if (dsp->fw_use_dma) { + ret = sst_dsp_dma_copyto(dsp, + dsp->addr.lpe_base + module->offset, + sst_fw->dmable_fw_paddr + module->data_offset, + module->size); + if (ret < 0) { + dev_err(dsp->dev, "error: module copy failed\n"); + goto err; + } + } else + sst_memcpy32(dsp->addr.lpe + module->offset, module->data, + module->size); mutex_unlock(&dsp->mutex); return ret; err: - block_module_remove(module); + block_list_remove(dsp, &module->block_list); mutex_unlock(&dsp->mutex); return ret; } -EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); +EXPORT_SYMBOL_GPL(sst_module_alloc_blocks); /* Unload entire module from DSP memory */ -int sst_block_module_remove(struct sst_module *module) +int sst_module_free_blocks(struct sst_module *module) { struct sst_dsp *dsp = module->dsp; mutex_lock(&dsp->mutex); - block_module_remove(module); + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_module_free_blocks); + +int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, + int offset) +{ + struct sst_dsp *dsp = runtime->dsp; + struct sst_module *module = runtime->module; + struct sst_block_allocator ba; + int ret; + + if (module->persistent_size == 0) + return 0; + + ba.size = module->persistent_size; + ba.type = SST_MEM_DRAM; + + mutex_lock(&dsp->mutex); + + /* do we need to allocate at a fixed address ? */ + if (offset != 0) { + + ba.offset = offset; + + dev_dbg(dsp->dev, "persistent fixed block request 0x%x bytes type %d offset 0x%x\n", + ba.size, ba.type, ba.offset); + + /* alloc blocks that includes this section */ + ret = block_alloc_fixed(dsp, &ba, &runtime->block_list); + + } else { + dev_dbg(dsp->dev, "persistent block request 0x%x bytes type %d\n", + ba.size, ba.type); + + /* alloc blocks that includes this section */ + ret = block_alloc(dsp, &ba, &runtime->block_list); + } + if (ret < 0) { + dev_err(dsp->dev, + "error: no free blocks for runtime module size 0x%x\n", + module->persistent_size); + mutex_unlock(&dsp->mutex); + return -ENOMEM; + } + runtime->persistent_offset = ba.offset; + + /* prepare DSP blocks for module copy */ + ret = block_list_prepare(dsp, &runtime->block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: runtime block prepare failed\n"); + goto err; + } + + mutex_unlock(&dsp->mutex); + return ret; + +err: + block_list_remove(dsp, &module->block_list); + mutex_unlock(&dsp->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_alloc_blocks); + +int sst_module_runtime_free_blocks(struct sst_module_runtime *runtime) +{ + struct sst_dsp *dsp = runtime->dsp; + + mutex_lock(&dsp->mutex); + block_list_remove(dsp, &runtime->block_list); mutex_unlock(&dsp->mutex); return 0; } -EXPORT_SYMBOL_GPL(sst_block_module_remove); +EXPORT_SYMBOL_GPL(sst_module_runtime_free_blocks); + +int sst_module_runtime_save(struct sst_module_runtime *runtime, + struct sst_module_runtime_context *context) +{ + struct sst_dsp *dsp = runtime->dsp; + struct sst_module *module = runtime->module; + int ret = 0; + + dev_dbg(dsp->dev, "saving runtime %d memory at 0x%x size 0x%x\n", + runtime->id, runtime->persistent_offset, + module->persistent_size); + + context->buffer = dma_alloc_coherent(dsp->dma_dev, + module->persistent_size, + &context->dma_buffer, GFP_DMA | GFP_KERNEL); + if (!context->buffer) { + dev_err(dsp->dev, "error: DMA context alloc failed\n"); + return -ENOMEM; + } + + mutex_lock(&dsp->mutex); + + if (dsp->fw_use_dma) { + + ret = sst_dsp_dma_get_channel(dsp, 0); + if (ret < 0) + goto err; + + ret = sst_dsp_dma_copyfrom(dsp, context->dma_buffer, + dsp->addr.lpe_base + runtime->persistent_offset, + module->persistent_size); + sst_dsp_dma_put_channel(dsp); + if (ret < 0) { + dev_err(dsp->dev, "error: context copy failed\n"); + goto err; + } + } else + sst_memcpy32(context->buffer, dsp->addr.lpe + + runtime->persistent_offset, + module->persistent_size); + +err: + mutex_unlock(&dsp->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_save); + +int sst_module_runtime_restore(struct sst_module_runtime *runtime, + struct sst_module_runtime_context *context) +{ + struct sst_dsp *dsp = runtime->dsp; + struct sst_module *module = runtime->module; + int ret = 0; + + dev_dbg(dsp->dev, "restoring runtime %d memory at 0x%x size 0x%x\n", + runtime->id, runtime->persistent_offset, + module->persistent_size); + + mutex_lock(&dsp->mutex); + + if (!context->buffer) { + dev_info(dsp->dev, "no context buffer need to restore!\n"); + goto err; + } + + if (dsp->fw_use_dma) { + + ret = sst_dsp_dma_get_channel(dsp, 0); + if (ret < 0) + goto err; + + ret = sst_dsp_dma_copyto(dsp, + dsp->addr.lpe_base + runtime->persistent_offset, + context->dma_buffer, module->persistent_size); + sst_dsp_dma_put_channel(dsp); + if (ret < 0) { + dev_err(dsp->dev, "error: module copy failed\n"); + goto err; + } + } else + sst_memcpy32(dsp->addr.lpe + runtime->persistent_offset, + context->buffer, module->persistent_size); + + dma_free_coherent(dsp->dma_dev, module->persistent_size, + context->buffer, context->dma_buffer); + context->buffer = NULL; + +err: + mutex_unlock(&dsp->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_restore); /* register a DSP memory block for use with FW based modules */ struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, @@ -516,80 +1061,83 @@ void sst_mem_block_unregister_all(struct sst_dsp *dsp) EXPORT_SYMBOL_GPL(sst_mem_block_unregister_all); /* allocate scratch buffer blocks */ -struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp) +int sst_block_alloc_scratch(struct sst_dsp *dsp) { - struct sst_module *sst_module, *scratch; - struct sst_mem_block *block, *tmp; - u32 block_size; - int ret = 0; - - scratch = kzalloc(sizeof(struct sst_module), GFP_KERNEL); - if (scratch == NULL) - return NULL; + struct sst_module *module; + struct sst_block_allocator ba; + int ret; mutex_lock(&dsp->mutex); /* calculate required scratch size */ - list_for_each_entry(sst_module, &dsp->module_list, list) { - if (scratch->s.size < sst_module->s.size) - scratch->s.size = sst_module->s.size; + dsp->scratch_size = 0; + list_for_each_entry(module, &dsp->module_list, list) { + dev_dbg(dsp->dev, "module %d scratch req 0x%x bytes\n", + module->id, module->scratch_size); + if (dsp->scratch_size < module->scratch_size) + dsp->scratch_size = module->scratch_size; } - dev_dbg(dsp->dev, "scratch buffer required is %d bytes\n", - scratch->s.size); - - /* init scratch module */ - scratch->dsp = dsp; - scratch->s.type = SST_MEM_DRAM; - scratch->s.data_type = SST_DATA_S; - INIT_LIST_HEAD(&scratch->block_list); + dev_dbg(dsp->dev, "scratch buffer required is 0x%x bytes\n", + dsp->scratch_size); - /* check free blocks before looking at used blocks for space */ - if (!list_empty(&dsp->free_block_list)) - block = list_first_entry(&dsp->free_block_list, - struct sst_mem_block, list); - else - block = list_first_entry(&dsp->used_block_list, - struct sst_mem_block, list); - block_size = block->size; + if (dsp->scratch_size == 0) { + dev_info(dsp->dev, "no modules need scratch buffer\n"); + mutex_unlock(&dsp->mutex); + return 0; + } /* allocate blocks for module scratch buffers */ dev_dbg(dsp->dev, "allocating scratch blocks\n"); - ret = block_alloc(scratch, &scratch->s); + + ba.size = dsp->scratch_size; + ba.type = SST_MEM_DRAM; + + /* do we need to allocate at fixed offset */ + if (dsp->scratch_offset != 0) { + + dev_dbg(dsp->dev, "block request 0x%x bytes type %d at 0x%x\n", + ba.size, ba.type, ba.offset); + + ba.offset = dsp->scratch_offset; + + /* alloc blocks that includes this section */ + ret = block_alloc_fixed(dsp, &ba, &dsp->scratch_block_list); + + } else { + dev_dbg(dsp->dev, "block request 0x%x bytes type %d\n", + ba.size, ba.type); + + ba.offset = 0; + ret = block_alloc(dsp, &ba, &dsp->scratch_block_list); + } if (ret < 0) { dev_err(dsp->dev, "error: can't alloc scratch blocks\n"); - goto err; + mutex_unlock(&dsp->mutex); + return ret; } - /* assign the same offset of scratch to each module */ - list_for_each_entry(sst_module, &dsp->module_list, list) - sst_module->s.offset = scratch->s.offset; - - mutex_unlock(&dsp->mutex); - return scratch; + ret = block_list_prepare(dsp, &dsp->scratch_block_list); + if (ret < 0) { + dev_err(dsp->dev, "error: scratch block prepare failed\n"); + return ret; + } -err: - list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) - list_del(&block->module_list); + /* assign the same offset of scratch to each module */ + dsp->scratch_offset = ba.offset; mutex_unlock(&dsp->mutex); - return NULL; + return dsp->scratch_size; } -EXPORT_SYMBOL_GPL(sst_mem_block_alloc_scratch); +EXPORT_SYMBOL_GPL(sst_block_alloc_scratch); /* free all scratch blocks */ -void sst_mem_block_free_scratch(struct sst_dsp *dsp, - struct sst_module *scratch) +void sst_block_free_scratch(struct sst_dsp *dsp) { - struct sst_mem_block *block, *tmp; - mutex_lock(&dsp->mutex); - - list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) - list_del(&block->module_list); - + block_list_remove(dsp, &dsp->scratch_block_list); mutex_unlock(&dsp->mutex); } -EXPORT_SYMBOL_GPL(sst_mem_block_free_scratch); +EXPORT_SYMBOL_GPL(sst_block_free_scratch); /* get a module from it's unique ID */ struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id) @@ -609,3 +1157,40 @@ struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id) return NULL; } EXPORT_SYMBOL_GPL(sst_module_get_from_id); + +struct sst_module_runtime *sst_module_runtime_get_from_id( + struct sst_module *module, u32 id) +{ + struct sst_module_runtime *runtime; + struct sst_dsp *dsp = module->dsp; + + mutex_lock(&dsp->mutex); + + list_for_each_entry(runtime, &module->runtime_list, list) { + if (runtime->id == id) { + mutex_unlock(&dsp->mutex); + return runtime; + } + } + + mutex_unlock(&dsp->mutex); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_module_runtime_get_from_id); + +/* returns block address in DSP address space */ +u32 sst_dsp_get_offset(struct sst_dsp *dsp, u32 offset, + enum sst_mem_type type) +{ + switch (type) { + case SST_MEM_IRAM: + return offset - dsp->addr.iram_offset + + dsp->addr.dsp_iram_offset; + case SST_MEM_DRAM: + return offset - dsp->addr.dram_offset + + dsp->addr.dsp_dram_offset; + default: + return 0; + } +} +EXPORT_SYMBOL_GPL(sst_dsp_get_offset); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 4b6c163..5058dc8 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -42,6 +42,10 @@ #define SST_LP_SHIM_OFFSET 0xE7000 #define SST_WPT_IRAM_OFFSET 0xA0000 #define SST_LP_IRAM_OFFSET 0x80000 +#define SST_WPT_DSP_DRAM_OFFSET 0x400000 +#define SST_WPT_DSP_IRAM_OFFSET 0x00000 +#define SST_LPT_DSP_DRAM_OFFSET 0x400000 +#define SST_LPT_DSP_IRAM_OFFSET 0x00000 #define SST_SHIM_PM_REG 0x84 @@ -86,9 +90,8 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, { struct dma_block_info *block; struct sst_module *mod; - struct sst_module_data block_data; struct sst_module_template template; - int count; + int count, ret; void __iomem *ram; /* TODO: allowed module types need to be configurable */ @@ -109,13 +112,9 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, memset(&template, 0, sizeof(template)); template.id = module->type; - template.entry = module->entry_point; - template.p.size = module->info.persistent_size; - template.p.type = SST_MEM_DRAM; - template.p.data_type = SST_DATA_P; - template.s.size = module->info.scratch_size; - template.s.type = SST_MEM_DRAM; - template.s.data_type = SST_DATA_S; + template.entry = module->entry_point - 4; + template.persistent_size = module->info.persistent_size; + template.scratch_size = module->info.scratch_size; mod = sst_module_new(fw, &template, NULL); if (mod == NULL) @@ -135,14 +134,14 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, switch (block->type) { case SST_HSW_IRAM: ram = dsp->addr.lpe; - block_data.offset = + mod->offset = block->ram_offset + dsp->addr.iram_offset; - block_data.type = SST_MEM_IRAM; + mod->type = SST_MEM_IRAM; break; case SST_HSW_DRAM: ram = dsp->addr.lpe; - block_data.offset = block->ram_offset; - block_data.type = SST_MEM_DRAM; + mod->offset = block->ram_offset; + mod->type = SST_MEM_DRAM; break; default: dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", @@ -151,30 +150,34 @@ static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, return -EINVAL; } - block_data.size = block->size; - block_data.data_type = SST_DATA_M; - block_data.data = (void *)block + sizeof(*block); - block_data.data_offset = block_data.data - fw->dma_buf; + mod->size = block->size; + mod->data = (void *)block + sizeof(*block); + mod->data_offset = mod->data - fw->dma_buf; - dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + dev_dbg(dsp->dev, "module block %d type 0x%x " "size 0x%x ==> ram %p offset 0x%x\n", - count, block->type, block->size, ram, + count, mod->type, block->size, ram, block->ram_offset); - sst_module_insert_fixed_block(mod, &block_data); + ret = sst_module_alloc_blocks(mod); + if (ret < 0) { + dev_err(dsp->dev, "error: could not allocate blocks for module %d\n", + count); + sst_module_free(mod); + return ret; + } block = (void *)block + sizeof(*block) + block->size; } + return 0; } static int hsw_parse_fw_image(struct sst_fw *sst_fw) { struct fw_header *header; - struct sst_module *scratch; struct fw_module_header *module; struct sst_dsp *dsp = sst_fw->dsp; - struct sst_hsw *hsw = sst_fw->private; int ret, count; /* Read the header information from the data pointer */ @@ -204,12 +207,8 @@ static int hsw_parse_fw_image(struct sst_fw *sst_fw) module = (void *)module + sizeof(*module) + module->mod_size; } - /* allocate persistent/scratch mem regions */ - scratch = sst_mem_block_alloc_scratch(dsp); - if (scratch == NULL) - return -ENOMEM; - - sst_hsw_set_scratch_module(hsw, scratch); + /* allocate scratch mem regions */ + sst_block_alloc_scratch(dsp); return 0; } @@ -467,12 +466,16 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) region = lp_region; region_count = ARRAY_SIZE(lp_region); sst->addr.iram_offset = SST_LP_IRAM_OFFSET; + sst->addr.dsp_iram_offset = SST_LPT_DSP_IRAM_OFFSET; + sst->addr.dsp_dram_offset = SST_LPT_DSP_DRAM_OFFSET; sst->addr.shim_offset = SST_LP_SHIM_OFFSET; break; case SST_DEV_ID_WILDCAT_POINT: region = wpt_region; region_count = ARRAY_SIZE(wpt_region); sst->addr.iram_offset = SST_WPT_IRAM_OFFSET; + sst->addr.dsp_iram_offset = SST_WPT_DSP_IRAM_OFFSET; + sst->addr.dsp_dram_offset = SST_WPT_DSP_DRAM_OFFSET; sst->addr.shim_offset = SST_WPT_SHIM_OFFSET; break; default: diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 4799768..770d467 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1351,10 +1351,11 @@ int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream, } int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, - u32 entry_point) + struct sst_hsw_stream *stream, struct sst_module_runtime *runtime) { struct sst_hsw_module_map *map = &stream->request.map; + struct sst_dsp *dsp = sst_hsw_get_dsp(hsw); + struct sst_module *module = runtime->module; if (stream->commited) { dev_err(hsw->dev, "error: stream committed for set module\n"); @@ -1363,36 +1364,25 @@ int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, /* only support initial module atm */ map->module_entries_count = 1; - map->module_entries[0].module_id = module_id; - map->module_entries[0].entry_point = entry_point; - - return 0; -} - -int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 offset, u32 size) -{ - if (stream->commited) { - dev_err(hsw->dev, "error: stream committed for set pmem\n"); - return -EINVAL; - } - - stream->request.persistent_mem.offset = offset; - stream->request.persistent_mem.size = size; - - return 0; -} - -int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 offset, u32 size) -{ - if (stream->commited) { - dev_err(hsw->dev, "error: stream committed for set smem\n"); - return -EINVAL; - } - - stream->request.scratch_mem.offset = offset; - stream->request.scratch_mem.size = size; + map->module_entries[0].module_id = module->id; + map->module_entries[0].entry_point = module->entry; + + stream->request.persistent_mem.offset = + sst_dsp_get_offset(dsp, runtime->persistent_offset, SST_MEM_DRAM); + stream->request.persistent_mem.size = module->persistent_size; + + stream->request.scratch_mem.offset = + sst_dsp_get_offset(dsp, dsp->scratch_offset, SST_MEM_DRAM); + stream->request.scratch_mem.size = dsp->scratch_size; + + dev_dbg(hsw->dev, "module %d runtime %d using:\n", module->id, + runtime->id); + dev_dbg(hsw->dev, " persistent offset 0x%x bytes 0x%x\n", + stream->request.persistent_mem.offset, + stream->request.persistent_mem.size); + dev_dbg(hsw->dev, " scratch offset 0x%x bytes 0x%x\n", + stream->request.scratch_mem.offset, + stream->request.scratch_mem.size); return 0; } @@ -1673,32 +1663,48 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", dx->entries_no, state); - memcpy(&hsw->dx, dx, sizeof(*dx)); - return 0; + return ret; } -/* Used to save state into hsw->dx_reply */ -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source) +struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, + int mod_id, int offset) { - struct sst_hsw_ipc_dx_memory_item *dx_mem; - struct sst_hsw_ipc_dx_reply *dx_reply; - int entry_no; + struct sst_dsp *dsp = hsw->dsp; + struct sst_module *module; + struct sst_module_runtime *runtime; + int err; - dx_reply = &hsw->dx; - entry_no = dx_reply->entries_no; + module = sst_module_get_from_id(dsp, mod_id); + if (module == NULL) { + dev_err(dsp->dev, "error: failed to get module %d for pcm\n", + mod_id); + return NULL; + } - trace_ipc_request("PM get Dx state", entry_no); + runtime = sst_module_runtime_new(module, mod_id, NULL); + if (runtime == NULL) { + dev_err(dsp->dev, "error: failed to create module %d runtime\n", + mod_id); + return NULL; + } - if (item >= entry_no) - return -EINVAL; + err = sst_module_runtime_alloc_blocks(runtime, offset); + if (err < 0) { + dev_err(dsp->dev, "error: failed to alloc blocks for module %d runtime\n", + mod_id); + sst_module_runtime_free(runtime); + return NULL; + } - dx_mem = &dx_reply->mem_info[item]; - *offset = dx_mem->offset; - *size = dx_mem->size; - *source = dx_mem->source; + dev_dbg(dsp->dev, "runtime id %d created for module %d\n", runtime->id, + mod_id); + return runtime; +} - return 0; +void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime) +{ + sst_module_runtime_free_blocks(runtime); + sst_module_runtime_free(runtime); } static int msg_empty_list_init(struct sst_hsw *hsw) @@ -1718,12 +1724,6 @@ static int msg_empty_list_init(struct sst_hsw *hsw) return 0; } -void sst_hsw_set_scratch_module(struct sst_hsw *hsw, - struct sst_module *scratch) -{ - hsw->scratch = scratch; -} - struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) { return hsw->dsp; diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 063dd6b..fe6e63f 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -40,6 +40,7 @@ struct sst_hsw_stream; struct sst_hsw_log_stream; struct sst_pdata; struct sst_module; +struct sst_module_runtime; extern struct sst_ops haswell_ops; /* Stream Allocate Path ID */ @@ -432,8 +433,7 @@ int sst_hsw_stream_set_map_config(struct sst_hsw *hsw, int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream, enum sst_hsw_interleaving style); int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, - u32 entry_point); + struct sst_hsw_stream *stream, struct sst_module_runtime *runtime); int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, @@ -486,7 +486,10 @@ int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); -void sst_hsw_set_scratch_module(struct sst_hsw *hsw, - struct sst_module *scratch); + +/* runtime module management */ +struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, + int mod_id, int offset); +void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime); #endif diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 32a6470..522edef 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -89,16 +89,23 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { .buffer_bytes_max = HSW_PCM_PERIODS_MAX * PAGE_SIZE, }; +struct hsw_pcm_module_map { + int dai_id; + enum sst_hsw_module_id mod_id; +}; + /* private data for each PCM DSP stream */ struct hsw_pcm_data { int dai_id; struct sst_hsw_stream *stream; + struct sst_module_runtime *runtime; u32 volume[2]; struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; unsigned int wpos; struct mutex mutex; bool allocated; + int persistent_offset; }; /* private data for the driver */ @@ -472,28 +479,8 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* we use hardcoded memory offsets atm, will be updated for new FW */ - if (stream_type == SST_HSW_STREAM_TYPE_CAPTURE) { - sst_hsw_stream_set_module_info(hsw, pcm_data->stream, - SST_HSW_MODULE_PCM_CAPTURE, module_data->entry); - sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, - 0x449400, 0x4000); - sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, - 0x400000, 0); - } else { /* stream_type == SST_HSW_STREAM_TYPE_SYSTEM */ - sst_hsw_stream_set_module_info(hsw, pcm_data->stream, - SST_HSW_MODULE_PCM_SYSTEM, module_data->entry); - - sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, - module_data->offset, module_data->size); - sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, - 0x44d400, 0x3800); - - sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, - module_data->offset, module_data->size); - sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, - 0x400000, 0); - } + sst_hsw_stream_set_module_info(hsw, pcm_data->stream, + pcm_data->runtime); ret = sst_hsw_stream_commit(hsw, pcm_data->stream); if (ret < 0) { @@ -654,6 +641,55 @@ static struct snd_pcm_ops hsw_pcm_ops = { .page = snd_pcm_sgbuf_ops_page, }; +/* static mappings between PCMs and modules - may be dynamic in future */ +static struct hsw_pcm_module_map mod_map[] = { + {0, SST_HSW_MODULE_PCM_SYSTEM}, /* "System Pin" */ + {1, SST_HSW_MODULE_PCM}, /* "Offload0 Pin" */ + {2, SST_HSW_MODULE_PCM}, /* "Offload1 Pin" */ + {3, SST_HSW_MODULE_PCM_REFERENCE}, /* "Loopback Pin" */ + {4, SST_HSW_MODULE_PCM_CAPTURE}, /* "Capture Pin" */ +}; + +static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) +{ + struct sst_hsw *hsw = pdata->hsw; + struct hsw_pcm_data *pcm_data; + int i; + + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[i]; + + pcm_data->runtime = sst_hsw_runtime_module_create(hsw, + mod_map[i].mod_id, pcm_data->persistent_offset); + if (pcm_data->runtime == NULL) + goto err; + pcm_data->persistent_offset = + pcm_data->runtime->persistent_offset; + } + + return 0; + +err: + for (--i; i >= 0; i--) { + pcm_data = &pdata->pcm[i]; + sst_hsw_runtime_module_free(pcm_data->runtime); + } + + return -ENODEV; +} + +static void hsw_pcm_free_modules(struct hsw_priv_data *pdata) +{ + struct hsw_pcm_data *pcm_data; + int i; + + for (i = 0; i < ARRAY_SIZE(mod_map); i++) { + pcm_data = &pdata->pcm[i]; + + sst_hsw_runtime_module_free(pcm_data->runtime); + } +} + static void hsw_pcm_free(struct snd_pcm *pcm) { snd_pcm_lib_preallocate_free_for_all(pcm); @@ -797,6 +833,9 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) } } + /* allocate runtime modules */ + hsw_pcm_create_modules(priv_data); + return 0; err: -- cgit v1.1 From 4e44923847b0b2597eaef07d5e700f5dbed2162e Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Tue, 28 Oct 2014 17:08:40 +0100 Subject: ASoC: cs42l51: make driver user-selectable Since we are removing the Armada 370 DB audio machine driver to use the 'simple-card' Device Tree binding, we can no longer select the CS42L51 codec driver using a Kconfig 'select', and we instead need it to be user-selectable. Therefore, this commit adds a prompt to make the CS42L51 I2C codec driver user-selectable. Signed-off-by: Thomas Petazzoni Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..f4fb12f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -336,7 +336,7 @@ config SND_SOC_CS42L51 tristate config SND_SOC_CS42L51_I2C - tristate + tristate "Cirrus Logic CS42L51 CODEC (I2C)" select SND_SOC_CS42L51 config SND_SOC_CS42L52 -- cgit v1.1 From e6f6ebc1f8f60d6d44f6be22c6386c238d6a9d97 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 22 Oct 2014 16:11:39 +0800 Subject: ASoC: rt5677: Add TDM channel mapping function It is for channel to slot mapping, and it is not only for 8 channels mapping, but also in 2, 4 and 6 channels mapping. If we want to use the 2 channels in the stereo2 adc path, we need to set the item "2/1/3/4" or "2/3/1/4". It also adds for stereo channel swap. It can map the sterero channels "L/R" to "R/L", "L/L" or R/R. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 239 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 38 +++++++- 2 files changed, 262 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 413bccb..ca264f8 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1834,6 +1834,93 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if4_adc_mux = SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum); +/* TDM IF1/2 ADC Data Selection */ /* MX-3B MX-40 [7:6][5:4][3:2][1:0] */ +static const char * const rt5677_if12_adc_swap_src[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc1_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC1_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC1 Swap Source", rt5677_if1_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc2_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc2_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if1_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc3_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc3_swap_mux = + SOC_DAPM_ENUM("IF1 ADC3 Swap Source", rt5677_if1_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc4_swap_enum, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc4_swap_mux = + SOC_DAPM_ENUM("IF1 ADC4 Swap Source", rt5677_if1_adc4_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc1_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF1_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc1_swap_mux = + SOC_DAPM_ENUM("IF1 ADC2 Swap Source", rt5677_if2_adc1_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc2_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC2_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc2_swap_mux = + SOC_DAPM_ENUM("IF2 ADC2 Swap Source", rt5677_if2_adc2_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc3_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC3_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc3_swap_mux = + SOC_DAPM_ENUM("IF2 ADC3 Swap Source", rt5677_if2_adc3_swap_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc4_swap_enum, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC4_SWAP_SFT, rt5677_if12_adc_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = + SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); + +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +static const char * const rt5677_if1_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "3/1/2/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_adc_tdm_swap_enum, RT5677_TDM1_CTRL2, + RT5677_IF1_ADC_CTRL_SFT, rt5677_if1_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if1_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF1 ADC TDM Swap Source", rt5677_if1_adc_tdm_swap_enum); + +/* TDM IF2 ADC Data Selection */ /* MX-41[2:0] */ +static const char * const rt5677_if2_adc_tdm_swap_src[] = { + "1/2/3/4", "2/1/3/4", "3/1/2/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", + "2/3/1/4", "3/4/1/2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_adc_tdm_swap_enum, RT5677_TDM2_CTRL2, + RT5677_IF2_ADC_CTRL_SFT, rt5677_if2_adc_tdm_swap_src); + +static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = + SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1945,6 +2032,52 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_if1_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM1_CTRL2, &value); + if (value & RT5677_IF1_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, + RT5677_IF1_ADC_MODE_MASK, + RT5677_IF1_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + unsigned int value; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(rt5677->regmap, RT5677_TDM2_CTRL2, &value); + if (value & RT5677_IF2_ADC_CTRL_MASK) + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, + RT5677_IF2_ADC_MODE_MASK, + RT5677_IF2_ADC_MODE_TDM); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), @@ -2104,10 +2237,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_PGA("Stereo4 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), /* DSP */ SND_SOC_DAPM_MUX("IB9 Mux", SND_SOC_NOPM, 0, 0, @@ -2230,6 +2361,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if1_adc3_mux), SND_SOC_DAPM_MUX("IF1 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if1_adc4_mux), + SND_SOC_DAPM_MUX("IF1 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF1 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF1 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_adc_tdm_swap_mux, rt5677_if1_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF2 ADC1 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc1_mux), SND_SOC_DAPM_MUX("IF2 ADC2 Mux", SND_SOC_NOPM, 0, 0, @@ -2238,6 +2380,17 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { &rt5677_if2_adc3_mux), SND_SOC_DAPM_MUX("IF2 ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_if2_adc4_mux), + SND_SOC_DAPM_MUX("IF2 ADC1 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc1_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC2 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc2_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC3 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc3_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC4 Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc4_swap_mux), + SND_SOC_DAPM_MUX_E("IF2 ADC TDM Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_adc_tdm_swap_mux, rt5677_if2_adc_tdm_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("IF3 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5677_if3_adc_mux), SND_SOC_DAPM_MUX("IF4 ADC Mux", SND_SOC_NOPM, 0, 0, @@ -2621,11 +2774,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 ADC4 Mux", "OB67", "OB67" }, { "IF1 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF1 ADC1 Swap Mux", "L/R", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "L/L", "IF1 ADC1 Mux" }, + { "IF1 ADC1 Swap Mux", "R/R", "IF1 ADC1 Mux" }, + + { "IF1 ADC2 Swap Mux", "L/R", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "L/L", "IF1 ADC2 Mux" }, + { "IF1 ADC2 Swap Mux", "R/R", "IF1 ADC2 Mux" }, + + { "IF1 ADC3 Swap Mux", "L/R", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "L/L", "IF1 ADC3 Mux" }, + { "IF1 ADC3 Swap Mux", "R/R", "IF1 ADC3 Mux" }, + + { "IF1 ADC4 Swap Mux", "L/R", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "L/L", "IF1 ADC4 Mux" }, + { "IF1 ADC4 Swap Mux", "R/R", "IF1 ADC4 Mux" }, + + { "IF1 ADC", NULL, "IF1 ADC1 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC2 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC3 Swap Mux" }, + { "IF1 ADC", NULL, "IF1 ADC4 Swap Mux" }, + + { "IF1 ADC TDM Swap Mux", "1/2/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/1/3/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "2/3/1/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "4/1/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/3/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "1/4/2/3", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/1/2/4", "IF1 ADC" }, + { "IF1 ADC TDM Swap Mux", "3/4/1/2", "IF1 ADC" }, + { "AIF1TX", NULL, "I2S1" }, - { "AIF1TX", NULL, "IF1 ADC1 Mux" }, - { "AIF1TX", NULL, "IF1 ADC2 Mux" }, - { "AIF1TX", NULL, "IF1 ADC3 Mux" }, - { "AIF1TX", NULL, "IF1 ADC4 Mux" }, + { "AIF1TX", NULL, "IF1 ADC TDM Swap Mux" }, { "IF2 ADC1 Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF2 ADC1 Mux", "OB01", "OB01 Bypass Mux" }, @@ -2642,11 +2826,42 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 ADC4 Mux", "OB67", "OB67" }, { "IF2 ADC4 Mux", "OB01", "OB01 Bypass Mux" }, + { "IF2 ADC1 Swap Mux", "L/R", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "L/L", "IF2 ADC1 Mux" }, + { "IF2 ADC1 Swap Mux", "R/R", "IF2 ADC1 Mux" }, + + { "IF2 ADC2 Swap Mux", "L/R", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "L/L", "IF2 ADC2 Mux" }, + { "IF2 ADC2 Swap Mux", "R/R", "IF2 ADC2 Mux" }, + + { "IF2 ADC3 Swap Mux", "L/R", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "L/L", "IF2 ADC3 Mux" }, + { "IF2 ADC3 Swap Mux", "R/R", "IF2 ADC3 Mux" }, + + { "IF2 ADC4 Swap Mux", "L/R", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "L/L", "IF2 ADC4 Mux" }, + { "IF2 ADC4 Swap Mux", "R/R", "IF2 ADC4 Mux" }, + + { "IF2 ADC", NULL, "IF2 ADC1 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC2 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC3 Swap Mux" }, + { "IF2 ADC", NULL, "IF2 ADC4 Swap Mux" }, + + { "IF2 ADC TDM Swap Mux", "1/2/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/1/3/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/1/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "4/1/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/3/2/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "1/4/2/3", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "2/3/1/4", "IF2 ADC" }, + { "IF2 ADC TDM Swap Mux", "3/4/1/2", "IF2 ADC" }, + { "AIF2TX", NULL, "I2S2" }, - { "AIF2TX", NULL, "IF2 ADC1 Mux" }, - { "AIF2TX", NULL, "IF2 ADC2 Mux" }, - { "AIF2TX", NULL, "IF2 ADC3 Mux" }, - { "AIF2TX", NULL, "IF2 ADC4 Mux" }, + { "AIF2TX", NULL, "IF2 ADC TDM Swap Mux" }, { "IF3 ADC Mux", "STO1 ADC MIX", "Stereo1 ADC MIX" }, { "IF3 ADC Mux", "STO2 ADC MIX", "Stereo2 ADC MIX" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d2c743c..2f5b8c6 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,21 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3C TDM1 control 1 (0x3c) */ +/* MX3B TDM1 control 1 (0x3b) */ +#define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF1_ADC_MODE_SFT 12 +#define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF1_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF1_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF1_ADC1_SWAP_SFT 6 +#define RT5677_IF1_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF1_ADC2_SWAP_SFT 4 +#define RT5677_IF1_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF1_ADC3_SWAP_SFT 2 +#define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF1_ADC4_SWAP_SFT 0 + +/* MX3C TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -808,8 +822,24 @@ #define RT5677_IF1_ADC2_SFT 6 #define RT5677_IF1_ADC1_MASK (0x3 << 4) #define RT5677_IF1_ADC1_SFT 4 - -/* MX41 TDM2 control 1 (0x41) */ +#define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF1_ADC_CTRL_SFT 0 + +/* MX40 TDM2 control 1 (0x40) */ +#define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) +#define RT5677_IF2_ADC_MODE_SFT 12 +#define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) +#define RT5677_IF2_ADC_MODE_TDM (0x1 << 12) +#define RT5677_IF2_ADC1_SWAP_MASK (0x3 << 6) +#define RT5677_IF2_ADC1_SWAP_SFT 6 +#define RT5677_IF2_ADC2_SWAP_MASK (0x3 << 4) +#define RT5677_IF2_ADC2_SWAP_SFT 4 +#define RT5677_IF2_ADC3_SWAP_MASK (0x3 << 2) +#define RT5677_IF2_ADC3_SWAP_SFT 2 +#define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) +#define RT5677_IF2_ADC4_SWAP_SFT 0 + +/* MX41 TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -818,6 +848,8 @@ #define RT5677_IF2_ADC2_SFT 6 #define RT5677_IF2_ADC1_MASK (0x3 << 4) #define RT5677_IF2_ADC1_SFT 4 +#define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) +#define RT5677_IF2_ADC_CTRL_SFT 0 /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) -- cgit v1.1 From e83280f96f108ee7af8c5669080cf664b0c2b8fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Oct 2014 08:20:46 +0100 Subject: ALSA: doc: Fix enum snd_jack_types comments Follow the proper kerneldoc rule, and complete enum item comments. Signed-off-by: Takashi Iwai --- include/sound/jack.h | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 37e612e..67f2bbc 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -28,7 +28,21 @@ struct input_dev; /** - * enum snd_jack_types: Jack types which can be reported + * enum snd_jack_types - Jack types which can be reported + * @SND_JACK_HEADPHONE: Headphone + * @SND_JACK_MICROPHONE: Microphone + * @SND_JACK_HEADSET: Headset + * @SND_JACK_LINEOUT: Line out + * @SND_JACK_MECHANICAL: Mechanical switch + * @SND_JACK_VIDEOOUT: Video out + * @SND_JACK_AVOUT: AV (Audio Video) out + * @SND_JACK_LINEIN: Line in + * @SND_JACK_BTN_0: Button 0 + * @SND_JACK_BTN_1: Button 1 + * @SND_JACK_BTN_2: Button 2 + * @SND_JACK_BTN_3: Button 3 + * @SND_JACK_BTN_4: Button 4 + * @SND_JACK_BTN_5: Button 5 * * These values are used as a bitmask. * -- cgit v1.1 From f533ccb61edf008e14c9e1b91b48cd2c7397f33d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Oct 2014 08:22:05 +0100 Subject: ALSA: doc: Fix uapi/sound/compress_offload.h kerneldoc comments so that make htmldocs works properly. Since kerneldoc can't handle noname enum properly, name enum sndrv_compress_encoder. Signed-off-by: Takashi Iwai --- include/uapi/sound/compress_offload.h | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 1964026..22ed8cb 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -32,7 +32,7 @@ #define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) /** - * struct snd_compressed_buffer: compressed buffer + * struct snd_compressed_buffer - compressed buffer * @fragment_size: size of buffer fragment in bytes * @fragments: number of such fragments */ @@ -42,7 +42,7 @@ struct snd_compressed_buffer { } __attribute__((packed, aligned(4))); /** - * struct snd_compr_params: compressed stream params + * struct snd_compr_params - compressed stream params * @buffer: buffer description * @codec: codec parameters * @no_wake_mode: dont wake on fragment elapsed @@ -54,7 +54,7 @@ struct snd_compr_params { } __attribute__((packed, aligned(4))); /** - * struct snd_compr_tstamp: timestamp descriptor + * struct snd_compr_tstamp - timestamp descriptor * @byte_offset: Byte offset in ring buffer to DSP * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by @@ -73,7 +73,7 @@ struct snd_compr_tstamp { } __attribute__((packed, aligned(4))); /** - * struct snd_compr_avail: avail descriptor + * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading * @tstamp: timestamp infomation */ @@ -88,7 +88,7 @@ enum snd_compr_direction { }; /** - * struct snd_compr_caps: caps descriptor + * struct snd_compr_caps - caps descriptor * @codecs: pointer to array of codecs * @direction: direction supported. Of type snd_compr_direction * @min_fragment_size: minimum fragment supported by DSP @@ -110,7 +110,7 @@ struct snd_compr_caps { } __attribute__((packed, aligned(4))); /** - * struct snd_compr_codec_caps: query capability of codec + * struct snd_compr_codec_caps - query capability of codec * @codec: codec for which capability is queried * @num_descriptors: number of codec descriptors * @descriptor: array of codec capability descriptor @@ -122,18 +122,19 @@ struct snd_compr_codec_caps { } __attribute__((packed, aligned(4))); /** + * enum sndrv_compress_encoder * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the * end of the track * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the * beginning of the track */ -enum { +enum sndrv_compress_encoder { SNDRV_COMPRESS_ENCODER_PADDING = 1, SNDRV_COMPRESS_ENCODER_DELAY = 2, }; /** - * struct snd_compr_metadata: compressed stream metadata + * struct snd_compr_metadata - compressed stream metadata * @key: key id * @value: key value */ -- cgit v1.1 From 00dad6cfd25b75cc9cc24b8abfc263177e581ae2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Oct 2014 08:23:31 +0100 Subject: ALSA: doc: Include uapi/sound/compress_*.h, too Signed-off-by: Takashi Iwai --- Documentation/DocBook/alsa-driver-api.tmpl | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 13f8b24..7bec270 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -99,6 +99,8 @@ Compress Offload Compress Offload API !Esound/core/compress_offload.c +!Iinclude/uapi/sound/compress_offload.h +!Iinclude/uapi/sound/compress_params.h !Iinclude/sound/compress_driver.h -- cgit v1.1 From 63ae1fe7739ec81eb63ad241b4e217d1fa0e8e53 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Oct 2014 10:33:31 +0000 Subject: ASoC: Intel: Add dependency on DesignWare DMA controller We have calls into the controller so we need to ensure it is being built. Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..ae7f872 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI depends on (X86 || COMPILE_TEST) + depends on DW_DMAC_CORE help This adds support for Intel(R) Smart Sound Technology (SST). Say Y if you have such a device -- cgit v1.1 From 7077148fb50a120d20a50516a332ed6eb9233c16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 Oct 2014 22:15:31 +0000 Subject: ASoC: core: Split ops out of soc-core.c The main ASoC source file is getting quite large and the standard ops don't really have anything to do with the rest of the file so split them out into a separate file. Signed-off-by: Mark Brown --- sound/soc/Makefile | 2 +- sound/soc/soc-core.c | 919 ------------------------------------------------- sound/soc/soc-ops.c | 952 +++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 953 insertions(+), 920 deletions(-) create mode 100644 sound/soc/soc-ops.c diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..a384d14 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 47c378a..a2b51ed 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2334,925 +2334,6 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); /** - * snd_soc_info_enum_double - enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double enumerated - * mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - - return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, - e->items, e->texts); -} -EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); - -/** - * snd_soc_get_enum_double - enumerated double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; - unsigned int reg_val; - int ret; - - ret = snd_soc_component_read(component, e->reg, ®_val); - if (ret) - return ret; - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[0] = item; - if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); - ucontrol->value.enumerated.item[1] = item; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); - -/** - * snd_soc_put_enum_double - enumerated double mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double enumerated mixer. - * - * Returns 0 for success. - */ -int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int *item = ucontrol->value.enumerated.item; - unsigned int val; - unsigned int mask; - - if (item[0] >= e->items) - return -EINVAL; - val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (item[1] >= e->items) - return -EINVAL; - val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; - } - - return snd_soc_component_update_bits(component, e->reg, mask, val); -} -EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); - -/** - * snd_soc_read_signed - Read a codec register and interprete as signed value - * @component: component - * @reg: Register to read - * @mask: Mask to use after shifting the register value - * @shift: Right shift of register value - * @sign_bit: Bit that describes if a number is negative or not. - * @signed_val: Pointer to where the read value should be stored - * - * This functions reads a codec register. The register value is shifted right - * by 'shift' bits and masked with the given 'mask'. Afterwards it translates - * the given registervalue into a signed integer if sign_bit is non-zero. - * - * Returns 0 on sucess, otherwise an error value - */ -static int snd_soc_read_signed(struct snd_soc_component *component, - unsigned int reg, unsigned int mask, unsigned int shift, - unsigned int sign_bit, int *signed_val) -{ - int ret; - unsigned int val; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - val = (val >> shift) & mask; - - if (!sign_bit) { - *signed_val = val; - return 0; - } - - /* non-negative number */ - if (!(val & BIT(sign_bit))) { - *signed_val = val; - return 0; - } - - ret = val; - - /* - * The register most probably does not contain a full-sized int. - * Instead we have an arbitrary number of bits in a signed - * representation which has to be translated into a full-sized int. - * This is done by filling up all bits above the sign-bit. - */ - ret |= ~((int)(BIT(sign_bit) - 1)); - - *signed_val = ret; - - return 0; -} - -/** - * snd_soc_info_volsw - single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single mixer control, or a double - * mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw); - -/** - * snd_soc_get_volsw - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int val; - int ret; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = val - min; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - - if (snd_soc_volsw_is_stereo(mc)) { - if (reg == reg2) - ret = snd_soc_read_signed(component, reg, mask, rshift, - sign_bit, &val); - else - ret = snd_soc_read_signed(component, reg2, mask, shift, - sign_bit, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = val - min; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw); - -/** - * snd_soc_put_volsw - single mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - unsigned int sign_bit = mc->sign_bit; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - int err; - bool type_2r = false; - unsigned int val2 = 0; - unsigned int val, val_mask; - - if (sign_bit) - mask = BIT(sign_bit + 1) - 1; - - val = ((ucontrol->value.integer.value[0] + min) & mask); - if (invert) - val = max - val; - val_mask = mask << shift; - val = val << shift; - if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); - if (invert) - val2 = max - val2; - if (reg == reg2) { - val_mask |= mask << rshift; - val |= val2 << rshift; - } else { - val2 = val2 << shift; - type_2r = true; - } - } - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (type_2r) - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw); - -/** - * snd_soc_get_volsw_sx - single mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a single mixer control, or a double mixer - * control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret < 0) - return ret; - - ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, reg2, &val); - if (ret < 0) - return ret; - - val = ((val >> rshift) - min) & mask; - ucontrol->value.integer.value[1] = val; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); - -/** - * snd_soc_put_volsw_sx - double mixer set callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to set the value of a double mixer control that spans 2 registers. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; - int max = mc->max; - int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; - int err = 0; - unsigned int val, val_mask, val2 = 0; - - val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; - val = val << shift; - - err = snd_soc_component_update_bits(component, reg, val_mask, val); - if (err < 0) - return err; - - if (snd_soc_volsw_is_stereo(mc)) { - val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; - val2 = val2 << rshift; - - err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); - } - return err; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); - -/** - * snd_soc_info_volsw_range - single mixer info callback with range. - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information, within a range, about a single - * mixer control. - * - * returns 0 for success. - */ -int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; - int min = mc->min; - - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - min; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); - -/** - * snd_soc_put_volsw_range - single mixer put value callback with range. - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value, within a range, for a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val, val_mask; - int ret; - - if (invert) - val = (max - ucontrol->value.integer.value[0]) & mask; - else - val = ((ucontrol->value.integer.value[0] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; - - if (snd_soc_volsw_is_stereo(mc)) { - if (invert) - val = (max - ucontrol->value.integer.value[1]) & mask; - else - val = ((ucontrol->value.integer.value[1] + min) & mask); - val_mask = mask << shift; - val = val << shift; - - ret = snd_soc_component_update_bits(component, rreg, val_mask, - val); - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); - -/** - * snd_soc_get_volsw_range - single mixer get callback with range - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value, within a range, of a single mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int rreg = mc->rreg; - unsigned int shift = mc->shift; - int min = mc->min; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[0] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; - else - ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[0] - min; - - if (snd_soc_volsw_is_stereo(mc)) { - ret = snd_soc_component_read(component, rreg, &val); - if (ret) - return ret; - - ucontrol->value.integer.value[1] = (val >> shift) & mask; - if (invert) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - else - ucontrol->value.integer.value[1] = - ucontrol->value.integer.value[1] - min; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); - -/** - * snd_soc_limit_volume - Set new limit to an existing volume control. - * - * @codec: where to look for the control - * @name: Name of the control - * @max: new maximum limit - * - * Return 0 for success, else error. - */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, - const char *name, int max) -{ - struct snd_card *card = codec->component.card->snd_card; - struct snd_kcontrol *kctl; - struct soc_mixer_control *mc; - int found = 0; - int ret = -EINVAL; - - /* Sanity check for name and max */ - if (unlikely(!name || max <= 0)) - return -EINVAL; - - list_for_each_entry(kctl, &card->controls, list) { - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { - found = 1; - break; - } - } - if (found) { - mc = (struct soc_mixer_control *)kctl->private_value; - if (max <= mc->max) { - mc->platform_max = max; - ret = 0; - } - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_limit_volume); - -int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - uinfo->count = params->num_regs * component->val_bytes; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info); - -int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret; - - if (component->regmap) - ret = regmap_raw_read(component->regmap, params->base, - ucontrol->value.bytes.data, - params->num_regs * component->val_bytes); - else - ret = -EINVAL; - - /* Hide any masked bytes to ensure consistent data reporting */ - if (ret == 0 && params->mask) { - switch (component->val_bytes) { - case 1: - ucontrol->value.bytes.data[0] &= ~params->mask; - break; - case 2: - ((u16 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be16(~params->mask); - break; - case 4: - ((u32 *)(&ucontrol->value.bytes.data))[0] - &= cpu_to_be32(~params->mask); - break; - default: - return -EINVAL; - } - } - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_get); - -int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_bytes *params = (void *)kcontrol->private_value; - int ret, len; - unsigned int val, mask; - void *data; - - if (!component->regmap || !params->num_regs) - return -EINVAL; - - len = params->num_regs * component->val_bytes; - - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; - - /* - * If we've got a mask then we need to preserve the register - * bits. We shouldn't modify the incoming data so take a - * copy. - */ - if (params->mask) { - ret = regmap_read(component->regmap, params->base, &val); - if (ret != 0) - goto out; - - val &= params->mask; - - switch (component->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u16 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(component->regmap, - &mask, &mask); - if (ret != 0) - goto out; - - ((u32 *)data)[0] &= mask; - - ret = regmap_parse_val(component->regmap, - &val, &val); - if (ret != 0) - goto out; - - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; - goto out; - } - } - - ret = regmap_raw_write(component->regmap, params->base, - data, len); - -out: - kfree(data); - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_put); - -int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *ucontrol) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - - ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; - ucontrol->count = params->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); - -int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *tlv) -{ - struct soc_bytes_ext *params = (void *)kcontrol->private_value; - unsigned int count = size < params->max ? size : params->max; - int ret = -ENXIO; - - switch (op_flag) { - case SNDRV_CTL_TLV_OP_READ: - if (params->get) - ret = params->get(tlv, count); - break; - case SNDRV_CTL_TLV_OP_WRITE: - if (params->put) - ret = params->put(tlv, count); - break; - } - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); - -/** - * snd_soc_info_xr_sx - signed multi register info callback - * @kcontrol: mreg control - * @uinfo: control element information - * - * Callback to provide information of a control that can - * span multiple codec registers which together - * forms a single signed value in a MSB/LSB manner. - * - * Returns 0 for success. - */ -int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = mc->min; - uinfo->value.integer.max = mc->max; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); - -/** - * snd_soc_get_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to get the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<invert; - unsigned long mask = (1UL<nbits)-1; - long min = mc->min; - long max = mc->max; - long val = 0; - unsigned int regval; - unsigned int i; - int ret; - - for (i = 0; i < regcount; i++) { - ret = snd_soc_component_read(component, regbase+i, ®val); - if (ret) - return ret; - val |= (regval & regwmask) << (regwshift*(regcount-i-1)); - } - val &= mask; - if (min < 0 && val > max) - val |= ~mask; - if (invert) - val = max - val; - ucontrol->value.integer.value[0] = val; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); - -/** - * snd_soc_put_xr_sx - signed multi register get callback - * @kcontrol: mreg control - * @ucontrol: control element information - * - * Callback to set the value of a control that can span - * multiple codec registers which together forms a single - * signed value in a MSB/LSB manner. The control supports - * specifying total no of bits used to allow for bitfields - * across the multiple codec registers. - * - * Returns 0 for success. - */ -int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mreg_control *mc = - (struct soc_mreg_control *)kcontrol->private_value; - unsigned int regbase = mc->regbase; - unsigned int regcount = mc->regcount; - unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<invert; - unsigned long mask = (1UL<nbits)-1; - long max = mc->max; - long val = ucontrol->value.integer.value[0]; - unsigned int i, regval, regmask; - int err; - - if (invert) - val = max - val; - val &= mask; - for (i = 0; i < regcount; i++) { - regval = (val >> (regwshift*(regcount-i-1))) & regwmask; - regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; - err = snd_soc_component_update_bits(component, regbase+i, - regmask, regval); - if (err < 0) - return err; - } - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); - -/** - * snd_soc_get_strobe - strobe get callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback get the value of a strobe mixer control. - * - * Returns 0 for success. - */ -int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int val; - int ret; - - ret = snd_soc_component_read(component, reg, &val); - if (ret) - return ret; - - val &= mask; - - if (shift != 0 && val != 0) - val = val >> shift; - ucontrol->value.enumerated.item[0] = val ^ invert; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_strobe); - -/** - * snd_soc_put_strobe - strobe put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback strobe a register bit to high then low (or the inverse) - * in one pass of a single mixer enum control. - * - * Returns 1 for success. - */ -int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = 1 << shift; - unsigned int invert = mc->invert != 0; - unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; - unsigned int val1 = (strobe ^ invert) ? mask : 0; - unsigned int val2 = (strobe ^ invert) ? 0 : mask; - int err; - - err = snd_soc_component_update_bits(component, reg, mask, val1); - if (err < 0) - return err; - - return snd_soc_component_update_bits(component, reg, mask, val2); -} -EXPORT_SYMBOL_GPL(snd_soc_put_strobe); - -/** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI * @clk_id: DAI specific clock ID diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c new file mode 100644 index 0000000..100d92b --- /dev/null +++ b/sound/soc/soc-ops.c @@ -0,0 +1,952 @@ +/* + * soc-ops.c -- Generic ASoC operations + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Author: Liam Girdwood + * with code, comments and ideas from :- + * Richard Purdie + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +/** + * snd_soc_info_enum_double - enumerated double mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a double enumerated + * mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + return snd_ctl_enum_info(uinfo, e->shift_l == e->shift_r ? 1 : 2, + e->items, e->texts); +} +EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); + +/** + * snd_soc_get_enum_double - enumerated double mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, item; + unsigned int reg_val; + int ret; + + ret = snd_soc_component_read(component, e->reg, ®_val); + if (ret) + return ret; + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[0] = item; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[1] = item; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); + +/** + * snd_soc_put_enum_double - enumerated double mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a double enumerated mixer. + * + * Returns 0 for success. + */ +int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val; + unsigned int mask; + + if (item[0] >= e->items) + return -EINVAL; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + mask = e->mask << e->shift_l; + if (e->shift_l != e->shift_r) { + if (item[1] >= e->items) + return -EINVAL; + val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; + mask |= e->mask << e->shift_r; + } + + return snd_soc_component_update_bits(component, e->reg, mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); + +/** + * snd_soc_read_signed - Read a codec register and interprete as signed value + * @component: component + * @reg: Register to read + * @mask: Mask to use after shifting the register value + * @shift: Right shift of register value + * @sign_bit: Bit that describes if a number is negative or not. + * @signed_val: Pointer to where the read value should be stored + * + * This functions reads a codec register. The register value is shifted right + * by 'shift' bits and masked with the given 'mask'. Afterwards it translates + * the given registervalue into a signed integer if sign_bit is non-zero. + * + * Returns 0 on sucess, otherwise an error value + */ +static int snd_soc_read_signed(struct snd_soc_component *component, + unsigned int reg, unsigned int mask, unsigned int shift, + unsigned int sign_bit, int *signed_val) +{ + int ret; + unsigned int val; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + val = (val >> shift) & mask; + + if (!sign_bit) { + *signed_val = val; + return 0; + } + + /* non-negative number */ + if (!(val & BIT(sign_bit))) { + *signed_val = val; + return 0; + } + + ret = val; + + /* + * The register most probably does not contain a full-sized int. + * Instead we have an arbitrary number of bits in a signed + * representation which has to be translated into a full-sized int. + * This is done by filling up all bits above the sign-bit. + */ + ret |= ~((int)(BIT(sign_bit) - 1)); + + *signed_val = ret; + + return 0; +} + +/** + * snd_soc_info_volsw - single mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a single mixer control, or a double + * mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - mc->min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw); + +/** + * snd_soc_get_volsw - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int val; + int ret; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + ret = snd_soc_read_signed(component, reg, mask, shift, sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = val - min; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + + if (snd_soc_volsw_is_stereo(mc)) { + if (reg == reg2) + ret = snd_soc_read_signed(component, reg, mask, rshift, + sign_bit, &val); + else + ret = snd_soc_read_signed(component, reg2, mask, shift, + sign_bit, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = val - min; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw); + +/** + * snd_soc_put_volsw - single mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + unsigned int sign_bit = mc->sign_bit; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + int err; + bool type_2r = false; + unsigned int val2 = 0; + unsigned int val, val_mask; + + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + if (snd_soc_volsw_is_stereo(mc)) { + val2 = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val2 = max - val2; + if (reg == reg2) { + val_mask |= mask << rshift; + val |= val2 << rshift; + } else { + val2 = val2 << shift; + type_2r = true; + } + } + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (type_2r) + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw); + +/** + * snd_soc_get_volsw_sx - single mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a single mixer control, or a double mixer + * control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret < 0) + return ret; + + ucontrol->value.integer.value[0] = ((val >> shift) - min) & mask; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, reg2, &val); + if (ret < 0) + return ret; + + val = ((val >> rshift) - min) & mask; + ucontrol->value.integer.value[1] = val; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); + +/** + * snd_soc_put_volsw_sx - double mixer set callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a double mixer control that spans 2 registers. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mask = (1 << (fls(min + max) - 1)) - 1; + int err = 0; + unsigned int val, val_mask, val2 = 0; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] + min) & mask; + val = val << shift; + + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + + if (snd_soc_volsw_is_stereo(mc)) { + val_mask = mask << rshift; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + val2 = val2 << rshift; + + err = snd_soc_component_update_bits(component, reg2, val_mask, + val2); + } + return err; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); + +/** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; + int ret; + + if (invert) + val = (max - ucontrol->value.integer.value[0]) & mask; + else + val = ((ucontrol->value.integer.value[0] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, reg, val_mask, val); + if (ret < 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + if (invert) + val = (max - ucontrol->value.integer.value[1]) & mask; + else + val = ((ucontrol->value.integer.value[1] + min) & mask); + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_component_update_bits(component, rreg, val_mask, + val); + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[0] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + else + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; + + if (snd_soc_volsw_is_stereo(mc)) { + ret = snd_soc_component_read(component, rreg, &val); + if (ret) + return ret; + + ucontrol->value.integer.value[1] = (val >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + else + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** + * snd_soc_limit_volume - Set new limit to an existing volume control. + * + * @codec: where to look for the control + * @name: Name of the control + * @max: new maximum limit + * + * Return 0 for success, else error. + */ +int snd_soc_limit_volume(struct snd_soc_codec *codec, + const char *name, int max) +{ + struct snd_card *card = codec->component.card->snd_card; + struct snd_kcontrol *kctl; + struct soc_mixer_control *mc; + int found = 0; + int ret = -EINVAL; + + /* Sanity check for name and max */ + if (unlikely(!name || max <= 0)) + return -EINVAL; + + list_for_each_entry(kctl, &card->controls, list) { + if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { + found = 1; + break; + } + } + if (found) { + mc = (struct soc_mixer_control *)kctl->private_value; + if (max <= mc->max) { + mc->platform_max = max; + ret = 0; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_limit_volume); + +int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = params->num_regs * component->val_bytes; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info); + +int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret; + + if (component->regmap) + ret = regmap_raw_read(component->regmap, params->base, + ucontrol->value.bytes.data, + params->num_regs * component->val_bytes); + else + ret = -EINVAL; + + /* Hide any masked bytes to ensure consistent data reporting */ + if (ret == 0 && params->mask) { + switch (component->val_bytes) { + case 1: + ucontrol->value.bytes.data[0] &= ~params->mask; + break; + case 2: + ((u16 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be16(~params->mask); + break; + case 4: + ((u32 *)(&ucontrol->value.bytes.data))[0] + &= cpu_to_be32(~params->mask); + break; + default: + return -EINVAL; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_get); + +int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + int ret, len; + unsigned int val, mask; + void *data; + + if (!component->regmap || !params->num_regs) + return -EINVAL; + + len = params->num_regs * component->val_bytes; + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + /* + * If we've got a mask then we need to preserve the register + * bits. We shouldn't modify the incoming data so take a + * copy. + */ + if (params->mask) { + ret = regmap_read(component->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= params->mask; + + switch (component->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u16 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(component->regmap, + &mask, &mask); + if (ret != 0) + goto out; + + ((u32 *)data)[0] &= mask; + + ret = regmap_parse_val(component->regmap, + &val, &val); + if (ret != 0) + goto out; + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; + } + } + + ret = regmap_raw_write(component->regmap, params->base, + data, len); + +out: + kfree(data); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_put); + +int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BYTES; + ucontrol->count = params->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); + +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + +/** + * snd_soc_info_xr_sx - signed multi register info callback + * @kcontrol: mreg control + * @uinfo: control element information + * + * Callback to provide information of a control that can + * span multiple codec registers which together + * forms a single signed value in a MSB/LSB manner. + * + * Returns 0 for success. + */ +int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = mc->min; + uinfo->value.integer.max = mc->max; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_xr_sx); + +/** + * snd_soc_get_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to get the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<invert; + unsigned long mask = (1UL<nbits)-1; + long min = mc->min; + long max = mc->max; + long val = 0; + unsigned int regval; + unsigned int i; + int ret; + + for (i = 0; i < regcount; i++) { + ret = snd_soc_component_read(component, regbase+i, ®val); + if (ret) + return ret; + val |= (regval & regwmask) << (regwshift*(regcount-i-1)); + } + val &= mask; + if (min < 0 && val > max) + val |= ~mask; + if (invert) + val = max - val; + ucontrol->value.integer.value[0] = val; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_xr_sx); + +/** + * snd_soc_put_xr_sx - signed multi register get callback + * @kcontrol: mreg control + * @ucontrol: control element information + * + * Callback to set the value of a control that can span + * multiple codec registers which together forms a single + * signed value in a MSB/LSB manner. The control supports + * specifying total no of bits used to allow for bitfields + * across the multiple codec registers. + * + * Returns 0 for success. + */ +int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mreg_control *mc = + (struct soc_mreg_control *)kcontrol->private_value; + unsigned int regbase = mc->regbase; + unsigned int regcount = mc->regcount; + unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; + unsigned int regwmask = (1<invert; + unsigned long mask = (1UL<nbits)-1; + long max = mc->max; + long val = ucontrol->value.integer.value[0]; + unsigned int i, regval, regmask; + int err; + + if (invert) + val = max - val; + val &= mask; + for (i = 0; i < regcount; i++) { + regval = (val >> (regwshift*(regcount-i-1))) & regwmask; + regmask = (mask >> (regwshift*(regcount-i-1))) & regwmask; + err = snd_soc_component_update_bits(component, regbase+i, + regmask, regval); + if (err < 0) + return err; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); + +/** + * snd_soc_get_strobe - strobe get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback get the value of a strobe mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int val; + int ret; + + ret = snd_soc_component_read(component, reg, &val); + if (ret) + return ret; + + val &= mask; + + if (shift != 0 && val != 0) + val = val >> shift; + ucontrol->value.enumerated.item[0] = val ^ invert; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_strobe); + +/** + * snd_soc_put_strobe - strobe put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback strobe a register bit to high then low (or the inverse) + * in one pass of a single mixer enum control. + * + * Returns 1 for success. + */ +int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = 1 << shift; + unsigned int invert = mc->invert != 0; + unsigned int strobe = ucontrol->value.enumerated.item[0] != 0; + unsigned int val1 = (strobe ^ invert) ? mask : 0; + unsigned int val2 = (strobe ^ invert) ? 0 : mask; + int err; + + err = snd_soc_component_update_bits(component, reg, mask, val1); + if (err < 0) + return err; + + return snd_soc_component_update_bits(component, reg, mask, val2); +} +EXPORT_SYMBOL_GPL(snd_soc_put_strobe); -- cgit v1.1 From 36bcecd0a73eb4a11c9748bc96c2d254d5364d12 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Oct 2014 13:55:44 +0200 Subject: ASoC: davinci-mcasp: Correct TX start sequence Follow the sequence described in the TRMs when starting TX. This sequence will make sure that we are not facing with initial channel swap caused by no data available in McASP for transmit. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 25 +++++++++---------------- sound/soc/davinci/davinci-mcasp.h | 6 ++++++ 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0eed9b1..e1c1f40 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -183,31 +183,24 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) static void mcasp_start_tx(struct davinci_mcasp *mcasp) { - u8 offset = 0, i; u32 cnt; + /* Start clocks */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + /* Activate serializer(s) */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); - mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); - mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); - for (i = 0; i < mcasp->num_serializer; i++) { - if (mcasp->serial_dir[i] == TX_MODE) { - offset = i; - break; - } - } - - /* wait for TX ready */ + /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(offset)) & - TXSTATE) && (cnt < 100000)) + while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & + ~XRDATA) && (cnt < 100000)) cnt++; - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXBUF_REG, 0); + /* Release TX state machine */ + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + /* Release Frame Sync generator */ + mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 98fbc45..9737108 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -253,6 +253,12 @@ #define TXFSRST BIT(12) /* Frame Sync Generator Reset */ /* + * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits + * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits + */ +#define XRDATA BIT(5) /* Transmit/Receive data ready */ + +/* * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits */ #define MUTENA(val) (val) -- cgit v1.1 From 4498273551d4e27c93d3585bc7e4676623c46da8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Oct 2014 13:55:45 +0200 Subject: ASoC: davinci-mcasp: Correct RX start sequence Follow the sequence described in the TRMs when starting RX. Write to RXBUF register was not correct and there is no need to release the RX state machine/Receive frame sync generator twice. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e1c1f40..142da94 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -154,9 +154,9 @@ static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) static void mcasp_start_rx(struct davinci_mcasp *mcasp) { + /* Start clocks */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); - /* * When ASYNC == 0 the transmit and receive sections operate * synchronously from the transmit clock and frame sync. We need to make @@ -167,16 +167,12 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); } + /* Activate serializer(s) */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - - mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); - mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXBUF_REG, 0); - + /* Release RX state machine */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); - if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); } -- cgit v1.1 From 0380866a9131646787dc60d19a6d5d2c22dffdd1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Oct 2014 13:55:46 +0200 Subject: ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step The AFIFO should not be stopped (or started for that matter) when McASP is running since it can cause unpredictable issues because we are switching off AFIFO for the direction which was handling the requests from McASP and was generating DMA request toward the system DMA. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 142da94..002351f 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -233,6 +233,12 @@ static void mcasp_stop_rx(struct davinci_mcasp *mcasp) mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + + if (mcasp->rxnumevt) { /* disable FIFO */ + u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + } } static void mcasp_stop_tx(struct davinci_mcasp *mcasp) @@ -248,27 +254,22 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val); mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + + if (mcasp->txnumevt) { /* disable FIFO */ + u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + } } static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { - u32 reg; - mcasp->streams--; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (mcasp->txnumevt) { /* disable FIFO */ - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); - } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) mcasp_stop_tx(mcasp); - } else { - if (mcasp->rxnumevt) { /* disable FIFO */ - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); - } + else mcasp_stop_rx(mcasp); - } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, -- cgit v1.1 From bb372af0f7040fb637bfe0859aaa0ba49018506b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Oct 2014 13:55:47 +0200 Subject: ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions In this way the start code for tx/rx going to be located at the same place. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 002351f..6b1bfd9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -154,6 +154,13 @@ static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp) static void mcasp_start_rx(struct davinci_mcasp *mcasp) { + if (mcasp->rxnumevt) { /* enable FIFO */ + u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; + + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); + } + /* Start clocks */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); @@ -181,6 +188,13 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) { u32 cnt; + if (mcasp->txnumevt) { /* enable FIFO */ + u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; + + mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); + mcasp_set_bits(mcasp, reg, FIFO_ENABLE); + } + /* Start clocks */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); @@ -201,25 +215,12 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { - u32 reg; - mcasp->streams++; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (mcasp->txnumevt) { /* enable FIFO */ - reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); - mcasp_set_bits(mcasp, reg, FIFO_ENABLE); - } + if (stream == SNDRV_PCM_STREAM_PLAYBACK) mcasp_start_tx(mcasp); - } else { - if (mcasp->rxnumevt) { /* enable FIFO */ - reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET; - mcasp_clr_bits(mcasp, reg, FIFO_ENABLE); - mcasp_set_bits(mcasp, reg, FIFO_ENABLE); - } + else mcasp_start_rx(mcasp); - } } static void mcasp_stop_rx(struct davinci_mcasp *mcasp) -- cgit v1.1 From a11e9b168646cfc5d3b8d605d430d7e4ff267d72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Oct 2014 15:06:01 +0100 Subject: ALSA: hda - Correct kerneldoc comments Complete the missing parameters and fix anything wrong there. Just comment changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 15 +++++- sound/pci/hda/hda_codec.c | 114 ++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_jack.c | 21 ++++++++ 4 files changed, 146 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index fcc5e47..7388958 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -464,8 +464,12 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr); /** * hda_get_input_pin_label - Give a label for the given input pin + * @codec: the HDA codec + * @item: ping config item to refer + * @pin: the pin NID + * @check_location: flag to add the jack location prefix * - * When check_location is true, the function checks the pin location + * When @check_location is true, the function checks the pin location * for mic and line-in pins, and set an appropriate prefix like "Front", * "Rear", "Internal". */ @@ -550,6 +554,9 @@ static int check_mic_location_need(struct hda_codec *codec, /** * hda_get_autocfg_input_label - Get a label for the given input + * @codec: the HDA codec + * @cfg: the parsed pin configuration + * @input: the input index number * * Get a label for the given input pin defined by the autocfg item. * Unlike hda_get_input_pin_label(), this function checks all inputs @@ -677,6 +684,12 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_get_pin_label - Get a label for the given I/O pin + * @codec: the HDA codec + * @nid: pin NID + * @cfg: the parsed pin configuration + * @label: the string buffer to store + * @maxlen: the max length of string buffer (including termination) + * @indexp: the pointer to return the index number (for multiple ctls) * * Get a label for the given pin. This function works for both input and * output pins. When @cfg is given as non-NULL, the function tries to get diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0025bf4..e152542 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -416,7 +416,6 @@ static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid) * snd_hda_get_conn_list - get connection list * @codec: the HDA codec * @nid: NID to parse - * @len: number of connection list entries * @listp: the pointer to store NID list * * Parses the connection list of the given widget and stores the pointer @@ -2004,6 +2003,7 @@ EXPORT_SYMBOL_GPL(query_amp_caps); * @codec: the HD-audio codec * @nid: the NID to query * @dir: either #HDA_INPUT or #HDA_OUTPUT + * @bits: bit mask to check the result * * Check whether the widget has the given amp capability for the direction. */ @@ -2023,7 +2023,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_caps); * snd_hda_override_amp_caps - Override the AMP capabilities * @codec: the CODEC to clean up * @nid: the NID to clean up - * @direction: either #HDA_INPUT or #HDA_OUTPUT + * @dir: either #HDA_INPUT or #HDA_OUTPUT * @caps: the capability bits to set * * Override the cached AMP caps bits value by the given one. @@ -2320,6 +2320,8 @@ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, /** * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2381,6 +2383,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2406,6 +2410,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_get); /** * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -2436,6 +2442,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); /** * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @_tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3012,6 +3022,8 @@ EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3031,6 +3043,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_info); /** * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3057,6 +3071,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get); /** * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_COMPOSE_AMP_VAL*() or related macros. @@ -3100,6 +3116,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); /** * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3123,6 +3141,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_get); /** * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_MUTE*() macros. @@ -3153,6 +3173,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_switch_put); /** * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control + * @kcontrol: referred ctl element + * @uinfo: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3176,6 +3198,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_info); /** * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3199,6 +3223,8 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_get); /** * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. @@ -3228,6 +3254,10 @@ EXPORT_SYMBOL_GPL(snd_hda_mixer_bind_ctls_put); /** * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control + * @kcontrol: ctl element + * @op_flag: operation flag + * @size: byte size of input TLV + * @tlv: TLV data * * The control element is supposed to have the private_value field * set up via HDA_BIND_VOL() macro. @@ -4305,6 +4335,7 @@ static struct hda_rate_tbl rate_bits[] = { * @channels: the number of channels * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) * @maxbps: the max. bps + * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant) * * Calculate the format bitset from the given rate, channels and th PCM format. * @@ -4980,6 +5011,7 @@ static void __snd_hda_power_down(struct hda_codec *codec) * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change + * @d3wait: sync for D3 transition complete * * Change the power-up counter via @delta, and power up or down the hardware * appropriately. For the power-down, queue to the delayed action. @@ -5055,6 +5087,10 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); /** * snd_hda_ch_mode_info - Info callback helper for the channel mode enum + * @codec: the HDA codec + * @uinfo: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size */ int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, @@ -5074,6 +5110,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); /** * snd_hda_ch_mode_get - Get callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channels: max number of channels */ int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5095,6 +5136,11 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); /** * snd_hda_ch_mode_put - Put callback helper for the channel mode enum + * @codec: the HDA codec + * @ucontrol: pointer to get/store the data + * @chmode: channel mode array + * @num_chmodes: channel mode array size + * @max_channelsp: pointer to store the max channels */ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, @@ -5123,6 +5169,8 @@ EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); /** * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * @imux: imux helper object + * @uinfo: pointer to get/store the data */ int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) @@ -5144,6 +5192,11 @@ EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * @codec: the HDA codec + * @imux: imux helper object + * @ucontrol: pointer to get/store the data + * @nid: input mux NID + * @cur_val: pointer to get/store the current imux value */ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, @@ -5168,7 +5221,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_input_mux_put); -/* +/** + * snd_hda_enum_helper_info - Helper for simple enum ctls + * @kcontrol: ctl element + * @uinfo: pointer to get/store the data + * @num_items: number of enum items + * @texts: enum item string array + * * process kcontrol info callback of a simple string enum array * when @num_items is 0 or @texts is NULL, assume a boolean enum array */ @@ -5257,6 +5316,8 @@ EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); /** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5273,6 +5334,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_open); /** * snd_hda_multi_out_dig_prepare - prepare the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign */ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -5289,6 +5355,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_prepare); /** * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5302,6 +5370,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_cleanup); /** * snd_hda_multi_out_dig_close - release the digital out stream + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5315,6 +5385,10 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close); /** * snd_hda_multi_out_analog_open - open analog outputs + * @codec: the HDA codec + * @mout: hda_multi_out object + * @substream: PCM substream to assign + * @hinfo: PCM information to assign * * Open analog outputs and set up the hw-constraints. * If the digital outputs can be opened as slave, open the digital @@ -5365,6 +5439,11 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_open); /** * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * @codec: the HDA codec + * @mout: hda_multi_out object + * @stream_tag: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign * * Set up the i/o for analog out. * When the digital out is available, copy the front out to digital out, too. @@ -5442,6 +5521,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_prepare); /** * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out + * @codec: the HDA codec + * @mout: hda_multi_out object */ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -5473,6 +5554,8 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_analog_cleanup); /** * snd_hda_get_default_vref - Get the default (mic) VREF pin bits + * @codec: the HDA codec + * @pin: referred pin NID * * Guess the suitable VREF pin bits to be set as the pin-control value. * Note: the function doesn't set the AC_PINCTL_IN_EN bit. @@ -5498,7 +5581,12 @@ unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin) } EXPORT_SYMBOL_GPL(snd_hda_get_default_vref); -/* correct the pin ctl value for matching with the pin cap */ +/** + * snd_hda_correct_pin_ctl - correct the pin ctl value for matching with the pin cap + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin ctl value to audit + */ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val) { @@ -5549,6 +5637,19 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); +/** + * _snd_hda_pin_ctl - Helper to set pin ctl value + * @codec: the HDA codec + * @pin: referred pin NID + * @val: pin control value to set + * @cached: access over codec pinctl cache or direct write + * + * This function is a helper to set a pin ctl value more safely. + * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the + * value in pin target array via snd_hda_codec_set_pin_target(), then + * actually writes the value via either snd_hda_codec_update_cache() or + * snd_hda_codec_write() depending on @cached flag. + */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val, bool cached) { @@ -5565,6 +5666,11 @@ EXPORT_SYMBOL_GPL(_snd_hda_set_pin_ctl); /** * snd_hda_add_imux_item - Add an item to input_mux + * @codec: the HDA codec + * @imux: imux helper object + * @label: the name of imux item to assign + * @index: index number of imux item to assign + * @type_idx: pointer to store the resultant label index * * When the same label is used already in the existing items, the number * suffix is appended to the label. This label index number is stored diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index e1cd34d..0e6d753 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -371,7 +371,7 @@ error: return ret; } -/** +/* * SNDRV_PCM_RATE_* and AC_PAR_PCM values don't match, print correct rates with * hdmi-specific routine. */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index f56765a..b2d81ab 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -57,6 +57,8 @@ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) /** * snd_hda_jack_tbl_get - query the jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get(struct hda_codec *codec, hda_nid_t nid) @@ -75,6 +77,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get); /** * snd_hda_jack_tbl_get_from_tag - query the jack-table entry for the given tag + * @codec: the HDA codec + * @tag: tag value to refer to */ struct hda_jack_tbl * snd_hda_jack_tbl_get_from_tag(struct hda_codec *codec, unsigned char tag) @@ -93,6 +97,8 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_tbl_get_from_tag); /** * snd_hda_jack_tbl_new - create a jack-table entry for the given NID + * @codec: the HDA codec + * @nid: pin NID to assign */ static struct hda_jack_tbl * snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) @@ -162,6 +168,7 @@ static void jack_detect_update(struct hda_codec *codec, /** * snd_hda_set_dirty_all - Mark all the cached as dirty + * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. * It's called from the resume path in hda_codec.c. @@ -218,6 +225,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection + * @codec: the HDA codec + * @nid: pin NID to enable + * @func: callback function to register * * In the case of error, the return value will be a pointer embedded with * errno. Check and handle the return value appropriately with standard @@ -266,6 +276,9 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable); /** * snd_hda_jack_set_gating_jack - Set gating jack. + * @codec: the HDA codec + * @gated_nid: gated pin NID + * @gating_nid: gating pin NID * * Indicates the gated jack is only valid when the gating jack is plugged. */ @@ -287,6 +300,7 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_set_gating_jack); /** * snd_hda_jack_report_sync - sync the states of all jacks and report if changed + * @codec: the HDA codec */ void snd_hda_jack_report_sync(struct hda_codec *codec) { @@ -349,6 +363,11 @@ static void hda_free_jack_priv(struct snd_jack *jack) /** * snd_hda_jack_add_kctl - Add a kctl for the given pin + * @codec: the HDA codec + * @nid: pin NID to assign + * @name: string name for the jack + * @idx: index number for the jack + * @phantom_jack: flag to deal as a phantom jack * * This assigns a jack-detection kctl to the given pin. The kcontrol * will have the given name and index. @@ -456,6 +475,8 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, /** * snd_hda_jack_add_kctls - Add kctls for all pins included in the given pincfg + * @codec: the HDA codec + * @cfg: pin config table to parse */ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) -- cgit v1.1 From 95a962c36f6e3c3edb438d1ba59e30964900d16a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 Oct 2014 16:03:58 +0100 Subject: ALSA: hda - More kerneldoc comments Put more kerneldoc comments to the exported functions. Still the generic parser code and the HD-audio controller code aren't covered yet, though. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 51 +++++++++++++- sound/pci/hda/hda_beep.c | 38 +++++++++++ sound/pci/hda/hda_codec.c | 148 ++++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_jack.c | 39 +++++++++++ sound/pci/hda/hda_jack.h | 5 ++ sound/pci/hda/hda_sysfs.c | 35 ++++++++-- 6 files changed, 297 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7388958..1ede822 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -441,6 +441,13 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_parse_pin_defcfg); +/** + * snd_hda_get_input_pin_attr - Get the input pin attribute from pin config + * @def_conf: pin configuration value + * + * Guess the input pin attribute (INPUT_PIN_ATTR_XXX) from the given + * default pin configuration value. + */ int snd_hda_get_input_pin_attr(unsigned int def_conf) { unsigned int loc = get_defcfg_location(def_conf); @@ -473,7 +480,6 @@ EXPORT_SYMBOL_GPL(snd_hda_get_input_pin_attr); * for mic and line-in pins, and set an appropriate prefix like "Front", * "Rear", "Internal". */ - static const char *hda_get_input_pin_label(struct hda_codec *codec, const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) @@ -761,6 +767,14 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_get_pin_label); +/** + * snd_hda_add_verbs - Add verbs to the init list + * @codec: the HDA codec + * @list: zero-terminated verb list to add + * + * Append the given verb list to the execution list. The verbs will be + * performed at init and resume time via snd_hda_apply_verbs(). + */ int snd_hda_add_verbs(struct hda_codec *codec, const struct hda_verb *list) { @@ -773,6 +787,10 @@ int snd_hda_add_verbs(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_verbs); +/** + * snd_hda_apply_verbs - Execute the init verb lists + * @codec: the HDA codec + */ void snd_hda_apply_verbs(struct hda_codec *codec) { int i; @@ -783,6 +801,11 @@ void snd_hda_apply_verbs(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_apply_verbs); +/** + * snd_hda_apply_pincfgs - Set each pin config in the given list + * @codec: the HDA codec + * @cfg: NULL-terminated pin config table + */ void snd_hda_apply_pincfgs(struct hda_codec *codec, const struct hda_pintbl *cfg) { @@ -850,6 +873,11 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } } +/** + * snd_hda_apply_fixup - Apply the fixup chain with the given action + * @codec: the HDA codec + * @action: fixup action (HDA_FIXUP_ACT_XXX) + */ void snd_hda_apply_fixup(struct hda_codec *codec, int action) { if (codec->fixup_list) @@ -868,6 +896,12 @@ static bool pin_config_match(struct hda_codec *codec, return true; } +/** + * snd_hda_pick_pin_fixup - Pick up a fixup matching with the pin quirk list + * @codec: the HDA codec + * @pin_quirk: zero-terminated pin quirk list + * @fixlist: the fixup list + */ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist) @@ -894,6 +928,21 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_pick_pin_fixup); +/** + * snd_hda_pick_fixup - Pick up a fixup matching with PCI/codec SSID or model string + * @codec: the HDA codec + * @models: NULL-terminated model string list + * @quirk: zero-terminated PCI/codec SSID quirk list + * @fixlist: the fixup list + * + * Pick up a fixup entry matching with the given model string or SSID. + * If a fixup was already set beforehand, the function doesn't do anything. + * When a special model string "nofixup" is given, also no fixup is applied. + * + * The function tries to find the matching model name at first, if given. + * If nothing matched, try to look up the PCI SSID. + * If still nothing matched, try to look up the codec SSID. + */ void snd_hda_pick_fixup(struct hda_codec *codec, const struct hda_model_fixup *models, const struct snd_pci_quirk *quirk, diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 8c6c50a..1e7de08 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -175,6 +175,11 @@ static int snd_hda_do_attach(struct hda_beep *beep) return 0; } +/** + * snd_hda_enable_beep_device - Turn on/off beep sound + * @codec: the HDA codec + * @enable: flag to turn on/off + */ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) { struct hda_beep *beep = codec->beep; @@ -191,6 +196,20 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) } EXPORT_SYMBOL_GPL(snd_hda_enable_beep_device); +/** + * snd_hda_attach_beep_device - Attach a beep input device + * @codec: the HDA codec + * @nid: beep NID + * + * Attach a beep object to the given widget. If beep hint is turned off + * explicitly or beep_mode of the codec is turned off, this doesn't nothing. + * + * The attached beep device has to be registered via + * snd_hda_register_beep_device() and released via snd_hda_detach_beep_device() + * appropriately. + * + * Currently, only one beep device is allowed to each codec. + */ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { struct hda_beep *beep; @@ -228,6 +247,10 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) } EXPORT_SYMBOL_GPL(snd_hda_attach_beep_device); +/** + * snd_hda_detach_beep_device - Detach the beep device + * @codec: the HDA codec + */ void snd_hda_detach_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; @@ -240,6 +263,10 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_detach_beep_device); +/** + * snd_hda_register_beep_device - Register the beep device + * @codec: the HDA codec + */ int snd_hda_register_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; @@ -269,6 +296,12 @@ static bool ctl_has_mute(struct snd_kcontrol *kcontrol) } /* get/put callbacks for beep mute mixer switches */ + +/** + * snd_hda_mixer_amp_switch_get_beep - Get callback for beep controls + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data + */ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -283,6 +316,11 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_get_beep); +/** + * snd_hda_mixer_amp_switch_put_beep - Put callback for beep controls + * @kcontrol: ctl element + * @ucontrol: pointer to get/store the data + */ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e152542..ca98f52 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -77,6 +77,10 @@ static struct hda_vendor_id hda_vendor_ids[] = { static DEFINE_MUTEX(preset_mutex); static LIST_HEAD(hda_preset_tables); +/** + * snd_hda_add_codec_preset - Add a codec preset to the chain + * @preset: codec preset table to add + */ int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) { mutex_lock(&preset_mutex); @@ -86,6 +90,10 @@ int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) } EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset); +/** + * snd_hda_delete_codec_preset - Delete a codec preset from the chain + * @preset: codec preset table to delete + */ int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) { mutex_lock(&preset_mutex); @@ -1187,7 +1195,16 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_GPL(snd_hda_codec_get_pincfg); -/* remember the current pinctl target value */ +/** + * snd_hda_codec_set_pin_target - remember the current pinctl target value + * @codec: the HDA codec + * @nid: pin NID + * @val: assigned pinctl value + * + * This function stores the given value to a pinctl target value in the + * pincfg table. This isn't always as same as the actually written value + * but can be referred at any time via snd_hda_codec_get_pin_target(). + */ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid, unsigned int val) { @@ -1201,7 +1218,11 @@ int snd_hda_codec_set_pin_target(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_codec_set_pin_target); -/* return the current pinctl target value */ +/** + * snd_hda_codec_get_pin_target - return the current pinctl target value + * @codec: the HDA codec + * @nid: pin NID + */ int snd_hda_codec_get_pin_target(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; @@ -1573,6 +1594,13 @@ int snd_hda_codec_new(struct hda_bus *bus, } EXPORT_SYMBOL_GPL(snd_hda_codec_new); +/** + * snd_hda_codec_update_widgets - Refresh widget caps and pin defaults + * @codec: the HDA codec + * + * Forcibly refresh the all widget caps and the init pin configurations of + * the given codec. + */ int snd_hda_codec_update_widgets(struct hda_codec *codec) { hda_nid_t fg; @@ -2239,7 +2267,17 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); -/* Works like snd_hda_codec_amp_update() but it writes the value only at +/** + * snd_hda_codec_amp_init - initialize the AMP value + * @codec: the HDA codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Works like snd_hda_codec_amp_update() but it writes the value only at * the first access. If the amp was already initialized / updated beforehand, * this does nothing. */ @@ -2250,6 +2288,17 @@ int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); +/** + * snd_hda_codec_amp_init_stereo - initialize the stereo AMP value + * @codec: the HDA codec + * @nid: NID to read the AMP value + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Call snd_hda_codec_amp_init() for both stereo channels. + */ int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val) { @@ -2644,7 +2693,10 @@ void snd_hda_ctls_clear(struct hda_codec *codec) snd_array_free(&codec->nids); } -/* pseudo device locking +/** + * snd_hda_lock_devices - pseudo device locking + * @bus: the BUS + * * toggle card->shutdown to allow/disallow the device access (as a hack) */ int snd_hda_lock_devices(struct hda_bus *bus) @@ -2681,6 +2733,10 @@ int snd_hda_lock_devices(struct hda_bus *bus) } EXPORT_SYMBOL_GPL(snd_hda_lock_devices); +/** + * snd_hda_unlock_devices - pseudo device unlocking + * @bus: the BUS + */ void snd_hda_unlock_devices(struct hda_bus *bus) { struct snd_card *card = bus->card; @@ -2867,7 +2923,7 @@ static int add_slave(struct hda_codec *codec, } /** - * snd_hda_add_vmaster - create a virtual master control and add slaves + * __snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec * @name: vmaster control name * @tlv: TLV data (optional) @@ -2970,10 +3026,15 @@ static struct snd_kcontrol_new vmaster_mute_mode = { .put = vmaster_mute_mode_put, }; -/* - * Add a mute-LED hook with the given vmaster switch kctl - * "Mute-LED Mode" control is automatically created and associated with - * the given hook. +/** + * snd_hda_add_vmaster_hook - Add a vmaster hook for mute-LED + * @codec: the HDA codec + * @hook: the vmaster hook object + * @expose_enum_ctl: flag to create an enum ctl + * + * Add a mute-LED hook with the given vmaster switch kctl. + * When @expose_enum_ctl is set, "Mute-LED Mode" control is automatically + * created and associated with the given hook. */ int snd_hda_add_vmaster_hook(struct hda_codec *codec, struct hda_vmaster_mute_hook *hook, @@ -2995,9 +3056,12 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_vmaster_hook); -/* - * Call the hook with the current value for synchronization - * Should be called in init callback +/** + * snd_hda_sync_vmaster_hook - Sync vmaster hook + * @hook: the vmaster hook + * + * Call the hook with the current value for synchronization. + * Should be called in init callback. */ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) { @@ -3599,7 +3663,11 @@ int snd_hda_create_dig_out_ctls(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls); -/* get the hda_spdif_out entry from the given NID +/** + * snd_hda_spdif_out_of_nid - get the hda_spdif_out entry from the given NID + * @codec: the HDA codec + * @nid: widget NID + * * call within spdif_mutex lock */ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, @@ -3616,6 +3684,13 @@ struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_spdif_out_of_nid); +/** + * snd_hda_spdif_ctls_unassign - Unassign the given SPDIF ctl + * @codec: the HDA codec + * @idx: the SPDIF ctl index + * + * Unassign the widget from the given SPDIF control. + */ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { struct hda_spdif_out *spdif; @@ -3627,6 +3702,14 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) } EXPORT_SYMBOL_GPL(snd_hda_spdif_ctls_unassign); +/** + * snd_hda_spdif_ctls_assign - Assign the SPDIF controls to the given NID + * @codec: the HDA codec + * @idx: the SPDIF ctl idx + * @nid: widget NID + * + * Assign the widget to the SPDIF control with the given index. + */ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) { struct hda_spdif_out *spdif; @@ -3946,6 +4029,16 @@ void snd_hda_codec_flush_cache(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_codec_flush_cache); +/** + * snd_hda_codec_set_power_to_all - Set the power state to all widgets + * @codec: the HDA codec + * @fg: function group (not used now) + * @power_state: the power state to set (AC_PWRST_*) + * + * Set the given power state to all widgets that have the power control. + * If the codec has power_filter set, it evaluates the power state and + * filter out if it's unchanged as D3. + */ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -4010,7 +4103,15 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec, return state; } -/* don't power down the widget if it controls eapd and EAPD_BTLENABLE is set */ +/** + * snd_hda_codec_eapd_power_filter - A power filter callback for EAPD + * @codec: the HDA codec + * @nid: widget NID + * @power_state: power state to evalue + * + * Don't power down the widget if it controls eapd and EAPD_BTLENABLE is set. + * This can be used a codec power_filter callback. + */ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) @@ -4671,6 +4772,17 @@ static int set_pcm_default_values(struct hda_codec *codec, /* * codec prepare/cleanup entries */ +/** + * snd_hda_codec_prepare - Prepare a stream + * @codec: the HDA codec + * @hinfo: PCM information + * @stream: stream tag to assign + * @format: format id to assign + * @substream: PCM substream to assign + * + * Calls the prepare callback set by the codec with the given arguments. + * Clean up the inactive streams when successful. + */ int snd_hda_codec_prepare(struct hda_codec *codec, struct hda_pcm_stream *hinfo, unsigned int stream, @@ -4687,6 +4799,14 @@ int snd_hda_codec_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_codec_prepare); +/** + * snd_hda_codec_cleanup - Prepare a stream + * @codec: the HDA codec + * @hinfo: PCM information + * @substream: PCM substream + * + * Calls the cleanup callback set by the codec with the given arguments. + */ void snd_hda_codec_cleanup(struct hda_codec *codec, struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b2d81ab..e664307 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -20,6 +20,16 @@ #include "hda_auto_parser.h" #include "hda_jack.h" +/** + * is_jack_detectable - Check whether the given pin is jack-detectable + * @codec: the HDA codec + * @nid: pin NID + * + * Check whether the given pin is capable to report the jack detection. + * The jack detection might not work by various reasons, e.g. the jack + * detection is prohibited in the codec level, the pin config has + * AC_DEFCFG_MISC_NO_PRESENCE bit, no unsol support, etc. + */ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { if (codec->no_jack_detect) @@ -268,6 +278,14 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback); +/** + * snd_hda_jack_detect_enable - Enable the jack detection on the given pin + * @codec: the HDA codec + * @nid: pin NID to enable jack detection + * + * Enable the jack detection with the default callback. Returns zero if + * successful or a negative error code. + */ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid) { return PTR_ERR_OR_ZERO(snd_hda_jack_detect_enable_callback(codec, nid, NULL)); @@ -410,6 +428,15 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; } +/** + * snd_hda_jack_add_kctl - Add a jack kctl for the given pin + * @codec: the HDA codec + * @nid: pin NID + * @name: the name string for the jack ctl + * @idx: the ctl index for the jack ctl + * + * This is a simple helper calling __snd_hda_jack_add_kctl(). + */ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, const char *name, int idx) { @@ -552,6 +579,11 @@ static void call_jack_callback(struct hda_codec *codec, } } +/** + * snd_hda_jack_unsol_event - Handle an unsolicited event + * @codec: the HDA codec + * @res: the unsolicited event data + */ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) { struct hda_jack_tbl *event; @@ -567,6 +599,13 @@ void snd_hda_jack_unsol_event(struct hda_codec *codec, unsigned int res) } EXPORT_SYMBOL_GPL(snd_hda_jack_unsol_event); +/** + * snd_hda_jack_poll_all - Poll all jacks + * @codec: the HDA codec + * + * Poll all detectable jacks with dirty flag, update the status, call + * callbacks and call snd_hda_jack_report_sync() if any changes are found. + */ void snd_hda_jack_poll_all(struct hda_codec *codec) { struct hda_jack_tbl *jack = codec->jacktbl.list; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 13cb375..b279e32 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -72,6 +72,11 @@ enum { int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); +/** + * snd_hda_jack_detect - Detect the jack + * @codec: the HDA codec + * @nid: pin NID to check jack detection + */ static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9b49f15..bef7215 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -417,8 +417,13 @@ static DEVICE_ATTR_RW(user_pin_configs); static DEVICE_ATTR_WO(reconfig); static DEVICE_ATTR_WO(clear); -/* - * Look for hint string +/** + * snd_hda_get_hint - Look for hint string + * @codec: the HDA codec + * @key: the hint key string + * + * Look for a hint key/value pair matching with the given key string + * and returns the value string. If nothing found, returns NULL. */ const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) { @@ -427,6 +432,15 @@ const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) } EXPORT_SYMBOL_GPL(snd_hda_get_hint); +/** + * snd_hda_get_bool_hint - Get a boolean hint value + * @codec: the HDA codec + * @key: the hint key string + * + * Look for a hint key/value pair matching with the given key string + * and returns a boolean value parsed from the value. If no matching + * key is found, return a negative value. + */ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) { const char *p; @@ -453,6 +467,16 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) } EXPORT_SYMBOL_GPL(snd_hda_get_bool_hint); +/** + * snd_hda_get_bool_hint - Get a boolean hint value + * @codec: the HDA codec + * @key: the hint key string + * @valp: pointer to store a value + * + * Look for a hint key/value pair matching with the given key string + * and stores the integer value to @valp. If no matching key is found, + * return a negative error code. Otherwise it returns zero. + */ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) { const char *p; @@ -690,8 +714,11 @@ static int get_line_from_fw(char *buf, int size, size_t *fw_size_p, return 1; } -/* - * load a "patch" firmware file and parse it +/** + * snd_hda_load_patch - load a "patch" firmware file and parse it + * @bus: HD-audio bus + * @fw_size: the firmware byte size + * @fw_buf: the firmware data */ int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf) { -- cgit v1.1 From df57de172a47f16548ee4bb69d1110e32686d6a9 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Wed, 29 Oct 2014 20:09:45 +0530 Subject: ALSA: hdspm: remove unused variable removed the unused variables. These variables were only being assigned some value, but the values were never being used. it has been build tested after removing the variables. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 23 +++-------------------- 1 file changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 7f7277b..e09348c1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1257,14 +1257,13 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) /* check for external sample rate, returns the sample rate in Hz*/ static int hdspm_external_sample_rate(struct hdspm *hdspm) { - unsigned int status, status2, timecode; + unsigned int status, status2; int syncref, rate = 0, rate_bits; switch (hdspm->io_type) { case AES32: status2 = hdspm_read(hdspm, HDSPM_statusRegister2); status = hdspm_read(hdspm, HDSPM_statusRegister); - timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); syncref = hdspm_autosync_ref(hdspm); switch (syncref) { @@ -4862,18 +4861,15 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status, status2, control, freq; + unsigned int status, status2; char *pref_sync_ref; char *autosync_ref; char *system_clock_mode; - char *insel; int x, x2; status = hdspm_read(hdspm, HDSPM_statusRegister); status2 = hdspm_read(hdspm, HDSPM_statusRegister2); - control = hdspm->control_register; - freq = hdspm_read(hdspm, HDSPM_timecodeRegister); snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", hdspm->card_name, hdspm->card->number + 1, @@ -4936,17 +4932,6 @@ snd_hdspm_proc_read_madi(struct snd_info_entry *entry, snd_iprintf(buffer, "Line out: %s\n", (hdspm->control_register & HDSPM_LineOut) ? "on " : "off"); - switch (hdspm->control_register & HDSPM_InputMask) { - case HDSPM_InputOptical: - insel = "Optical"; - break; - case HDSPM_InputCoaxial: - insel = "Coaxial"; - break; - default: - insel = "Unknown"; - } - snd_iprintf(buffer, "ClearTrackMarker = %s, Transmit in %s Channel Mode, " "Auto Input %s\n", @@ -5191,15 +5176,13 @@ snd_hdspm_proc_read_raydat(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status1, status2, status3, control, i; + unsigned int status1, status2, status3, i; unsigned int lock, sync; status1 = hdspm_read(hdspm, HDSPM_RD_STATUS_1); /* s1 */ status2 = hdspm_read(hdspm, HDSPM_RD_STATUS_2); /* freq */ status3 = hdspm_read(hdspm, HDSPM_RD_STATUS_3); /* s2 */ - control = hdspm->control_register; - snd_iprintf(buffer, "STATUS1: 0x%08x\n", status1); snd_iprintf(buffer, "STATUS2: 0x%08x\n", status2); snd_iprintf(buffer, "STATUS3: 0x%08x\n", status3); -- cgit v1.1 From 137f6d541ae75b3769c4c67e61c25340789b3cbc Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Wed, 29 Oct 2014 15:40:27 +0000 Subject: ASoC: Intel: dw_pdata can be static sound/soc/intel/sst-firmware.c:172:29: sparse: symbol 'dw_pdata' was not declared. Should it be static? Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 692a6ae..35788ad 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -169,7 +169,7 @@ err: return ret; } -struct dw_dma_platform_data dw_pdata = { +static struct dw_dma_platform_data dw_pdata = { .is_private = 1, .chan_allocation_order = CHAN_ALLOCATION_ASCENDING, .chan_priority = CHAN_PRIORITY_ASCENDING, -- cgit v1.1 From d96c53a193dd65380452c8e9f6dcf15cf829c7dc Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 29 Oct 2014 15:40:28 +0000 Subject: ASoC: Intel: Add generic support for DSP wake, sleep and stall Add generic functions to support DSP sleep, wake and stall. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp-priv.h | 3 +++ sound/soc/intel/sst-dsp.c | 23 +++++++++++++++++++++++ sound/soc/intel/sst-dsp.h | 4 ++++ 3 files changed, 30 insertions(+) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index be81b86..b9da030 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -36,6 +36,9 @@ struct sst_ops { /* DSP core boot / reset */ void (*boot)(struct sst_dsp *); void (*reset)(struct sst_dsp *); + int (*wake)(struct sst_dsp *); + void (*sleep)(struct sst_dsp *); + void (*stall)(struct sst_dsp *); /* Shim IO */ void (*write)(void __iomem *addr, u32 offset, u32 value); diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index d0fc685..86e4108 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -245,6 +245,29 @@ int sst_dsp_boot(struct sst_dsp *sst) } EXPORT_SYMBOL_GPL(sst_dsp_boot); +int sst_dsp_wake(struct sst_dsp *sst) +{ + if (sst->ops->wake) + return sst->ops->wake(sst); + + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_wake); + +void sst_dsp_sleep(struct sst_dsp *sst) +{ + if (sst->ops->sleep) + sst->ops->sleep(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_sleep); + +void sst_dsp_stall(struct sst_dsp *sst) +{ + if (sst->ops->stall) + sst->ops->stall(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_stall); + void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg) { sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 17ee923..7bb4820 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -245,6 +245,10 @@ void sst_memcpy_fromio_32(struct sst_dsp *sst, /* DSP reset & boot */ void sst_dsp_reset(struct sst_dsp *sst); int sst_dsp_boot(struct sst_dsp *sst); +int sst_dsp_wake(struct sst_dsp *sst); +void sst_dsp_sleep(struct sst_dsp *sst); +void sst_dsp_stall(struct sst_dsp *sst); + /* DMA */ int sst_dsp_dma_get_channel(struct sst_dsp *dsp, int chan_id); void sst_dsp_dma_put_channel(struct sst_dsp *dsp); -- cgit v1.1 From 6b7b4b8941b727af5fdc73b6a0910ede4c06cf11 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 29 Oct 2014 17:40:41 +0000 Subject: ASoC: Intel: Add PM support to the HSW/BDW DSP core. Add support for PM wake, sleep and stall calls to the core HSW/BDW driver. This includes reworking the reset and boot code and adding new calls for setting D3/D0 state. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-dsp.c | 158 ++++++++++++++++++++++++++++---------- 1 file changed, 119 insertions(+), 39 deletions(-) diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 5058dc8..86aea34 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -247,8 +247,67 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -static void hsw_boot(struct sst_dsp *sst) +static void hsw_set_dsp_D3(struct sst_dsp *sst) +{ + u32 val; + + /* switch off audio PLL, DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* Set D3 state */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +static int hsw_set_dsp_D0(struct sst_dsp *sst) { + int tries = 10; + u32 reg; + + /* Set D0 state */ + reg = readl(sst->addr.pci_cfg + SST_PMCS); + reg &= ~SST_PMCS_PS_MASK; + writel(reg, sst->addr.pci_cfg + SST_PMCS); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; + +finish: + hsw_reset(sst); + + /* switch on audio PLL, DRAM & IRAM blocks */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -267,30 +326,73 @@ static void hsw_boot(struct sst_dsp *sst) sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); - /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), - SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); + /* set on-demond mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH); + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits(sst, SST_IMRX, (SST_IMRX_BUSY | SST_IMRX_DONE), + 0x0); + sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | + SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + return 0; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* set oportunistic mode on engine 0,1 for all channels */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, 0); + /* set DSP to RUN */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); } -static void hsw_reset(struct sst_dsp *sst) +static void hsw_stall(struct sst_dsp *sst) { + /* stall DSP */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL, + SST_CSR_STALL | SST_CSR_24MHZ_LPCS); +} + +static void hsw_sleep(struct sst_dsp *sst) +{ + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); - /* keep in reset for 10ms */ - mdelay(10); + hsw_set_dsp_D3(sst); + dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); +} - /* take DSP out of reset and keep stalled for FW loading */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +static int hsw_wake(struct sst_dsp *sst) +{ + int ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume\n"); + + ret = hsw_set_dsp_D0(sst); + if (ret < 0) + return ret; + + dev_dbg(sst->dev, "HSW_PM dsp runtime resume exit\n"); + + return 0; } struct sst_adsp_memregion { @@ -431,27 +533,6 @@ static struct sst_block_ops sst_hsw_ops = { .disable = hsw_block_disable, }; -static int hsw_enable_shim(struct sst_dsp *sst) -{ - int tries = 10; - u32 reg; - - /* enable shim */ - reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); - writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); - - /* check that ADSP shim is enabled */ - while (tries--) { - reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); - if (reg != 0xffffffff) - return 0; - - msleep(1); - } - - return -ENODEV; -} - static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) { const struct sst_adsp_memregion *region; @@ -490,7 +571,7 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } /* enable the DSP SHIM */ - ret = hsw_enable_shim(sst); + ret = hsw_set_dsp_D0(sst); if (ret < 0) { dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); return ret; @@ -500,10 +581,6 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) if (ret) return ret; - /* Enable Interrupt from both sides */ - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); - sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, - (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); /* register DSP memory blocks - ideally we should get this from ACPI */ for (i = 0; i < region_count; i++) { @@ -535,6 +612,9 @@ static void hsw_free(struct sst_dsp *sst) struct sst_ops haswell_ops = { .reset = hsw_reset, .boot = hsw_boot, + .stall = hsw_stall, + .wake = hsw_wake, + .sleep = hsw_sleep, .write = sst_shim32_write, .read = sst_shim32_read, .write64 = sst_shim32_write64, -- cgit v1.1 From aed3c7b77c85ed7060f1f56bfa909d2a86ab2a20 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 29 Oct 2014 17:40:42 +0000 Subject: ASoC: Intel: Add PM support to HSW/BDW IPC driver Add PM and RTD3 support to the HSW/BDW IPC driver. This patch saves and restores the DSP context, loads and unloads FW and drops any pending IPC messages after suspend. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 256 +++++++++++++++++++++++++++++++++++++- sound/soc/intel/sst-haswell-ipc.h | 7 ++ 2 files changed, 258 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 770d467..b37d3ee 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "sst-haswell-ipc.h" #include "sst-dsp.h" @@ -276,6 +277,7 @@ struct sst_hsw { struct sst_hsw_ipc_fw_version version; struct sst_module *scratch; bool fw_done; + struct sst_fw *sst_fw; /* stream */ struct list_head stream_list; @@ -289,6 +291,8 @@ struct sst_hsw { /* DX */ struct sst_hsw_ipc_dx_reply dx; + void *dx_context; + dma_addr_t dx_context_paddr; /* boot */ wait_queue_head_t boot_wait; @@ -1707,6 +1711,237 @@ void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime) sst_module_runtime_free(runtime); } +#ifdef CONFIG_PM_RUNTIME +static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) +{ + struct sst_dsp *sst = hsw->dsp; + u32 item, offset, size; + int ret = 0; + + trace_ipc_request("PM state dump. Items #", SST_HSW_MAX_DX_REGIONS); + + if (hsw->dx.entries_no > SST_HSW_MAX_DX_REGIONS) { + dev_err(hsw->dev, + "error: number of FW context regions greater than %d\n", + SST_HSW_MAX_DX_REGIONS); + memset(&hsw->dx, 0, sizeof(hsw->dx)); + return -EINVAL; + } + + ret = sst_dsp_dma_get_channel(sst, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + return ret; + } + + /* set on-demond mode on engine 0 channel 3 */ + sst_dsp_shim_update_bits(sst, SST_HMDC, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH, + SST_HMDC_HDDA_E0_ALLCH | SST_HMDC_HDDA_E1_ALLCH); + + for (item = 0; item < hsw->dx.entries_no; item++) { + if (hsw->dx.mem_info[item].source == SST_HSW_DX_TYPE_MEMORY_DUMP + && hsw->dx.mem_info[item].offset > DSP_DRAM_ADDR_OFFSET + && hsw->dx.mem_info[item].offset < + DSP_DRAM_ADDR_OFFSET + SST_HSW_DX_CONTEXT_SIZE) { + + offset = hsw->dx.mem_info[item].offset + - DSP_DRAM_ADDR_OFFSET; + size = (hsw->dx.mem_info[item].size + 3) & (~3); + + ret = sst_dsp_dma_copyfrom(sst, hsw->dx_context_paddr + offset, + sst->addr.lpe_base + offset, size); + if (ret < 0) { + dev_err(hsw->dev, + "error: FW context dump failed\n"); + memset(&hsw->dx, 0, sizeof(hsw->dx)); + goto out; + } + } + } + +out: + sst_dsp_dma_put_channel(sst); + return ret; +} + +static int sst_hsw_dx_state_restore(struct sst_hsw *hsw) +{ + struct sst_dsp *sst = hsw->dsp; + u32 item, offset, size; + int ret; + + for (item = 0; item < hsw->dx.entries_no; item++) { + if (hsw->dx.mem_info[item].source == SST_HSW_DX_TYPE_MEMORY_DUMP + && hsw->dx.mem_info[item].offset > DSP_DRAM_ADDR_OFFSET + && hsw->dx.mem_info[item].offset < + DSP_DRAM_ADDR_OFFSET + SST_HSW_DX_CONTEXT_SIZE) { + + offset = hsw->dx.mem_info[item].offset + - DSP_DRAM_ADDR_OFFSET; + size = (hsw->dx.mem_info[item].size + 3) & (~3); + + ret = sst_dsp_dma_copyto(sst, sst->addr.lpe_base + offset, + hsw->dx_context_paddr + offset, size); + if (ret < 0) { + dev_err(hsw->dev, + "error: FW context restore failed\n"); + return ret; + } + } + } + + return 0; +} + +static void sst_hsw_drop_all(struct sst_hsw *hsw) +{ + struct ipc_message *msg, *tmp; + unsigned long flags; + int tx_drop_cnt = 0, rx_drop_cnt = 0; + + /* drop all TX and Rx messages before we stall + reset DSP */ + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + + list_for_each_entry_safe(msg, tmp, &hsw->tx_list, list) { + list_move(&msg->list, &hsw->empty_list); + tx_drop_cnt++; + } + + list_for_each_entry_safe(msg, tmp, &hsw->rx_list, list) { + list_move(&msg->list, &hsw->empty_list); + rx_drop_cnt++; + } + + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + + if (tx_drop_cnt || rx_drop_cnt) + dev_err(hsw->dev, "dropped IPC msg RX=%d, TX=%d\n", + tx_drop_cnt, rx_drop_cnt); +} + +int sst_hsw_dsp_load(struct sst_hsw *hsw) +{ + struct sst_dsp *dsp = hsw->dsp; + int ret; + + dev_dbg(hsw->dev, "loading audio DSP...."); + + ret = sst_dsp_wake(dsp); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to wake audio DSP\n"); + return -ENODEV; + } + + ret = sst_dsp_dma_get_channel(dsp, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + return ret; + } + + ret = sst_fw_reload(hsw->sst_fw); + if (ret < 0) { + dev_err(hsw->dev, "error: SST FW reload failed\n"); + sst_dsp_dma_put_channel(dsp); + return -ENOMEM; + } + + sst_dsp_dma_put_channel(dsp); + return 0; +} + +static int sst_hsw_dsp_restore(struct sst_hsw *hsw) +{ + struct sst_dsp *dsp = hsw->dsp; + int ret; + + dev_dbg(hsw->dev, "restoring audio DSP...."); + + ret = sst_dsp_dma_get_channel(dsp, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: cant allocate dma channel %d\n", ret); + return ret; + } + + ret = sst_hsw_dx_state_restore(hsw); + if (ret < 0) { + dev_err(hsw->dev, "error: SST FW context restore failed\n"); + sst_dsp_dma_put_channel(dsp); + return -ENOMEM; + } + sst_dsp_dma_put_channel(dsp); + + /* wait for DSP boot completion */ + sst_dsp_boot(dsp); + + return ret; +} + +int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw) +{ + int ret; + + dev_dbg(hsw->dev, "audio dsp runtime suspend\n"); + + ret = sst_hsw_dx_set_state(hsw, SST_HSW_DX_STATE_D3, &hsw->dx); + if (ret < 0) + return ret; + + sst_dsp_stall(hsw->dsp); + + ret = sst_hsw_dx_state_dump(hsw); + if (ret < 0) + return ret; + + sst_hsw_drop_all(hsw); + + return 0; +} + +int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw) +{ + sst_fw_unload(hsw->sst_fw); + sst_block_free_scratch(hsw->dsp); + + hsw->boot_complete = false; + + sst_dsp_sleep(hsw->dsp); + + return 0; +} + +int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) +{ + struct device *dev = hsw->dev; + int ret; + + dev_dbg(dev, "audio dsp runtime resume\n"); + + if (hsw->boot_complete) + return 1; /* tell caller no action is required */ + + ret = sst_hsw_dsp_restore(hsw); + if (ret < 0) + dev_err(dev, "error: audio DSP boot failure\n"); + + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, + msecs_to_jiffies(IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(hsw->dev, "error: audio DSP boot timeout\n"); + return -EIO; + } + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) + dev_err(dev, "error: SSP re-initialization failed\n"); + + return ret; +} +#endif + static int msg_empty_list_init(struct sst_hsw *hsw) { int i; @@ -1738,7 +1973,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_hsw_ipc_fw_version version; struct sst_hsw *hsw; - struct sst_fw *hsw_sst_fw; int ret; dev_dbg(dev, "initialising Audio DSP IPC\n"); @@ -1780,12 +2014,19 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) goto dsp_err; } + /* allocate DMA buffer for context storage */ + hsw->dx_context = dma_alloc_coherent(hsw->dsp->dma_dev, + SST_HSW_DX_CONTEXT_SIZE, &hsw->dx_context_paddr, GFP_KERNEL); + if (hsw->dx_context == NULL) { + ret = -ENOMEM; + goto dma_err; + } + /* keep the DSP in reset state for base FW loading */ sst_dsp_reset(hsw->dsp); - hsw_sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); - - if (hsw_sst_fw == NULL) { + hsw->sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); + if (hsw->sst_fw == NULL) { ret = -ENODEV; dev_err(dev, "error: failed to load firmware\n"); goto fw_err; @@ -1816,8 +2057,11 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) boot_err: sst_dsp_reset(hsw->dsp); - sst_fw_free(hsw_sst_fw); + sst_fw_free(hsw->sst_fw); fw_err: + dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, + hsw->dx_context, hsw->dx_context_paddr); +dma_err: sst_dsp_free(hsw->dsp); dsp_err: kthread_stop(hsw->tx_thread); @@ -1834,6 +2078,8 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) sst_dsp_reset(hsw->dsp); sst_fw_free_all(hsw->dsp); + dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, + hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); kfree(hsw->scratch); kthread_stop(hsw->tx_thread); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index fe6e63f..afd0ae1 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -23,6 +23,7 @@ #define SST_HSW_NO_CHANNELS 2 #define SST_HSW_MAX_DX_REGIONS 14 +#define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) #define SST_HSW_FW_LOG_CONFIG_DWORDS 12 #define SST_HSW_GLOBAL_LOG 15 @@ -492,4 +493,10 @@ struct sst_module_runtime *sst_hsw_runtime_module_create(struct sst_hsw *hsw, int mod_id, int offset); void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime); +/* PM */ +int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw); +int sst_hsw_dsp_runtime_suspend(struct sst_hsw *hsw); +int sst_hsw_dsp_load(struct sst_hsw *hsw); +int sst_hsw_dsp_runtime_sleep(struct sst_hsw *hsw); + #endif -- cgit v1.1 From 2e4f75919e5a02d605b0d84425251654d48fb056 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 29 Oct 2014 17:40:43 +0000 Subject: ASoC: Intel: Add PM support to HSW/BDW PCM driver Add PM and RTD3 support to the HSW/BDW PCM driver. The PCM driver will now save DSP context and then power off the DSP when it's not in use. DSP power and context is then restored when it's next used. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 3 + sound/soc/intel/sst-haswell-pcm.c | 271 ++++++++++++++++++++++++++++++++++++-- 2 files changed, 260 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 7bb4820..2753b85 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -30,6 +30,9 @@ #define SST_DMA_TYPE_DW 1 #define SST_DMA_TYPE_MID 2 +/* autosuspend delay 5s*/ +#define SST_RUNTIME_SUSPEND_DELAY (5 * 1000) + /* SST Shim register map * The register naming can differ between products. Some products also * contain extra functionality. diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 522edef..4489a35 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -73,6 +74,13 @@ static const u32 volume_map[] = { #define HSW_PCM_PERIODS_MAX 64 #define HSW_PCM_PERIODS_MIN 2 +#define HSW_PCM_DAI_ID_SYSTEM 0 +#define HSW_PCM_DAI_ID_OFFLOAD0 1 +#define HSW_PCM_DAI_ID_OFFLOAD1 2 +#define HSW_PCM_DAI_ID_LOOPBACK 3 +#define HSW_PCM_DAI_ID_CAPTURE 4 + + static const struct snd_pcm_hardware hsw_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -99,6 +107,8 @@ struct hsw_pcm_data { int dai_id; struct sst_hsw_stream *stream; struct sst_module_runtime *runtime; + struct sst_module_runtime_context context; + struct snd_pcm *hsw_pcm; u32 volume[2]; struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; @@ -108,10 +118,18 @@ struct hsw_pcm_data { int persistent_offset; }; +enum hsw_pm_state { + HSW_PM_STATE_D3 = 0, + HSW_PM_STATE_D0 = 1, +}; + /* private data for the driver */ struct hsw_priv_data { /* runtime DSP */ struct sst_hsw *hsw; + struct device *dev; + enum hsw_pm_state pm_state; + struct snd_soc_card *soc_card; /* page tables */ struct snd_dma_buffer dmab[HSW_PCM_COUNT][2]; @@ -145,21 +163,25 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value) static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); - struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; mutex_lock(&pcm_data->mutex); + pm_runtime_get_sync(pdata->dev); if (!pcm_data->stream) { pcm_data->volume[0] = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); pcm_data->volume[1] = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return 0; } @@ -175,6 +197,8 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 1, volume); } + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return 0; } @@ -182,21 +206,25 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); - struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; mutex_lock(&pcm_data->mutex); + pm_runtime_get_sync(pdata->dev); if (!pcm_data->stream) { ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(pcm_data->volume[0]); ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(pcm_data->volume[1]); + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return 0; } @@ -205,6 +233,9 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 1, &volume); ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return 0; @@ -213,11 +244,13 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, static int hsw_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); - struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); struct sst_hsw *hsw = pdata->hsw; u32 volume; + pm_runtime_get_sync(pdata->dev); + if (ucontrol->value.integer.value[0] == ucontrol->value.integer.value[1]) { @@ -232,23 +265,28 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, sst_hsw_mixer_set_volume(hsw, 0, 1, volume); } + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); return 0; } static int hsw_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); - struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); + struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); struct sst_hsw *hsw = pdata->hsw; unsigned int volume = 0; + pm_runtime_get_sync(pdata->dev); sst_hsw_mixer_get_volume(hsw, 0, 0, &volume); ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); sst_hsw_mixer_get_volume(hsw, 0, 1, &volume); ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); return 0; } @@ -577,6 +615,7 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) pcm_data = &pdata->pcm[rtd->cpu_dai->id]; mutex_lock(&pcm_data->mutex); + pm_runtime_get_sync(pdata->dev); snd_soc_pcm_set_drvdata(rtd, pcm_data); pcm_data->substream = substream; @@ -587,6 +626,8 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) hsw_notify_pointer, pcm_data); if (pcm_data->stream == NULL) { dev_err(rtd->dev, "error: failed to create stream\n"); + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return -EINVAL; } @@ -626,6 +667,8 @@ static int hsw_pcm_close(struct snd_pcm_substream *substream) pcm_data->stream = NULL; out: + pm_runtime_mark_last_busy(pdata->dev); + pm_runtime_put_autosuspend(pdata->dev); mutex_unlock(&pcm_data->mutex); return ret; } @@ -643,11 +686,11 @@ static struct snd_pcm_ops hsw_pcm_ops = { /* static mappings between PCMs and modules - may be dynamic in future */ static struct hsw_pcm_module_map mod_map[] = { - {0, SST_HSW_MODULE_PCM_SYSTEM}, /* "System Pin" */ - {1, SST_HSW_MODULE_PCM}, /* "Offload0 Pin" */ - {2, SST_HSW_MODULE_PCM}, /* "Offload1 Pin" */ - {3, SST_HSW_MODULE_PCM_REFERENCE}, /* "Loopback Pin" */ - {4, SST_HSW_MODULE_PCM_CAPTURE}, /* "Capture Pin" */ + {HSW_PCM_DAI_ID_SYSTEM, SST_HSW_MODULE_PCM_SYSTEM}, + {HSW_PCM_DAI_ID_OFFLOAD0, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_OFFLOAD1, SST_HSW_MODULE_PCM}, + {HSW_PCM_DAI_ID_LOOPBACK, SST_HSW_MODULE_PCM_REFERENCE}, + {HSW_PCM_DAI_ID_CAPTURE, SST_HSW_MODULE_PCM_CAPTURE}, }; static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) @@ -659,6 +702,7 @@ static int hsw_pcm_create_modules(struct hsw_priv_data *pdata) for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[i]; + /* create new runtime module, use same offset if recreated */ pcm_data->runtime = sst_hsw_runtime_module_create(hsw, mod_map[i].mod_id, pcm_data->persistent_offset); if (pcm_data->runtime == NULL) @@ -700,6 +744,7 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_pcm *pcm = rtd->pcm; struct snd_soc_platform *platform = rtd->platform; struct sst_pdata *pdata = dev_get_platdata(platform->dev); + struct hsw_priv_data *priv_data = dev_get_drvdata(platform->dev); struct device *dev = pdata->dma_dev; int ret = 0; @@ -716,6 +761,7 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) return ret; } } + priv_data->pcm[rtd->cpu_dai->id].hsw_pcm = pcm; return ret; } @@ -728,6 +774,7 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_driver hsw_dais[] = { { .name = "System Pin", + .id = HSW_PCM_DAI_ID_SYSTEM, .playback = { .stream_name = "System Playback", .channels_min = 2, @@ -739,6 +786,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { { /* PCM */ .name = "Offload0 Pin", + .id = HSW_PCM_DAI_ID_OFFLOAD0, .playback = { .stream_name = "Offload0 Playback", .channels_min = 2, @@ -750,6 +798,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { { /* PCM */ .name = "Offload1 Pin", + .id = HSW_PCM_DAI_ID_OFFLOAD1, .playback = { .stream_name = "Offload1 Playback", .channels_min = 2, @@ -760,6 +809,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { }, { .name = "Loopback Pin", + .id = HSW_PCM_DAI_ID_LOOPBACK, .capture = { .stream_name = "Loopback Capture", .channels_min = 2, @@ -770,6 +820,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { }, { .name = "Capture Pin", + .id = HSW_PCM_DAI_ID_CAPTURE, .capture = { .stream_name = "Analog Capture", .channels_min = 2, @@ -808,9 +859,21 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) { struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform); struct sst_pdata *pdata = dev_get_platdata(platform->dev); - struct device *dma_dev = pdata->dma_dev; + struct device *dma_dev, *dev; int i, ret = 0; + if (!pdata) + return -ENODEV; + + dev = platform->dev; + dma_dev = pdata->dma_dev; + + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); + priv_data->hsw = pdata->dsp; + priv_data->dev = platform->dev; + priv_data->pm_state = HSW_PM_STATE_D0; + priv_data->soc_card = platform->component.card; + /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { @@ -836,6 +899,13 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) /* allocate runtime modules */ hsw_pcm_create_modules(priv_data); + /* enable runtime PM with auto suspend */ + pm_runtime_set_autosuspend_delay(platform->dev, + SST_RUNTIME_SUSPEND_DELAY); + pm_runtime_use_autosuspend(platform->dev); + pm_runtime_enable(platform->dev); + pm_runtime_idle(platform->dev); + return 0; err: @@ -854,6 +924,9 @@ static int hsw_pcm_remove(struct snd_soc_platform *platform) snd_soc_platform_get_drvdata(platform); int i; + pm_runtime_disable(platform->dev); + hsw_pcm_free_modules(priv_data); + for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { if (hsw_dais[i].playback.channels_min) snd_dma_free_pages(&priv_data->dmab[i][0]); @@ -931,10 +1004,180 @@ static int hsw_pcm_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +#ifdef CONFIG_PM_RUNTIME + +static int hsw_pcm_runtime_idle(struct device *dev) +{ + return 0; +} + +static int hsw_pcm_runtime_suspend(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + + if (pdata->pm_state == HSW_PM_STATE_D3) + return 0; + + sst_hsw_dsp_runtime_suspend(hsw); + sst_hsw_dsp_runtime_sleep(hsw); + pdata->pm_state = HSW_PM_STATE_D3; + + return 0; +} + +static int hsw_pcm_runtime_resume(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + int ret; + + if (pdata->pm_state == HSW_PM_STATE_D0) + return 0; + + ret = sst_hsw_dsp_load(hsw); + if (ret < 0) { + dev_err(dev, "failed to reload %d\n", ret); + return ret; + } + + ret = hsw_pcm_create_modules(pdata); + if (ret < 0) { + dev_err(dev, "failed to create modules %d\n", ret); + return ret; + } + + ret = sst_hsw_dsp_runtime_resume(hsw); + if (ret < 0) + return ret; + else if (ret == 1) /* no action required */ + return 0; + + pdata->pm_state = HSW_PM_STATE_D0; + return ret; +} + +static void hsw_pcm_complete(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + struct hsw_pcm_data *pcm_data; + int i, err; + + if (pdata->pm_state == HSW_PM_STATE_D0) + return; + + err = sst_hsw_dsp_load(hsw); + if (err < 0) { + dev_err(dev, "failed to reload %d\n", err); + return; + } + + err = hsw_pcm_create_modules(pdata); + if (err < 0) { + dev_err(dev, "failed to create modules %d\n", err); + return; + } + + for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + pcm_data = &pdata->pcm[i]; + + if (!pcm_data->substream) + continue; + + err = sst_module_runtime_restore(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to restore context for PCM %d\n", i); + } + + snd_soc_resume(pdata->soc_card->dev); + + err = sst_hsw_dsp_runtime_resume(hsw); + if (err < 0) + return; + else if (err == 1) /* no action required */ + return; + + pdata->pm_state = HSW_PM_STATE_D0; + return; +} + +static int hsw_pcm_prepare(struct device *dev) +{ + struct hsw_priv_data *pdata = dev_get_drvdata(dev); + struct sst_hsw *hsw = pdata->hsw; + struct hsw_pcm_data *pcm_data; + int i, err; + + if (pdata->pm_state == HSW_PM_STATE_D3) + return 0; + /* suspend all active streams */ + for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + pcm_data = &pdata->pcm[i]; + + if (!pcm_data->substream) + continue; + dev_dbg(dev, "suspending pcm %d\n", i); + snd_pcm_suspend_all(pcm_data->hsw_pcm); + + /* We need to wait until the DSP FW stops the streams */ + msleep(2); + } + + snd_soc_suspend(pdata->soc_card->dev); + snd_soc_poweroff(pdata->soc_card->dev); + + /* enter D3 state and stall */ + sst_hsw_dsp_runtime_suspend(hsw); + + /* preserve persistent memory */ + for (i = 0; i < HSW_PCM_DAI_ID_CAPTURE + 1; i++) { + pcm_data = &pdata->pcm[i]; + + if (!pcm_data->substream) + continue; + + dev_dbg(dev, "saving context pcm %d\n", i); + err = sst_module_runtime_save(pcm_data->runtime, + &pcm_data->context); + if (err < 0) + dev_err(dev, "failed to save context for PCM %d\n", i); + } + + /* put the DSP to sleep */ + sst_hsw_dsp_runtime_sleep(hsw); + pdata->pm_state = HSW_PM_STATE_D3; + + return 0; +} + +#else +#define hsw_pcm_runtime_idle NULL +#define hsw_pcm_runtime_suspend NULL +#define hsw_pcm_runtime_resume NULL +#define hsw_pcm_runtime_complete NULL +#define hsw_pcm_runtime_prepare NULL +#endif + +static const struct dev_pm_ops hsw_pcm_pm = { + .runtime_idle = hsw_pcm_runtime_idle, + .runtime_suspend = hsw_pcm_runtime_suspend, + .runtime_resume = hsw_pcm_runtime_resume, + .prepare = hsw_pcm_prepare, + .complete = hsw_pcm_complete, +}; +#else +#define hsw_pcm_pm NULL +#endif + static struct platform_driver hsw_pcm_driver = { .driver = { .name = "haswell-pcm-audio", .owner = THIS_MODULE, + .pm = &hsw_pcm_pm, + }, .probe = hsw_pcm_dev_probe, -- cgit v1.1 From eafe8404c103b3051b6421fc17e0e8b91d369f0b Mon Sep 17 00:00:00 2001 From: Tina Ruchandani Date: Wed, 29 Oct 2014 10:48:10 -0700 Subject: ALSA: es1968: Replace timeval with ktime_t es1968_measure_clock uses struct timeval, which on 32-bit systems will overflow in 2038, leading to incorrect interpretation of time.This patch changes the function to use ktime_t instead of struct timeval, which implies: - no y2038: ktime_t uses a 64-bit datatype explicitly. - efficent subtraction: The earlier version computes the difference in usecs while dealing with secs and nsecs. It requires checks to see if the nsecs of stop is less than start. This patch uses a direct subtract of ktime_t and converts to usecs. - use of monotonic clock (ktime_get) over real time (do_gettimeofday), which simplifies timekeeping, as it does not have to deal with cases where stop_time is less than start_time. Signed-off-by: Tina Ruchandani Reviewed-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a9956a7..6039700 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1710,7 +1710,8 @@ static void es1968_measure_clock(struct es1968 *chip) int i, apu; unsigned int pa, offset, t; struct esm_memory *memory; - struct timeval start_time, stop_time; + ktime_t start_time, stop_time; + ktime_t diff; if (chip->clock == 0) chip->clock = 48000; /* default clock value */ @@ -1761,12 +1762,12 @@ static void es1968_measure_clock(struct es1968 *chip) snd_es1968_bob_inc(chip, ESM_BOB_FREQ); __apu_set_register(chip, apu, 5, pa & 0xffff); snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); - do_gettimeofday(&start_time); + start_time = ktime_get(); spin_unlock_irq(&chip->reg_lock); msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); - do_gettimeofday(&stop_time); + stop_time = ktime_get(); snd_es1968_trigger_apu(chip, apu, 0); /* stop */ snd_es1968_bob_dec(chip); chip->in_measurement = 0; @@ -1777,12 +1778,8 @@ static void es1968_measure_clock(struct es1968 *chip) offset &= 0xfffe; offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2); - t = stop_time.tv_sec - start_time.tv_sec; - t *= 1000000; - if (stop_time.tv_usec < start_time.tv_usec) - t -= start_time.tv_usec - stop_time.tv_usec; - else - t += stop_time.tv_usec - start_time.tv_usec; + diff = ktime_sub(stop_time, start_time); + t = ktime_to_us(diff); if (t == 0) { dev_err(chip->card->dev, "?? calculation error..\n"); } else { -- cgit v1.1 From dda42bd0c3a4b7be1561546914eda59b68a58be4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 12:04:50 +0100 Subject: ALSA: hda - Add kerneldoc comments to hda_generic.c Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 164 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 135 insertions(+), 29 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 06d7210..63b69f7 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -40,7 +40,12 @@ #include "hda_generic.h" -/* initialize hda_gen_spec struct */ +/** + * snd_hda_gen_spec_init - initialize hda_gen_spec struct + * @spec: hda_gen_spec object to initialize + * + * Initialize the given hda_gen_spec object. + */ int snd_hda_gen_spec_init(struct hda_gen_spec *spec) { snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32); @@ -51,6 +56,17 @@ int snd_hda_gen_spec_init(struct hda_gen_spec *spec) } EXPORT_SYMBOL_GPL(snd_hda_gen_spec_init); +/** + * snd_hda_gen_add_kctl - Add a new kctl_new struct from the template + * @spec: hda_gen_spec object + * @name: name string to override the template, NULL if unchanged + * @temp: template for the new kctl + * + * Add a new kctl (actually snd_kcontrol_new to be instantiated later) + * element based on the given snd_kcontrol_new template @temp and the + * name string @name to the list in @spec. + * Returns the newly created object or NULL as error. + */ struct snd_kcontrol_new * snd_hda_gen_add_kctl(struct hda_gen_spec *spec, const char *name, const struct snd_kcontrol_new *temp) @@ -259,8 +275,14 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, return NULL; } -/* get the path between the given NIDs; - * passing 0 to either @pin or @dac behaves as a wildcard +/** + * snd_hda_get_nid_path - get the path between the given NIDs + * @codec: the HDA codec + * @from_nid: the NID where the path start from + * @to_nid: the NID where the path ends at + * + * Return the found nid_path object or NULL for error. + * Passing 0 to either @from_nid or @to_nid behaves as a wildcard. */ struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, hda_nid_t from_nid, hda_nid_t to_nid) @@ -269,8 +291,14 @@ struct nid_path *snd_hda_get_nid_path(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_get_nid_path); -/* get the index number corresponding to the path instance; - * the index starts from 1, for easier checking the invalid value +/** + * snd_hda_get_path_idx - get the index number corresponding to the path + * instance + * @codec: the HDA codec + * @path: nid_path object + * + * The returned index starts from 1, i.e. the actual array index with offset 1, + * and zero is handled as an invalid path */ int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path) { @@ -287,7 +315,12 @@ int snd_hda_get_path_idx(struct hda_codec *codec, struct nid_path *path) } EXPORT_SYMBOL_GPL(snd_hda_get_path_idx); -/* get the path instance corresponding to the given index number */ +/** + * snd_hda_get_path_from_idx - get the path instance corresponding to the + * given index number + * @codec: the HDA codec + * @idx: the path index + */ struct nid_path *snd_hda_get_path_from_idx(struct hda_codec *codec, int idx) { struct hda_gen_spec *spec = codec->spec; @@ -415,7 +448,18 @@ static bool __parse_nid_path(struct hda_codec *codec, return true; } -/* parse the widget path from the given nid to the target nid; +/** + * snd_hda_parse_nid_path - parse the widget path from the given nid to + * the target nid + * @codec: the HDA codec + * @from_nid: the NID where the path start from + * @to_nid: the NID where the path ends at + * @anchor_nid: the anchor indication + * @path: the path object to store the result + * + * Returns true if a matching path is found. + * + * The parsing behavior depends on parameters: * when @from_nid is 0, try to find an empty DAC; * when @anchor_nid is set to a positive value, only paths through the widget * with the given value are evaluated. @@ -436,9 +480,15 @@ bool snd_hda_parse_nid_path(struct hda_codec *codec, hda_nid_t from_nid, } EXPORT_SYMBOL_GPL(snd_hda_parse_nid_path); -/* - * parse the path between the given NIDs and add to the path list. - * if no valid path is found, return NULL +/** + * snd_hda_add_new_path - parse the path between the given NIDs and + * add to the path list + * @codec: the HDA codec + * @from_nid: the NID where the path start from + * @to_nid: the NID where the path ends at + * @anchor_nid: the anchor indication, see snd_hda_parse_nid_path() + * + * If no valid path is found, returns NULL. */ struct nid_path * snd_hda_add_new_path(struct hda_codec *codec, hda_nid_t from_nid, @@ -724,8 +774,14 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* activate or deactivate the given path - * if @add_aamix is set, enable the input from aa-mix NID as well (if any) +/** + * snd_hda_activate_path - activate or deactivate the given path + * @codec: the HDA codec + * @path: the path to activate/deactivate + * @enable: flag to activate or not + * @add_aamix: enable the input from aamix NID + * + * If @add_aamix is set, enable the input from aa-mix NID as well (if any). */ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, bool enable, bool add_aamix) @@ -3883,7 +3939,12 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } } -/* Toggle outputs muting */ +/** + * snd_hda_gen_update_outputs - Toggle outputs muting + * @codec: the HDA codec + * + * Update the mute status of all outputs based on the current jack states. + */ void snd_hda_gen_update_outputs(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; @@ -3944,7 +4005,11 @@ static void call_update_outputs(struct hda_codec *codec) snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false); } -/* standard HP-automute helper */ +/** + * snd_hda_gen_hp_automute - standard HP-automute helper + * @codec: the HDA codec + * @jack: jack object, NULL for the whole + */ void snd_hda_gen_hp_automute(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -3965,7 +4030,11 @@ void snd_hda_gen_hp_automute(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_gen_hp_automute); -/* standard line-out-automute helper */ +/** + * snd_hda_gen_line_automute - standard line-out-automute helper + * @codec: the HDA codec + * @jack: jack object, NULL for the whole + */ void snd_hda_gen_line_automute(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -3986,7 +4055,11 @@ void snd_hda_gen_line_automute(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_gen_line_automute); -/* standard mic auto-switch helper */ +/** + * snd_hda_gen_mic_autoswitch - standard mic auto-switch helper + * @codec: the HDA codec + * @jack: jack object, NULL for the whole + */ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -4318,7 +4391,13 @@ static int check_auto_mic_availability(struct hda_codec *codec) return 0; } -/* power_filter hook; make inactive widgets into power down */ +/** + * snd_hda_gen_path_power_filter - power_filter hook to make inactive widgets + * into power down + * @codec: the HDA codec + * @nid: NID to evalute + * @power_state: target power state + */ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state) @@ -4354,8 +4433,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) } } -/* - * Parse the given BIOS configuration and set up the hda_gen_spec +/** + * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and + * set up the hda_gen_spec + * @codec: the HDA codec + * @cfg: Parsed pin configuration * * return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -4541,6 +4623,12 @@ static const char * const slave_pfxs[] = { NULL, }; +/** + * snd_hda_gen_build_controls - Build controls from the parsed results + * @codec: the HDA codec + * + * Pass this to build_controls patch_ops. + */ int snd_hda_gen_build_controls(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; @@ -5018,7 +5106,12 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, strlcat(str, sfx, len); } -/* build PCM streams based on the parsed results */ +/** + * snd_hda_gen_build_pcms - build PCM streams based on the parsed results + * @codec: the HDA codec + * + * Pass this to build_pcms patch_ops. + */ int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; @@ -5313,9 +5406,11 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec) } } -/* - * initialize the generic spec; - * this can be put as patch_ops.init function +/** + * snd_hda_gen_init - initialize the generic spec + * @codec: the HDA codec + * + * This can be put as patch_ops init function. */ int snd_hda_gen_init(struct hda_codec *codec) { @@ -5351,9 +5446,11 @@ int snd_hda_gen_init(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_gen_init); -/* - * free the generic spec; - * this can be put as patch_ops.free function +/** + * snd_hda_gen_free - free the generic spec + * @codec: the HDA codec + * + * This can be put as patch_ops free function. */ void snd_hda_gen_free(struct hda_codec *codec) { @@ -5365,9 +5462,12 @@ void snd_hda_gen_free(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_gen_free); #ifdef CONFIG_PM -/* - * check the loopback power save state; - * this can be put as patch_ops.check_power_status function +/** + * snd_hda_gen_check_power_status - check the loopback power save state + * @codec: the HDA codec + * @nid: NID to inspect + * + * This can be put as patch_ops check_power_status function. */ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid) { @@ -5393,6 +5493,12 @@ static const struct hda_codec_ops generic_patch_ops = { #endif }; +/** + * snd_hda_parse_generic_codec - Generic codec parser + * @codec: the HDA codec + * + * This should be called from the HDA codec core. + */ int snd_hda_parse_generic_codec(struct hda_codec *codec) { struct hda_gen_spec *spec; -- cgit v1.1 From 438f4e2801d730442d6fc36ab2b265c4b54cff8b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 12:44:00 +0100 Subject: ALSA: doc: Add ASoC codes into API documentation Signed-off-by: Takashi Iwai --- Documentation/DocBook/alsa-driver-api.tmpl | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 7bec270..b2d2c3b 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -104,13 +104,30 @@ !Iinclude/sound/compress_driver.h + ASoC + ASoC Core API +!Iinclude/sound/soc.h +!Esound/soc/soc-core.c +!Esound/soc/soc-cache.c +!Esound/soc/soc-devres.c +!Esound/soc/soc-io.c +!Esound/soc/soc-pcm.c + + ASoC DAPM API +!Esound/soc/soc-dapm.c + + ASoC DMA Engine API +!Esound/soc/soc-generic-dmaengine-pcm.c + + Miscellaneous Functions Hardware-Dependent Devices API !Esound/core/hwdep.c Jack Abstraction Layer API -!Esound/core/jack.c !Iinclude/sound/jack.h +!Esound/core/jack.c +!Esound/soc/soc-jack.c ISA DMA Helpers !Esound/core/isadma.c -- cgit v1.1 From 12cddbd8696657ff405e054be48747b906731698 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 13:44:34 +0100 Subject: ALSA: control: Add missing kerneldoc comments to exported functions A few functions have no proper documentation yet, so let's add them. Along with it, remove superfluous blank line between the closing brace and EXPORT_SYMBOL() line. Signed-off-by: Takashi Iwai --- sound/core/control.c | 64 ++++++++++++++++++++++++++++++++++++++++------------ 1 file changed, 49 insertions(+), 15 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 5c35bba..99aa3aa 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -141,6 +141,16 @@ static int snd_ctl_release(struct inode *inode, struct file *file) return 0; } +/** + * snd_ctl_notify - Send notification to user-space for a control change + * @card: the card to send notification + * @mask: the event mask, SNDRV_CTL_EVENT_* + * @id: the ctl element id to send notification + * + * This function adds an event record with the given id and mask, appends + * to the list and wakes up the user-space for notification. This can be + * called in the atomic context. + */ void snd_ctl_notify(struct snd_card *card, unsigned int mask, struct snd_ctl_elem_id *id) { @@ -179,7 +189,6 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, } read_unlock(&card->ctl_files_rwlock); } - EXPORT_SYMBOL(snd_ctl_notify); /** @@ -261,7 +270,6 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.private_data = private_data; return snd_ctl_new(&kctl, access); } - EXPORT_SYMBOL(snd_ctl_new1); /** @@ -280,7 +288,6 @@ void snd_ctl_free_one(struct snd_kcontrol *kcontrol) kfree(kcontrol); } } - EXPORT_SYMBOL(snd_ctl_free_one); static bool snd_ctl_remove_numid_conflict(struct snd_card *card, @@ -376,7 +383,6 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) snd_ctl_free_one(kcontrol); return err; } - EXPORT_SYMBOL(snd_ctl_add); /** @@ -471,7 +477,6 @@ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) snd_ctl_free_one(kcontrol); return 0; } - EXPORT_SYMBOL(snd_ctl_remove); /** @@ -499,7 +504,6 @@ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) up_write(&card->controls_rwsem); return ret; } - EXPORT_SYMBOL(snd_ctl_remove_id); /** @@ -617,7 +621,6 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, up_write(&card->controls_rwsem); return 0; } - EXPORT_SYMBOL(snd_ctl_rename_id); /** @@ -645,7 +648,6 @@ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numi } return NULL; } - EXPORT_SYMBOL(snd_ctl_find_numid); /** @@ -687,7 +689,6 @@ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, } return NULL; } - EXPORT_SYMBOL(snd_ctl_find_id); static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, @@ -1526,19 +1527,28 @@ static int _snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn, struct list_head * return 0; } +/** + * snd_ctl_register_ioctl - register the device-specific control-ioctls + * @fcn: ioctl callback function + * + * called from each device manager like pcm.c, hwdep.c, etc. + */ int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_register_ioctl(fcn, &snd_control_ioctls); } - EXPORT_SYMBOL(snd_ctl_register_ioctl); #ifdef CONFIG_COMPAT +/** + * snd_ctl_register_ioctl_compat - register the device-specific 32bit compat + * control-ioctls + * @fcn: ioctl callback function + */ int snd_ctl_register_ioctl_compat(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_register_ioctl(fcn, &snd_control_compat_ioctls); } - EXPORT_SYMBOL(snd_ctl_register_ioctl_compat); #endif @@ -1566,19 +1576,26 @@ static int _snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn, return -EINVAL; } +/** + * snd_ctl_unregister_ioctl - de-register the device-specific control-ioctls + * @fcn: ioctl callback function to unregister + */ int snd_ctl_unregister_ioctl(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_unregister_ioctl(fcn, &snd_control_ioctls); } - EXPORT_SYMBOL(snd_ctl_unregister_ioctl); #ifdef CONFIG_COMPAT +/** + * snd_ctl_unregister_ioctl - de-register the device-specific compat 32bit + * control-ioctls + * @fcn: ioctl callback function to unregister + */ int snd_ctl_unregister_ioctl_compat(snd_kctl_ioctl_func_t fcn) { return _snd_ctl_unregister_ioctl(fcn, &snd_control_compat_ioctls); } - EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat); #endif @@ -1702,6 +1719,16 @@ int snd_ctl_create(struct snd_card *card) /* * Frequently used control callbacks/helpers */ + +/** + * snd_ctl_boolean_mono_info - Helper function for a standard boolean info + * callback with a mono channel + * @kcontrol: the kcontrol instance + * @uinfo: info to store + * + * This is a function that can be used as info callback for a standard + * boolean control with a single mono channel. + */ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1711,9 +1738,17 @@ int snd_ctl_boolean_mono_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 1; return 0; } - EXPORT_SYMBOL(snd_ctl_boolean_mono_info); +/** + * snd_ctl_boolean_stereo_info - Helper function for a standard boolean info + * callback with stereo two channels + * @kcontrol: the kcontrol instance + * @uinfo: info to store + * + * This is a function that can be used as info callback for a standard + * boolean control with stereo two channels. + */ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1723,7 +1758,6 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 1; return 0; } - EXPORT_SYMBOL(snd_ctl_boolean_stereo_info); /** -- cgit v1.1 From f213d8f79a3928eaa2e2936f8ab40761aac04b95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 13:45:58 +0100 Subject: ALSA: pcm: Use static inline for snd_pcm_lib_alloc_vmalloc_buffer() ... instead of #if 0 hack. It's more straightforward and obvious. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e862497..5c3310d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -942,7 +942,6 @@ int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, unsigned long offset); -#if 0 /* for kernel-doc */ /** * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer * @substream: the substream to allocate the buffer to @@ -955,8 +954,13 @@ struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, * Return: 1 if the buffer was changed, 0 if not changed, or a negative error * code. */ -static int snd_pcm_lib_alloc_vmalloc_buffer - (struct snd_pcm_substream *substream, size_t size); +static inline int snd_pcm_lib_alloc_vmalloc_buffer + (struct snd_pcm_substream *substream, size_t size) +{ + return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size, + GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO); +} + /** * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer * @substream: the substream to allocate the buffer to @@ -968,15 +972,12 @@ static int snd_pcm_lib_alloc_vmalloc_buffer * Return: 1 if the buffer was changed, 0 if not changed, or a negative error * code. */ -static int snd_pcm_lib_alloc_vmalloc_32_buffer - (struct snd_pcm_substream *substream, size_t size); -#endif -#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \ - _snd_pcm_lib_alloc_vmalloc_buffer \ - (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO) -#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \ - _snd_pcm_lib_alloc_vmalloc_buffer \ - (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO) +static inline int snd_pcm_lib_alloc_vmalloc_32_buffer + (struct snd_pcm_substream *substream, size_t size) +{ + return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size, + GFP_KERNEL | GFP_DMA32 | __GFP_ZERO); +} #define snd_pcm_get_dma_buf(substream) ((substream)->runtime->dma_buffer_p) -- cgit v1.1 From 35c0a8c0178ad3f6f14e1dd76f0317156deaae51 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 30 Oct 2014 21:21:52 +0800 Subject: ASoC: Intel: Fix block is enabled multiple times issue During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 35788ad..c451398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -149,7 +149,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- cgit v1.1 From 30b771cf8c3120c5c946811ecc5a9b87a34003a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 15:02:50 +0100 Subject: ALSA: pcm: More kerneldoc updates Add proper kerneldoc comments to the exported functions. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 160 +++++++++++++++++++++++++++++++++++++++++++++--- sound/core/pcm.c | 15 ++++- sound/core/pcm_native.c | 53 ++++++++++++++++ 3 files changed, 219 insertions(+), 9 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 5c3310d..f78a572 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -533,6 +533,12 @@ snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) * PCM library */ +/** + * snd_pcm_stream_linked - Check whether the substream is linked with others + * @substream: substream to check + * + * Returns true if the given substream is being linked with others. + */ static inline int snd_pcm_stream_linked(struct snd_pcm_substream *substream) { return substream->group != &substream->self_group; @@ -543,6 +549,16 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream); void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); + +/** + * snd_pcm_stream_lock_irqsave - Lock the PCM stream + * @substream: PCM substream + * @flags: irq flags + * + * This locks the PCM stream like snd_pcm_stream_lock() but with the local + * IRQ (only when nonatomic is false). In nonatomic case, this is identical + * as snd_pcm_stream_lock(). + */ #define snd_pcm_stream_lock_irqsave(substream, flags) \ do { \ typecheck(unsigned long, flags); \ @@ -551,9 +567,25 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags); +/** + * snd_pcm_group_for_each_entry - iterate over the linked substreams + * @s: the iterator + * @substream: the substream + * + * Iterate over the all linked substreams to the given @substream. + * When @substream isn't linked with any others, this gives returns @substream + * itself once. + */ #define snd_pcm_group_for_each_entry(s, substream) \ list_for_each_entry(s, &substream->group->substreams, link_list) +/** + * snd_pcm_running - Check whether the substream is in a running state + * @substream: substream to check + * + * Returns true if the given substream is in the state RUNNING, or in the + * state DRAINING for playback. + */ static inline int snd_pcm_running(struct snd_pcm_substream *substream) { return (substream->runtime->status->state == SNDRV_PCM_STATE_RUNNING || @@ -561,45 +593,81 @@ static inline int snd_pcm_running(struct snd_pcm_substream *substream) substream->stream == SNDRV_PCM_STREAM_PLAYBACK)); } +/** + * bytes_to_samples - Unit conversion of the size from bytes to samples + * @runtime: PCM runtime instance + * @size: size in bytes + */ static inline ssize_t bytes_to_samples(struct snd_pcm_runtime *runtime, ssize_t size) { return size * 8 / runtime->sample_bits; } +/** + * bytes_to_frames - Unit conversion of the size from bytes to frames + * @runtime: PCM runtime instance + * @size: size in bytes + */ static inline snd_pcm_sframes_t bytes_to_frames(struct snd_pcm_runtime *runtime, ssize_t size) { return size * 8 / runtime->frame_bits; } +/** + * samples_to_bytes - Unit conversion of the size from samples to bytes + * @runtime: PCM runtime instance + * @size: size in samples + */ static inline ssize_t samples_to_bytes(struct snd_pcm_runtime *runtime, ssize_t size) { return size * runtime->sample_bits / 8; } +/** + * frames_to_bytes - Unit conversion of the size from frames to bytes + * @runtime: PCM runtime instance + * @size: size in frames + */ static inline ssize_t frames_to_bytes(struct snd_pcm_runtime *runtime, snd_pcm_sframes_t size) { return size * runtime->frame_bits / 8; } +/** + * frame_aligned - Check whether the byte size is aligned to frames + * @runtime: PCM runtime instance + * @bytes: size in bytes + */ static inline int frame_aligned(struct snd_pcm_runtime *runtime, ssize_t bytes) { return bytes % runtime->byte_align == 0; } +/** + * snd_pcm_lib_buffer_bytes - Get the buffer size of the current PCM in bytes + * @substream: PCM substream + */ static inline size_t snd_pcm_lib_buffer_bytes(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; return frames_to_bytes(runtime, runtime->buffer_size); } +/** + * snd_pcm_lib_period_bytes - Get the period size of the current PCM in bytes + * @substream: PCM substream + */ static inline size_t snd_pcm_lib_period_bytes(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; return frames_to_bytes(runtime, runtime->period_size); } -/* - * result is: 0 ... (boundary - 1) +/** + * snd_pcm_playback_avail - Get the available (writable) space for playback + * @runtime: PCM runtime instance + * + * Result is between 0 ... (boundary - 1) */ static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *runtime) { @@ -611,8 +679,11 @@ static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *r return avail; } -/* - * result is: 0 ... (boundary - 1) +/** + * snd_pcm_playback_avail - Get the available (readable) space for capture + * @runtime: PCM runtime instance + * + * Result is between 0 ... (boundary - 1) */ static inline snd_pcm_uframes_t snd_pcm_capture_avail(struct snd_pcm_runtime *runtime) { @@ -622,11 +693,19 @@ static inline snd_pcm_uframes_t snd_pcm_capture_avail(struct snd_pcm_runtime *ru return avail; } +/** + * snd_pcm_playback_hw_avail - Get the queued space for playback + * @runtime: PCM runtime instance + */ static inline snd_pcm_sframes_t snd_pcm_playback_hw_avail(struct snd_pcm_runtime *runtime) { return runtime->buffer_size - snd_pcm_playback_avail(runtime); } +/** + * snd_pcm_capture_hw_avail - Get the free space for capture + * @runtime: PCM runtime instance + */ static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime *runtime) { return runtime->buffer_size - snd_pcm_capture_avail(runtime); @@ -706,6 +785,20 @@ static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream) return snd_pcm_capture_avail(runtime) == 0; } +/** + * snd_pcm_trigger_done - Mark the master substream + * @substream: the pcm substream instance + * @master: the linked master substream + * + * When multiple substreams of the same card are linked and the hardware + * supports the single-shot operation, the driver calls this in the loop + * in snd_pcm_group_for_each_entry() for marking the substream as "done". + * Then most of trigger operations are performed only to the given master + * substream. + * + * The trigger_master mark is cleared at timestamp updates at the end + * of trigger operations. + */ static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, struct snd_pcm_substream *master) { @@ -881,6 +974,14 @@ unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, unsigned int rates_b); +/** + * snd_pcm_set_runtime_buffer - Set the PCM runtime buffer + * @substream: PCM substream to set + * @bufp: the buffer information, NULL to clear + * + * Copy the buffer information to runtime->dma_buffer when @bufp is non-NULL. + * Otherwise it clears the current buffer information. + */ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) { @@ -906,6 +1007,11 @@ void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); +/** + * snd_pcm_gettime - Fill the timespec depending on the timestamp mode + * @runtime: PCM runtime instance + * @tv: timespec to fill + */ static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, struct timespec *tv) { @@ -997,18 +1103,35 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, #define snd_pcm_sgbuf_ops_page NULL #endif /* SND_DMA_SGBUF */ +/** + * snd_pcm_sgbuf_get_addr - Get the DMA address at the corresponding offset + * @substream: PCM substream + * @ofs: byte offset + */ static inline dma_addr_t snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs) { return snd_sgbuf_get_addr(snd_pcm_get_dma_buf(substream), ofs); } +/** + * snd_pcm_sgbuf_get_ptr - Get the virtual address at the corresponding offset + * @substream: PCM substream + * @ofs: byte offset + */ static inline void * snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) { return snd_sgbuf_get_ptr(snd_pcm_get_dma_buf(substream), ofs); } +/** + * snd_pcm_sgbuf_chunk_size - Compute the max size that fits within the contig. + * page from the given size + * @substream: PCM substream + * @ofs: byte offset + * @size: byte size to examine + */ static inline unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, unsigned int ofs, unsigned int size) @@ -1016,13 +1139,24 @@ snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, return snd_sgbuf_get_chunk_size(snd_pcm_get_dma_buf(substream), ofs, size); } -/* handle mmap counter - PCM mmap callback should handle this counter properly */ +/** + * snd_pcm_mmap_data_open - increase the mmap counter + * @area: VMA + * + * PCM mmap callback should handle this counter properly + */ static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area) { struct snd_pcm_substream *substream = (struct snd_pcm_substream *)area->vm_private_data; atomic_inc(&substream->mmap_count); } +/** + * snd_pcm_mmap_data_close - decrease the mmap counter + * @area: VMA + * + * PCM mmap callback should handle this counter properly + */ static inline void snd_pcm_mmap_data_close(struct vm_area_struct *area) { struct snd_pcm_substream *substream = (struct snd_pcm_substream *)area->vm_private_data; @@ -1042,6 +1176,11 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_vmalloc NULL +/** + * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer + * @dma: DMA number + * @max: pointer to store the max size + */ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { *max = dma < 4 ? 64 * 1024 : 128 * 1024; @@ -1094,7 +1233,11 @@ struct snd_pcm_chmap { void *private_data; /* optional: private data pointer */ }; -/* get the PCM substream assigned to the given chmap info */ +/** + * snd_pcm_chmap_substream - get the PCM substream assigned to the given chmap info + * @info: chmap information + * @idx: the substream number index + */ static inline struct snd_pcm_substream * snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx) { @@ -1121,7 +1264,10 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, unsigned long private_value, struct snd_pcm_chmap **info_ret); -/* Strong-typed conversion of pcm_format to bitwise */ +/** + * pcm_format_to_bits - Strong-typed conversion of pcm_format to bitwise + * @pcm_format: PCM format + */ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) { return 1ULL << (__force int) pcm_format; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 42ded99..31acc3d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -218,6 +218,10 @@ static char *snd_pcm_format_names[] = { FORMAT(DSD_U32_LE), }; +/** + * snd_pcm_format_name - Return a name string for the given PCM format + * @format: PCM format + */ const char *snd_pcm_format_name(snd_pcm_format_t format) { if ((__force unsigned int)format >= ARRAY_SIZE(snd_pcm_format_names)) @@ -707,7 +711,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) } return 0; } - EXPORT_SYMBOL(snd_pcm_new_stream); static int _snd_pcm_new(struct snd_card *card, const char *id, int device, @@ -1155,6 +1158,15 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) return 0; } +/** + * snd_pcm_notify - Add/remove the notify list + * @notify: PCM notify list + * @nfree: 0 = register, 1 = unregister + * + * This adds the given notifier to the global list so that the callback is + * called for each registered PCM devices. This exists only for PCM OSS + * emulation, so far. + */ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) { struct snd_pcm *pcm; @@ -1177,7 +1189,6 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) mutex_unlock(®ister_mutex); return 0; } - EXPORT_SYMBOL(snd_pcm_notify); #ifdef CONFIG_PROC_FS diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 2f7ad10..4d5795d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -74,6 +74,14 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); static DEFINE_RWLOCK(snd_pcm_link_rwlock); static DECLARE_RWSEM(snd_pcm_link_rwsem); +/** + * snd_pcm_stream_lock - Lock the PCM stream + * @substream: PCM substream + * + * This locks the PCM stream's spinlock or mutex depending on the nonatomic + * flag of the given substream. This also takes the global link rw lock + * (or rw sem), too, for avoiding the race with linked streams. + */ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) { if (substream->pcm->nonatomic) { @@ -86,6 +94,12 @@ void snd_pcm_stream_lock(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock); +/** + * snd_pcm_stream_lock - Unlock the PCM stream + * @substream: PCM substream + * + * This unlocks the PCM stream that has been locked via snd_pcm_stream_lock(). + */ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream) { if (substream->pcm->nonatomic) { @@ -98,6 +112,14 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock); +/** + * snd_pcm_stream_lock_irq - Lock the PCM stream + * @substream: PCM substream + * + * This locks the PCM stream like snd_pcm_stream_lock() and disables the local + * IRQ (only when nonatomic is false). In nonatomic case, this is identical + * as snd_pcm_stream_lock(). + */ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) { if (!substream->pcm->nonatomic) @@ -106,6 +128,12 @@ void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq); +/** + * snd_pcm_stream_unlock_irq - Unlock the PCM stream + * @substream: PCM substream + * + * This is a counter-part of snd_pcm_stream_lock_irq(). + */ void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream) { snd_pcm_stream_unlock(substream); @@ -124,6 +152,13 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); +/** + * snd_pcm_stream_unlock_irqrestore - Unlock the PCM stream + * @substream: PCM substream + * @flags: irq flags + * + * This is a counter-part of snd_pcm_stream_lock_irqsave(). + */ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags) { @@ -3312,6 +3347,15 @@ static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { /* * mmap the DMA buffer on RAM */ + +/** + * snd_pcm_lib_default_mmap - Default PCM data mmap function + * @substream: PCM substream + * @area: VMA + * + * This is the default mmap handler for PCM data. When mmap pcm_ops is NULL, + * this function is invoked implicitly. + */ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3343,6 +3387,15 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM +/** + * snd_pcm_lib_mmap_iomem - Default PCM data mmap function for I/O mem + * @substream: PCM substream + * @area: VMA + * + * When your hardware uses the iomapped pages as the hardware buffer and + * wants to mmap it, pass this function as mmap pcm_ops. Note that this + * is supposed to work only on limited architectures. + */ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { -- cgit v1.1 From 85926e0fe8b48fe6227614ddc3a1cb41c0a3c51a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 15:06:13 +0100 Subject: ALSA: pcm: Convert params_*() with static inline functions ... and add proper kerneldoc comments. There is no big reason to keep them as macros. Static inline functions are safer in general, and suitable for kerneldoc, too. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 65 +++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 53 insertions(+), 12 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f78a572..29eb09ef 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -841,18 +841,59 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } -#define params_channels(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min) -#define params_rate(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_RATE)->min) -#define params_period_size(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min) -#define params_periods(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIODS)->min) -#define params_buffer_size(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min) -#define params_buffer_bytes(p) \ - (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min) +/** + * params_channels - Get the number of channels from the hw params + * @p: hw params + */ +static inline unsigned int params_channels(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_CHANNELS)->min; +} + +/** + * params_channels - Get the sample rate from the hw params + * @p: hw params + */ +static inline unsigned int params_rate(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_RATE)->min; +} + +/** + * params_channels - Get the period size (in frames) from the hw params + * @p: hw params + */ +static inline unsigned int params_period_size(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min; +} + +/** + * params_channels - Get the number of periods from the hw params + * @p: hw params + */ +static inline unsigned int params_periods(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_PERIODS)->min; +} + +/** + * params_channels - Get the buffer size (in frames) from the hw params + * @p: hw params + */ +static inline unsigned int params_buffer_size(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min; +} + +/** + * params_channels - Get the buffer size (in bytes) from the hw params + * @p: hw params + */ +static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p) +{ + return hw_param_interval_c(p, SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min; +} int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); -- cgit v1.1 From eb9c38d54c9cad72101dfe7fefe4a784dd67da86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 15:19:43 +0100 Subject: ALSA: doc: More kerneldoc comments on core components Some functions missed the proper kerneldoc comments. Signed-off-by: Takashi Iwai --- sound/core/init.c | 33 ++++++++++++++++++++++----------- sound/core/sound.c | 9 +++++++-- 2 files changed, 29 insertions(+), 13 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 7bdfd19..074875d 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -438,17 +438,6 @@ int snd_card_disconnect(struct snd_card *card) EXPORT_SYMBOL(snd_card_disconnect); -/** - * snd_card_free - frees given soundcard structure - * @card: soundcard structure - * - * This function releases the soundcard structure and the all assigned - * devices automatically. That is, you don't have to release the devices - * by yourself. - * - * Return: Zero. Frees all associated devices and frees the control - * interface associated to given soundcard. - */ static int snd_card_do_free(struct snd_card *card) { #if IS_ENABLED(CONFIG_SND_MIXER_OSS) @@ -469,6 +458,15 @@ static int snd_card_do_free(struct snd_card *card) return 0; } +/** + * snd_card_free_when_closed - Disconnect the card, free it later eventually + * @card: soundcard structure + * + * Unlike snd_card_free(), this function doesn't try to release the card + * resource immediately, but tries to disconnect at first. When the card + * is still in use, the function returns before freeing the resources. + * The card resources will be freed when the refcount gets to zero. + */ int snd_card_free_when_closed(struct snd_card *card) { int ret = snd_card_disconnect(card); @@ -479,6 +477,19 @@ int snd_card_free_when_closed(struct snd_card *card) } EXPORT_SYMBOL(snd_card_free_when_closed); +/** + * snd_card_free - frees given soundcard structure + * @card: soundcard structure + * + * This function releases the soundcard structure and the all assigned + * devices automatically. That is, you don't have to release the devices + * by yourself. + * + * This function waits until the all resources are properly released. + * + * Return: Zero. Frees all associated devices and frees the control + * interface associated to given soundcard. + */ int snd_card_free(struct snd_card *card) { struct completion released; diff --git a/sound/core/sound.c b/sound/core/sound.c index 38ad1a0..f133306 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -355,8 +355,13 @@ int snd_unregister_device(int type, struct snd_card *card, int dev) EXPORT_SYMBOL(snd_unregister_device); -/* get the assigned device to the given type and device number; - * the caller needs to release it via put_device() after using it +/** + * snd_get_device - get the assigned device to the given type and device number + * @type: the device type, SNDRV_DEVICE_TYPE_XXX + * @card:the card instance + * @dev: the device index + * + * The caller needs to release it via put_device() after using it. */ struct device *snd_get_device(int type, struct snd_card *card, int dev) { -- cgit v1.1 From 67faa6ebd7fc0a811f6c1f3e3c113571953489ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2014 15:37:37 +0100 Subject: ALSA: doc: Fix missing "I" for kerneldoc inclusion Fixes: 90446d0746c3 ('ALSA: doc: Add missing headers and compress stuff to alsa-driver-api.tmpl') Signed-off-by: Takashi Iwai --- Documentation/DocBook/alsa-driver-api.tmpl | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index b2d2c3b..71f9246 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -57,7 +57,7 @@ !Esound/core/pcm.c !Esound/core/pcm_lib.c !Esound/core/pcm_native.c -!include/sound/pcm.h +!Iinclude/sound/pcm.h PCM Format Helpers !Esound/core/pcm_misc.c -- cgit v1.1 From b891f62fcd28a46ab0818cd9acbb5bbb20542ab6 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 30 Oct 2014 14:34:00 +0000 Subject: ASoC: Intel: Add debug output when boot fails. Add the debug output from IPCD and IPCX when booting fails. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b37d3ee..0ea7c3d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1927,7 +1927,9 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { - dev_err(hsw->dev, "error: audio DSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); return -EIO; } @@ -2038,7 +2040,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { ret = -EIO; - dev_err(hsw->dev, "error: ADSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); goto boot_err; } -- cgit v1.1 From 35e03a884c41b8fecf77e20de89759deb7c9078a Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 30 Oct 2014 14:58:19 +0000 Subject: ASoC: Intel: fix build with runtime PM disabled. Fix the following errors: All error/warnings: >> sound/soc/intel/sst-haswell-pcm.c:1168:13: error: 'hsw_pcm_prepare' undeclared here (not in a function) .prepare = hsw_pcm_prepare, ^ >> sound/soc/intel/sst-haswell-pcm.c:1169:14: error: 'hsw_pcm_complete' undeclared here (not in a function) .complete = hsw_pcm_complete, ^ Reported-by: kbuild test robot Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 14 ++++++-------- 2 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ea7c3d..ffd5728 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1711,7 +1711,7 @@ void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime) sst_module_runtime_free(runtime); } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) { struct sst_dsp *sst = hsw->dsp; diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 4489a35..cd54dd9 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1058,6 +1058,12 @@ static int hsw_pcm_runtime_resume(struct device *dev) return ret; } +#else +#define hsw_pcm_runtime_idle NULL +#define hsw_pcm_runtime_suspend NULL +#define hsw_pcm_runtime_resume NULL +#endif + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1153,14 +1159,6 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } -#else -#define hsw_pcm_runtime_idle NULL -#define hsw_pcm_runtime_suspend NULL -#define hsw_pcm_runtime_resume NULL -#define hsw_pcm_runtime_complete NULL -#define hsw_pcm_runtime_prepare NULL -#endif - static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, -- cgit v1.1 From 0d2135ecadb0b2eec5338a7587ba29724ddf612b Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 30 Oct 2014 22:57:58 +0800 Subject: ASoC: Intel: Work around to fix HW D3 potential crash issue When using clock gatings to save power, there are some known issues: 1. core clock gating (DCLCGE) must be disabled during D0 and D3 entry and updating SRAM banks (VDRTCTL0). 2. DSP trunk clock gating (DTCGE) can cause FW crashes, disable it in D0. To align with the new W/A flow from FW team, we must set VDRTCTL0.D3PGD to 1 (D3 power gating disabled) at first startup and keep it all the time. ADSP will be in D0 on first boot by BIOS part of WA. Required delays must be preserved (waiting for HW to stabilize, after enabling CCG, changing SRAM PG, D3PG). D3->D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Enable other CG apart from DTCG and DCLCG (VDRTCTL2. DCLCGE and DTCGE = 0) 3. Disable D3PG (VDRTCTL0.D3PGD = 1) 4. Power up necessary SRAM and wait at least for 18 clock cycles for every bank you have powered up 5. Set D0 state(PMCS.PS = 0), wait for HW 6. Restore MCLK (clkctl.smos, disabled in D3 entry point 4) 7. Stall and reset core, set CSR 8. Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us 9. Unreset core 10.Load FW, configure PLL and other necessary things 11.Unstall core Changing SRAM PG during D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Set PG mask 3. Wait at least for 18 clock cycles for every bank you have powered up 4. Enable core clock gating, delay 50 us D0->D3: 1. Disable core clock gating (DCLCGE = 0) 2. Stall and reset core 3. Power down entire SRAM and wait at least for 18 clock cycles for every bank (Enable SRAM PG (ISRAMPGE = 0x3FF, DSRAMPGE = 0xFFFFF, D3SRAMPGD = 0), remember about preserving VDRTCTL0.D3PGD = 1) 4. Shutdown PLL, disable MCLK(clkctl.smos = 0), Enable DTCG to save power 5. Set D3 state(PMCS.PS = 3), delay 50 us 6. Enable core clock gating, delay 50 us Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-dsp.h | 14 ++++-- sound/soc/intel/sst-haswell-dsp.c | 102 ++++++++++++++++++++++++++++++++------ 2 files changed, 98 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 2753b85..f291e32 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -159,12 +159,18 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ -#define SST_VDRTCL0_APLLSE_MASK 1 -#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 -#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) -#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_D3PGD (1 << 0) +#define SST_VDRTCL0_D3SRAMPGD (1 << 1) +#define SST_VDRTCL0_DSRAMPGE_SHIFT 12 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xfffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 2 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* VDRTCTL2 */ +#define SST_VDRTCL2_DCLCGE (1 << 1) +#define SST_VDRTCL2_DTCGE (1 << 10) +#define SST_VDRTCL2_APLLSE_MASK (1 << 31) + /* PMCS */ #define SST_PMCS 0x84 #define SST_PMCS_PS_MASK 0x3 diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 86aea34..57039b0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -250,17 +250,42 @@ static irqreturn_t hsw_irq(int irq, void *context) static void hsw_set_dsp_D3(struct sst_dsp *sst) { u32 val; + u32 reg; + + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch off audio PLL, DRAM & IRAM blocks */ + /* enable power gating and switch off DRAM & IRAM blocks */ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + val |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* Set D3 state */ + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* disable MCLK(clkctl.smos = 0) */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK, 0); + + /* Set D3 state, delay 50 us */ val = readl(sst->addr.pci_cfg + SST_PMCS); val |= SST_PMCS_PS_MASK; writel(val, sst->addr.pci_cfg + SST_PMCS); + udelay(50); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -283,6 +308,16 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) int tries = 10; u32 reg; + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); reg &= ~SST_PMCS_PS_MASK; @@ -300,14 +335,6 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) return -ENODEV; finish: - hsw_reset(sst); - - /* switch on audio PLL, DRAM & IRAM blocks */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | - SST_VDRTCL0_ISRAMPGE_MASK); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -322,6 +349,28 @@ finish: SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + + /* switch on audio PLL */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~SST_VDRTCL2_APLLSE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); @@ -343,9 +392,6 @@ finish: sst_dsp_shim_write(sst, 0x80, 0x6); sst_dsp_shim_write(sst, 0xe0, 0x300a); - /* disable all clock gating */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); - return 0; } @@ -497,6 +543,11 @@ static int hsw_block_enable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); @@ -504,6 +555,13 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ sst_mem_block_dummy_read(block); return 0; @@ -521,10 +579,26 @@ static int hsw_block_disable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* wait 18 DSP clock ticks */ + udelay(10); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + return 0; } -- cgit v1.1 From 6879db7648b6b995122afa98df31778c7af0855d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 31 Oct 2014 14:52:16 +0800 Subject: ASoC: rt286: reduce power consumption This patch will optimize the power consumption of rt286. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 211 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 155 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 4aa555c..97daa80 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,11 +36,13 @@ struct rt286_priv { struct regmap *regmap; + struct snd_soc_codec *codec; struct rt286_platform_data pdata; struct i2c_client *i2c; struct snd_soc_jack *jack; struct delayed_work jack_detect_work; int sys_clk; + int clk_id; struct reg_default *index_cache; }; @@ -298,7 +300,6 @@ static int rt286_support_power_controls[] = { static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) { unsigned int val, buf; - int i; *hp = false; *mic = false; @@ -309,67 +310,44 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) if (*hp) { /* power on HV,VERF */ regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x0); + RT286_DC_GAIN, 0x200, 0x200); + + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "HV"); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "VREF"); /* power LDO1 */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL2, 0x4, 0x4); - regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); - regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); - msleep(200); - i = 40; - while (((val & 0x0800) == 0) && (i > 0)) { - regmap_read(rt286->regmap, - RT286_CBJ_CTRL2, &val); - i--; - msleep(20); - } + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); + msleep(50); - if (0x0400 == (val & 0x0700)) { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xd400); + msleep(300); + regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); - regmap_write(rt286->regmap, - RT286_SET_MIC1, 0x20); - /* power off HV,VERF */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x1001); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x0000); - } else if ((0x0200 == (val & 0x0700)) || - (0x0100 == (val & 0x0700))) { + if (0x0070 == (val & 0x0070)) { *mic = true; - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x8000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x8000); } else { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xe400); + msleep(300); + regmap_read(rt286->regmap, + RT286_CBJ_CTRL2, &val); + if (0x0070 == (val & 0x0070)) + *mic = true; + else + *mic = false; } - - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0000); - } else { - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, - 0xc000, 0x8000); regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, - 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, - 0xc000, 0x8000); + RT286_DC_GAIN, 0x200, 0x0); + } else { *mic = false; + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20); } } else { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); @@ -378,6 +356,12 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *mic = buf & 0x80000000; } + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV"); + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF"); + if (!*hp) + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; } @@ -415,6 +399,17 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +static int is_mclk_mode(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec); + + if (rt286->clk_id == RT286_SCLK_S_MCLK) + return 1; + else + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -568,7 +563,84 @@ static int rt286_adc_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt286_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0000); + mdelay(50); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x08); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x30); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, + 12, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, + 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, + 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, + 0, 0, rt286_mic1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1 Pin"), SND_SOC_DAPM_INPUT("DMIC2 Pin"), @@ -642,6 +714,25 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"ADC 0", NULL, "MCLK MODE", is_mclk_mode}, + {"ADC 1", NULL, "MCLK MODE", is_mclk_mode}, + {"Front", NULL, "MCLK MODE", is_mclk_mode}, + {"Surround", NULL, "MCLK MODE", is_mclk_mode}, + + {"HP Power", NULL, "LDO1"}, + {"HP Power", NULL, "LDO2"}, + + {"MIC1", NULL, "LDO1"}, + {"MIC1", NULL, "LDO2"}, + {"MIC1", NULL, "HV"}, + {"MIC1", NULL, "VREF"}, + {"MIC1", NULL, "MIC1 Input Buffer"}, + + {"SPO", NULL, "LDO1"}, + {"SPO", NULL, "LDO2"}, + {"SPO", NULL, "HV"}, + {"SPO", NULL, "VREF"}, + {"DMIC1", NULL, "DMIC1 Pin"}, {"DMIC2", NULL, "DMIC2 Pin"}, {"DMIC1", NULL, "DMIC Receiver"}, @@ -880,6 +971,7 @@ static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, } rt286->sys_clk = freq; + rt286->clk_id = clk_id; return 0; } @@ -915,13 +1007,18 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: mdelay(10); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0400); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); snd_soc_update_bits(codec, - RT286_DC_GAIN, 0x200, 0x0); + RT286_CBJ_CTRL1, 0x0400, 0x0000); break; default: @@ -962,6 +1059,7 @@ static int rt286_probe(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->codec = codec; codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { @@ -1152,7 +1250,6 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (!rt286->pdata.cbj_en) { regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); - regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { @@ -1169,8 +1266,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); - /*Power down LDO2*/ - regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + /*Power down LDO, VREF*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); /*Set depop parameter*/ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); -- cgit v1.1 From d004ebbef7292848f5f7ecae50824c04780baaac Mon Sep 17 00:00:00 2001 From: Max Filippov Date: Wed, 29 Oct 2014 16:25:38 +0300 Subject: ASoC: tlv320aic23: make codecs selectable in Kconfig Now that manual selection of drivers for audio subsystem components is preferred AIC23 codec must be selectable in Kconfig to make it possible. Signed-off-by: Max Filippov Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..7881b3c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -581,11 +581,11 @@ config SND_SOC_TLV320AIC23 tristate config SND_SOC_TLV320AIC23_I2C - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- cgit v1.1 From 1a6db0bd26a72027d6a5ea006d64d4021fd0326e Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 10:38:54 +0530 Subject: ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream It's already done in open/close. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 183b1eb..4187057 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -163,16 +163,11 @@ static int sst_open_pcm_stream(struct device *dev, if (!str_param) return -EINVAL; - retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) - return retval; retval = sst_get_stream(ctx, str_param); - if (retval > 0) { + if (retval > 0) ctx->stream_cnt++; - } else { + else dev_err(ctx->dev, "sst_get_stream returned err %d\n", retval); - sst_pm_runtime_put(ctx); - } return retval; } @@ -212,7 +207,8 @@ put: stream->period_elapsed = NULL; ctx->stream_cnt--; - sst_pm_runtime_put(ctx); + if (retval) + dev_err(ctx->dev, "free stream returned err %d\n", retval); dev_dbg(ctx->dev, "Exit\n"); return 0; -- cgit v1.1 From d62f2a08b9d657344b2e271e8274f9d8f746e543 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 31 Oct 2014 12:38:20 +0530 Subject: ASoC: Intel: sst: add runtime power management handling This patch adds the runtime pm handlers, the driver already has code for get/put for runtime pm and only these handlers being missing. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 67 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 67 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index fa34217..7b8a110 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -152,6 +152,23 @@ static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) return IRQ_HANDLED; } +static int sst_save_dsp_context_v2(struct intel_sst_drv *sst) +{ + int ret = 0; + + ret = sst_prepare_and_post_msg(sst, SST_TASK_ID_MEDIA, IPC_CMD, + IPC_PREP_D3, PIPE_RSVD, 0, NULL, NULL, + true, true, false, true); + + if (ret < 0) { + dev_err(sst->dev, "not suspending FW!!, Err: %d\n", ret); + return -EIO; + } + + return 0; +} + + static struct intel_sst_ops mrfld_ops = { .interrupt = intel_sst_interrupt_mrfld, .irq_thread = intel_sst_irq_thread_mrfld, @@ -160,6 +177,7 @@ static struct intel_sst_ops mrfld_ops = { .reset = intel_sst_reset_dsp_mrfld, .post_message = sst_post_message_mrfld, .process_reply = sst_process_reply_mrfld, + .save_dsp_context = sst_save_dsp_context_v2, .alloc_stream = sst_alloc_stream_mrfld, .post_download = sst_post_download_mrfld, }; @@ -418,6 +436,50 @@ static void intel_sst_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +static int intel_sst_runtime_suspend(struct device *dev) +{ + int ret = 0; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state == SST_RESET) { + dev_dbg(dev, "LPE is already in RESET state, No action\n"); + return 0; + } + /* save fw context */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + return ret; +} + +static int intel_sst_runtime_resume(struct device *dev) +{ + int ret = 0; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + mutex_lock(&ctx->sst_lock); + if (ctx->sst_state == SST_RESET) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + ctx->sst_state = SST_RESET; + } + } + mutex_unlock(&ctx->sst_lock); + return ret; +} + +static const struct dev_pm_ops intel_sst_pm = { + .runtime_suspend = intel_sst_runtime_suspend, + .runtime_resume = intel_sst_runtime_resume, +}; + /* PCI Routines */ static struct pci_device_id intel_sst_ids[] = { { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, @@ -429,6 +491,11 @@ static struct pci_driver sst_driver = { .id_table = intel_sst_ids, .probe = intel_sst_probe, .remove = intel_sst_remove, +#ifdef CONFIG_PM + .driver = { + .pm = &intel_sst_pm, + }, +#endif }; module_pci_driver(sst_driver); -- cgit v1.1 From 45f31bfcda0c6e5f11168de10c85f3dd20337bdf Mon Sep 17 00:00:00 2001 From: Mythri P K Date: Fri, 31 Oct 2014 12:38:21 +0530 Subject: ASoC: Intel: use lock when changing SST state. SST state change should be done under sst_lock Signed-off-by: Mythri P K Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 4 +--- sound/soc/intel/sst/sst_ipc.c | 2 +- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 7b8a110..04af246 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -463,15 +463,13 @@ static int intel_sst_runtime_resume(struct device *dev) int ret = 0; struct intel_sst_drv *ctx = dev_get_drvdata(dev); - mutex_lock(&ctx->sst_lock); if (ctx->sst_state == SST_RESET) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); - ctx->sst_state = SST_RESET; + sst_set_fw_state_locked(ctx, SST_RESET); } } - mutex_unlock(&ctx->sst_lock); return ret; } diff --git a/sound/soc/intel/sst/sst_ipc.c b/sound/soc/intel/sst/sst_ipc.c index 2126f5b..484e609 100644 --- a/sound/soc/intel/sst/sst_ipc.c +++ b/sound/soc/intel/sst/sst_ipc.c @@ -230,7 +230,7 @@ static void process_fw_init(struct intel_sst_drv *sst_drv_ctx, dev_dbg(sst_drv_ctx->dev, "*** FW Init msg came***\n"); if (init->result) { - sst_drv_ctx->sst_state = SST_RESET; + sst_set_fw_state_locked(sst_drv_ctx, SST_RESET); dev_err(sst_drv_ctx->dev, "FW Init failed, Error %x\n", init->result); retval = init->result; -- cgit v1.1 From 6e9b05607fe896627ab7c15efff01b6dcae71a56 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 30 Oct 2014 16:20:56 +0530 Subject: ASoC: Intel: sst: load firmware using async callback We would like the DSP firmware to be available in driver as soon as possible. So use the async callback in driver to probe to load the firmware as soon as usermode is up Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 04af246..fdada40 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -364,6 +364,21 @@ static int intel_sst_probe(struct pci_dev *pci, sst_set_fw_state_locked(sst_drv_ctx, SST_RESET); + snprintf(sst_drv_ctx->firmware_name, sizeof(sst_drv_ctx->firmware_name), + "%s%04x%s", "fw_sst_", + sst_drv_ctx->dev_id, ".bin"); + dev_dbg(sst_drv_ctx->dev, + "Requesting FW %s now...\n", sst_drv_ctx->firmware_name); + ret = request_firmware_nowait(THIS_MODULE, 1, + sst_drv_ctx->firmware_name, sst_drv_ctx->dev, + GFP_KERNEL, sst_drv_ctx, sst_firmware_load_cb); + + if (ret) { + dev_err(sst_drv_ctx->dev, + "Firmware load failed with error: %d\n", ret); + goto do_release_regions; + } + sst_drv_ctx->irq_num = pci->irq; /* Register the ISR */ ret = devm_request_threaded_irq(&pci->dev, pci->irq, -- cgit v1.1 From 5794b7ec62d85700d372b07d88eaf71e807f542f Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Thu, 30 Oct 2014 16:20:57 +0530 Subject: ASoC: Intel: use correct firmware name The firmware name was used worngly, so fix it up Signed-off-by: Fang, Yang A Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_loader.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 00f60c1..b580f96 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -344,12 +344,9 @@ void sst_firmware_load_cb(const struct firmware *fw, void *context) static int sst_request_fw(struct intel_sst_drv *sst) { int retval = 0; - char name[20]; const struct firmware *fw; - dev_dbg(sst->dev, "Requesting FW %s now...\n", name); - - retval = request_firmware(&fw, name, sst->dev); + retval = request_firmware(&fw, sst->firmware_name, sst->dev); if (fw == NULL) { dev_err(sst->dev, "fw is returning as null\n"); return -EINVAL; -- cgit v1.1 From fdcc4a039f0263f4674e363ebed14783b2f0543d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:20:58 +0530 Subject: ASoC: mfld-compress: implement .power callback .power callback is required to invoked for compressed audio as well to turn on/off sst, so invoke them Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-compress.c | 8 +++++++- sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 59467775..3951689 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -67,8 +67,11 @@ static int sst_platform_compr_open(struct snd_compr_stream *cstream) goto out_ops; } stream->compr_ops = sst->compr_ops; - stream->id = 0; + + /* Turn on LPE */ + sst->compr_ops->power(sst->dev, true); + sst_set_stream_status(stream, SST_PLATFORM_INIT); runtime->private_data = stream; return 0; @@ -83,6 +86,9 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream) int ret_val = 0, str_id; stream = cstream->runtime->private_data; + /* Turn off LPE */ + sst->compr_ops->power(sst->dev, false); + /*need to check*/ str_id = stream->id; if (str_id) diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index d41d1c3..79c8d12 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -117,6 +117,7 @@ struct compress_sst_ops { int (*get_codec_caps)(struct snd_compr_codec_caps *codec); int (*set_metadata)(struct device *dev, unsigned int str_id, struct snd_compr_metadata *mdata); + int (*power)(struct device *dev, bool state); }; struct sst_ops { -- cgit v1.1 From 7adab122a57c5ade8efc2e4de67c72b084c31cda Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 30 Oct 2014 16:20:59 +0530 Subject: ASoC: Intel: sst - add compressed ops handling This patch add low level IPC handling for compressed stream operations Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 285 ++++++++++++++++++++++++++++++++ 1 file changed, 285 insertions(+) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 4187057..5f75ef3 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -172,6 +172,273 @@ static int sst_open_pcm_stream(struct device *dev, return retval; } +static int sst_cdev_open(struct device *dev, + struct snd_sst_params *str_params, struct sst_compress_cb *cb) +{ + int str_id, retval; + struct stream_info *stream; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + retval = pm_runtime_get_sync(ctx->dev); + if (retval < 0) + return retval; + + str_id = sst_get_stream(ctx, str_params); + if (str_id > 0) { + dev_dbg(dev, "stream allocated in sst_cdev_open %d\n", str_id); + stream = &ctx->streams[str_id]; + stream->compr_cb = cb->compr_cb; + stream->compr_cb_param = cb->param; + stream->drain_notify = cb->drain_notify; + stream->drain_cb_param = cb->drain_cb_param; + } else { + dev_err(dev, "stream encountered error during alloc %d\n", str_id); + str_id = -EINVAL; + sst_pm_runtime_put(ctx); + } + return str_id; +} + +static int sst_cdev_close(struct device *dev, unsigned int str_id) +{ + int retval; + struct stream_info *stream; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + stream = get_stream_info(ctx, str_id); + if (!stream) { + dev_err(dev, "stream info is NULL for str %d!!!\n", str_id); + return -EINVAL; + } + + if (stream->status == STREAM_RESET) { + dev_dbg(dev, "stream in reset state...\n"); + stream->status = STREAM_UN_INIT; + + retval = 0; + goto put; + } + + retval = sst_free_stream(ctx, str_id); +put: + stream->compr_cb_param = NULL; + stream->compr_cb = NULL; + + if (retval) + dev_err(dev, "free stream returned err %d\n", retval); + + dev_dbg(dev, "End\n"); + return retval; + +} + +static int sst_cdev_ack(struct device *dev, unsigned int str_id, + unsigned long bytes) +{ + struct stream_info *stream; + struct snd_sst_tstamp fw_tstamp = {0,}; + int offset; + void __iomem *addr; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + stream = get_stream_info(ctx, str_id); + if (!stream) + return -EINVAL; + + /* update bytes sent */ + stream->cumm_bytes += bytes; + dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); + + memcpy_fromio(&fw_tstamp, + ((void *)(ctx->mailbox + ctx->tstamp) + +(str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + + fw_tstamp.bytes_copied = stream->cumm_bytes; + dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", + fw_tstamp.bytes_copied, bytes); + + addr = ((void *)(ctx->mailbox + ctx->tstamp)) + + (str_id * sizeof(fw_tstamp)); + offset = offsetof(struct snd_sst_tstamp, bytes_copied); + sst_shim_write(addr, offset, fw_tstamp.bytes_copied); + return 0; +} + +static int sst_cdev_set_metadata(struct device *dev, + unsigned int str_id, struct snd_compr_metadata *metadata) +{ + int retval = 0; + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + dev_dbg(dev, "set metadata for stream %d\n", str_id); + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + dev_dbg(dev, "pipe id = %d\n", str_info->pipe_id); + retval = sst_prepare_and_post_msg(ctx, str_info->task_id, IPC_CMD, + IPC_IA_SET_STREAM_PARAMS_MRFLD, str_info->pipe_id, + sizeof(*metadata), metadata, NULL, + true, true, true, false); + + return retval; +} + +static int sst_cdev_stream_pause(struct device *dev, unsigned int str_id) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + return sst_pause_stream(ctx, str_id); +} + +static int sst_cdev_stream_pause_release(struct device *dev, + unsigned int str_id) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + return sst_resume_stream(ctx, str_id); +} + +static int sst_cdev_stream_start(struct device *dev, unsigned int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + str_info->prev = str_info->status; + str_info->status = STREAM_RUNNING; + return sst_start_stream(ctx, str_id); +} + +static int sst_cdev_stream_drop(struct device *dev, unsigned int str_id) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + return sst_drop_stream(ctx, str_id); +} + +static int sst_cdev_stream_drain(struct device *dev, unsigned int str_id) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + return sst_drain_stream(ctx, str_id, false); +} + +static int sst_cdev_stream_partial_drain(struct device *dev, + unsigned int str_id) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + return sst_drain_stream(ctx, str_id, true); +} + +static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, + struct snd_compr_tstamp *tstamp) +{ + struct snd_sst_tstamp fw_tstamp = {0,}; + struct stream_info *stream; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + memcpy_fromio(&fw_tstamp, + ((void *)(ctx->mailbox + ctx->tstamp) + +(str_id * sizeof(fw_tstamp))), + sizeof(fw_tstamp)); + + stream = get_stream_info(ctx, str_id); + if (!stream) + return -EINVAL; + dev_dbg(dev, "rb_counter %llu in bytes\n", fw_tstamp.ring_buffer_counter); + + tstamp->copied_total = fw_tstamp.ring_buffer_counter; + tstamp->pcm_frames = fw_tstamp.frames_decoded; + tstamp->pcm_io_frames = div_u64(fw_tstamp.hardware_counter, + (u64)((stream->num_ch) * SST_GET_BYTES_PER_SAMPLE(24))); + tstamp->sampling_rate = fw_tstamp.sampling_frequency; + + dev_dbg(dev, "PCM = %u\n", tstamp->pcm_io_frames); + dev_dbg(dev, "Ptr Query on strid = %d copied_total %d, decodec %d\n", + str_id, tstamp->copied_total, tstamp->pcm_frames); + dev_dbg(dev, "rendered %d\n", tstamp->pcm_io_frames); + + return 0; +} + +static int sst_cdev_caps(struct snd_compr_caps *caps) +{ + caps->num_codecs = NUM_CODEC; + caps->min_fragment_size = MIN_FRAGMENT_SIZE; /* 50KB */ + caps->max_fragment_size = MAX_FRAGMENT_SIZE; /* 1024KB */ + caps->min_fragments = MIN_FRAGMENT; + caps->max_fragments = MAX_FRAGMENT; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_AAC; + return 0; +} + +static struct snd_compr_codec_caps caps_mp3 = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates[0] = 48000, + .descriptor[0].sample_rates[1] = 44100, + .descriptor[0].sample_rates[2] = 32000, + .descriptor[0].sample_rates[3] = 16000, + .descriptor[0].sample_rates[4] = 8000, + .descriptor[0].num_sample_rates = 5, + .descriptor[0].bit_rate[0] = 320, + .descriptor[0].bit_rate[1] = 192, + .descriptor[0].num_bitrates = 2, + .descriptor[0].profiles = 0, + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, + .descriptor[0].formats = 0, +}; + +static struct snd_compr_codec_caps caps_aac = { + .num_descriptors = 2, + .descriptor[1].max_ch = 2, + .descriptor[0].sample_rates[0] = 48000, + .descriptor[0].sample_rates[1] = 44100, + .descriptor[0].sample_rates[2] = 32000, + .descriptor[0].sample_rates[3] = 16000, + .descriptor[0].sample_rates[4] = 8000, + .descriptor[0].num_sample_rates = 5, + .descriptor[1].bit_rate[0] = 320, + .descriptor[1].bit_rate[1] = 192, + .descriptor[1].num_bitrates = 2, + .descriptor[1].profiles = 0, + .descriptor[1].modes = 0, + .descriptor[1].formats = + (SND_AUDIOSTREAMFORMAT_MP4ADTS | + SND_AUDIOSTREAMFORMAT_RAW), +}; + +static int sst_cdev_codec_caps(struct snd_compr_codec_caps *codec) +{ + if (codec->codec == SND_AUDIOCODEC_MP3) + *codec = caps_mp3; + else if (codec->codec == SND_AUDIOCODEC_AAC) + *codec = caps_aac; + else + return -EINVAL; + + return 0; +} + +void sst_cdev_fragment_elapsed(struct intel_sst_drv *ctx, int str_id) +{ + struct stream_info *stream; + + dev_dbg(ctx->dev, "fragment elapsed from firmware for str_id %d\n", + str_id); + stream = &ctx->streams[str_id]; + if (stream->compr_cb) + stream->compr_cb(stream->compr_cb_param); +} + /* * sst_close_pcm_stream - Close PCM interface * @@ -372,10 +639,28 @@ static struct sst_ops pcm_ops = { .power = sst_power_control, }; +static struct compress_sst_ops compr_ops = { + .open = sst_cdev_open, + .close = sst_cdev_close, + .stream_pause = sst_cdev_stream_pause, + .stream_pause_release = sst_cdev_stream_pause_release, + .stream_start = sst_cdev_stream_start, + .stream_drop = sst_cdev_stream_drop, + .stream_drain = sst_cdev_stream_drain, + .stream_partial_drain = sst_cdev_stream_partial_drain, + .tstamp = sst_cdev_tstamp, + .ack = sst_cdev_ack, + .get_caps = sst_cdev_caps, + .get_codec_caps = sst_cdev_codec_caps, + .set_metadata = sst_cdev_set_metadata, + .power = sst_power_control, +}; + static struct sst_device sst_dsp_device = { .name = "Intel(R) SST LPE", .dev = NULL, .ops = &pcm_ops, + .compr_ops = &compr_ops, }; /* -- cgit v1.1 From e3a4bd5eec52912108e287146052f2624acbec7a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Oct 2014 14:45:04 +0100 Subject: ALSA: pcm: Simplify snd_pcm_action_lock_irq() The function snd_pcm_action_lock_irq() can be much simplified by simply wrapping snd_pcm_action() with the stream lock. This was rather the original idea, but later it was open coded for optimization. However, looking at the optimization part closely, one notices that the probability of the optimized path is quite low; in normal situations, the linked stream action happens only for the triggered substream, thus the operation becomes identical. So the code simplification has a clear win, especially because we have now doubly codes for both atomic and non-atomic locks. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 41 +++-------------------------------------- 1 file changed, 3 insertions(+), 38 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4d5795d..b92b605 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -947,28 +947,6 @@ static int snd_pcm_action(struct action_ops *ops, return res; } -static int snd_pcm_action_lock_mutex(struct action_ops *ops, - struct snd_pcm_substream *substream, - int state) -{ - int res; - - down_read(&snd_pcm_link_rwsem); - if (snd_pcm_stream_linked(substream)) { - mutex_lock(&substream->group->mutex); - mutex_lock(&substream->self_group.mutex); - res = snd_pcm_action_group(ops, substream, state, 1); - mutex_unlock(&substream->self_group.mutex); - mutex_unlock(&substream->group->mutex); - } else { - mutex_lock(&substream->self_group.mutex); - res = snd_pcm_action_single(ops, substream, state); - mutex_unlock(&substream->self_group.mutex); - } - up_read(&snd_pcm_link_rwsem); - return res; -} - /* * Note: don't use any locks before */ @@ -978,22 +956,9 @@ static int snd_pcm_action_lock_irq(struct action_ops *ops, { int res; - if (substream->pcm->nonatomic) - return snd_pcm_action_lock_mutex(ops, substream, state); - - read_lock_irq(&snd_pcm_link_rwlock); - if (snd_pcm_stream_linked(substream)) { - spin_lock(&substream->group->lock); - spin_lock(&substream->self_group.lock); - res = snd_pcm_action_group(ops, substream, state, 1); - spin_unlock(&substream->self_group.lock); - spin_unlock(&substream->group->lock); - } else { - spin_lock(&substream->self_group.lock); - res = snd_pcm_action_single(ops, substream, state); - spin_unlock(&substream->self_group.lock); - } - read_unlock_irq(&snd_pcm_link_rwlock); + snd_pcm_stream_lock_irq(substream); + res = snd_pcm_action(ops, substream, state); + snd_pcm_stream_unlock_irq(substream); return res; } -- cgit v1.1 From aa8edd8ca6a110349b00e360823c64e8cd106289 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Oct 2014 15:19:36 +0100 Subject: ALSA: pcm: Refactoring snd_pcm_action() Just a small code refactoring to reduce more lines. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 35 ++++++++++------------------------- 1 file changed, 10 insertions(+), 25 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b92b605..ca224fa 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -900,14 +900,19 @@ static int snd_pcm_action_single(struct action_ops *ops, return res; } -/* call in mutex-protected context */ -static int snd_pcm_action_mutex(struct action_ops *ops, - struct snd_pcm_substream *substream, - int state) +/* + * Note: call with stream lock + */ +static int snd_pcm_action(struct action_ops *ops, + struct snd_pcm_substream *substream, + int state) { int res; - if (snd_pcm_stream_linked(substream)) { + if (!snd_pcm_stream_linked(substream)) + return snd_pcm_action_single(ops, substream, state); + + if (substream->pcm->nonatomic) { if (!mutex_trylock(&substream->group->mutex)) { mutex_unlock(&substream->self_group.mutex); mutex_lock(&substream->group->mutex); @@ -916,24 +921,6 @@ static int snd_pcm_action_mutex(struct action_ops *ops, res = snd_pcm_action_group(ops, substream, state, 1); mutex_unlock(&substream->group->mutex); } else { - res = snd_pcm_action_single(ops, substream, state); - } - return res; -} - -/* - * Note: call with stream lock - */ -static int snd_pcm_action(struct action_ops *ops, - struct snd_pcm_substream *substream, - int state) -{ - int res; - - if (substream->pcm->nonatomic) - return snd_pcm_action_mutex(ops, substream, state); - - if (snd_pcm_stream_linked(substream)) { if (!spin_trylock(&substream->group->lock)) { spin_unlock(&substream->self_group.lock); spin_lock(&substream->group->lock); @@ -941,8 +928,6 @@ static int snd_pcm_action(struct action_ops *ops, } res = snd_pcm_action_group(ops, substream, state, 1); spin_unlock(&substream->group->lock); - } else { - res = snd_pcm_action_single(ops, substream, state); } return res; } -- cgit v1.1 From 3172fcddcea230f129e8916628672617ef3c836c Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:44 +0530 Subject: ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data Some debug prints use dev context in sst_data. Store the device context for the same. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index e7cf18d..6032f18 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -706,6 +706,7 @@ static int sst_platform_probe(struct platform_device *pdev) pdata->pdev_strm_map = dpcm_strm_map; pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); drv->pdata = pdata; + drv->pdev = pdev; mutex_init(&drv->lock); dev_set_drvdata(&pdev->dev, drv); -- cgit v1.1 From 7e73e4d80539d0392010dfac3116307e7c9cf33d Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:45 +0530 Subject: ASoC: Intel: move the driver wq init to a routine This will be used by ACPI code as well, so moving to common routine helps Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 34 +++++++++++++++++++--------------- 1 file changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index fdada40..f9a6d6d 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -206,6 +206,22 @@ void sst_process_pending_msg(struct work_struct *work) ctx->ops->post_message(ctx, NULL, false); } +static int sst_workqueue_init(struct intel_sst_drv *ctx) +{ + INIT_LIST_HEAD(&ctx->memcpy_list); + INIT_LIST_HEAD(&ctx->rx_list); + INIT_LIST_HEAD(&ctx->ipc_dispatch_list); + INIT_LIST_HEAD(&ctx->block_list); + INIT_WORK(&ctx->ipc_post_msg_wq, sst_process_pending_msg); + init_waitqueue_head(&ctx->wait_queue); + + ctx->post_msg_wq = + create_singlethread_workqueue("sst_post_msg_wq"); + if (!ctx->post_msg_wq) + return -EBUSY; + return 0; +} + /* * intel_sst_probe - PCI probe function * @@ -254,24 +270,13 @@ static int intel_sst_probe(struct pci_dev *pci, sst_drv_ctx->use_dma = 0; sst_drv_ctx->use_lli = 0; - INIT_LIST_HEAD(&sst_drv_ctx->memcpy_list); - INIT_LIST_HEAD(&sst_drv_ctx->ipc_dispatch_list); - INIT_LIST_HEAD(&sst_drv_ctx->block_list); - INIT_LIST_HEAD(&sst_drv_ctx->rx_list); - - sst_drv_ctx->post_msg_wq = - create_singlethread_workqueue("sst_post_msg_wq"); - if (!sst_drv_ctx->post_msg_wq) { - ret = -EINVAL; - goto do_free_drv_ctx; - } - INIT_WORK(&sst_drv_ctx->ipc_post_msg_wq, sst_process_pending_msg); - init_waitqueue_head(&sst_drv_ctx->wait_queue); - spin_lock_init(&sst_drv_ctx->ipc_spin_lock); spin_lock_init(&sst_drv_ctx->block_lock); spin_lock_init(&sst_drv_ctx->rx_msg_lock); + if (sst_workqueue_init(sst_drv_ctx)) + return -EINVAL; + dev_info(sst_drv_ctx->dev, "Got drv data max stream %d\n", sst_drv_ctx->info.max_streams); for (i = 1; i <= sst_drv_ctx->info.max_streams; i++) { @@ -414,7 +419,6 @@ do_release_regions: pci_release_regions(pci); do_free_mem: destroy_workqueue(sst_drv_ctx->post_msg_wq); -do_free_drv_ctx: dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); return ret; } -- cgit v1.1 From 54adc0ad647792b3a8557520477a40f76d99a007 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:46 +0530 Subject: ASoC: Intel: move the lock and wq initialization to routine This will be used by ACPI code as well, so moving to common routine helps Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index f9a6d6d..0863471 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -222,6 +222,14 @@ static int sst_workqueue_init(struct intel_sst_drv *ctx) return 0; } +static void sst_init_locks(struct intel_sst_drv *ctx) +{ + mutex_init(&ctx->sst_lock); + spin_lock_init(&ctx->rx_msg_lock); + spin_lock_init(&ctx->ipc_spin_lock); + spin_lock_init(&ctx->block_lock); +} + /* * intel_sst_probe - PCI probe function * @@ -259,7 +267,7 @@ static int intel_sst_probe(struct pci_dev *pci, return -EINVAL; ops = sst_drv_ctx->ops; - mutex_init(&sst_drv_ctx->sst_lock); + sst_init_locks(sst_drv_ctx); /* pvt_id 0 reserved for async messages */ sst_drv_ctx->pvt_id = 1; @@ -270,10 +278,6 @@ static int intel_sst_probe(struct pci_dev *pci, sst_drv_ctx->use_dma = 0; sst_drv_ctx->use_lli = 0; - spin_lock_init(&sst_drv_ctx->ipc_spin_lock); - spin_lock_init(&sst_drv_ctx->block_lock); - spin_lock_init(&sst_drv_ctx->rx_msg_lock); - if (sst_workqueue_init(sst_drv_ctx)) return -EINVAL; -- cgit v1.1 From 2559d9928f36f3c0bfb4ded9bb47d47b36337b09 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:47 +0530 Subject: ASoC: Intel: move the driver context allocation to routine This will be used by ACPI code as well, so moving to common routine helps Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 0863471..55bb1f7 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -230,6 +230,20 @@ static void sst_init_locks(struct intel_sst_drv *ctx) spin_lock_init(&ctx->block_lock); } +int sst_alloc_drv_context(struct intel_sst_drv **ctx, + struct device *dev, unsigned int dev_id) +{ + *ctx = devm_kzalloc(dev, sizeof(struct intel_sst_drv), GFP_KERNEL); + if (!(*ctx)) + return -ENOMEM; + + (*ctx)->dev = dev; + (*ctx)->dev_id = dev_id; + + return 0; +} + + /* * intel_sst_probe - PCI probe function * @@ -247,12 +261,11 @@ static int intel_sst_probe(struct pci_dev *pci, int ddr_base; dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); - sst_drv_ctx = devm_kzalloc(&pci->dev, sizeof(*sst_drv_ctx), GFP_KERNEL); - if (!sst_drv_ctx) - return -ENOMEM; - sst_drv_ctx->dev = &pci->dev; - sst_drv_ctx->dev_id = pci->device; + ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); + if (ret < 0) + return ret; + if (!sst_pdata) return -EINVAL; -- cgit v1.1 From 250454d8fe65680b26f2917b806e2caf49126a01 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:48 +0530 Subject: ASoC: Intel: modularize driver probe and remove The driver probe which initializes driver and remove which cleans up can be shared with APCI as well, so move them to common init_context and cleanup_context routines which can be used by ACPI as well Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 165 ++++++++++++++++++++++++++-------------------- 1 file changed, 94 insertions(+), 71 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 55bb1f7..09d367a 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -243,6 +243,94 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, return 0; } +int sst_context_init(struct intel_sst_drv *ctx) +{ + int ret = 0, i; + + if (!ctx->pdata) + return -EINVAL; + + if (!ctx->pdata->probe_data) + return -EINVAL; + + memcpy(&ctx->info, ctx->pdata->probe_data, sizeof(ctx->info)); + + ret = sst_driver_ops(ctx); + if (ret != 0) + return -EINVAL; + + sst_init_locks(ctx); + + /* pvt_id 0 reserved for async messages */ + ctx->pvt_id = 1; + ctx->stream_cnt = 0; + ctx->fw_in_mem = NULL; + /* we use memcpy, so set to 0 */ + ctx->use_dma = 0; + ctx->use_lli = 0; + + if (sst_workqueue_init(ctx)) + return -EINVAL; + + ctx->mailbox_recv_offset = ctx->pdata->ipc_info->mbox_recv_off; + ctx->ipc_reg.ipcx = SST_IPCX + ctx->pdata->ipc_info->ipc_offset; + ctx->ipc_reg.ipcd = SST_IPCD + ctx->pdata->ipc_info->ipc_offset; + + dev_info(ctx->dev, "Got drv data max stream %d\n", + ctx->info.max_streams); + + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + memset(stream, 0, sizeof(*stream)); + stream->pipe_id = PIPE_RSVD; + mutex_init(&stream->lock); + } + + /* Register the ISR */ + ret = devm_request_threaded_irq(ctx->dev, ctx->irq_num, ctx->ops->interrupt, + ctx->ops->irq_thread, 0, SST_DRV_NAME, + ctx); + if (ret) + goto do_free_mem; + + dev_dbg(ctx->dev, "Registered IRQ %#x\n", ctx->irq_num); + + /* default intr are unmasked so set this as masked */ + sst_shim_write64(ctx->shim, SST_IMRX, 0xFFFF0038); + + ctx->qos = devm_kzalloc(ctx->dev, + sizeof(struct pm_qos_request), GFP_KERNEL); + if (!ctx->qos) { + ret = -ENOMEM; + goto do_free_mem; + } + pm_qos_add_request(ctx->qos, PM_QOS_CPU_DMA_LATENCY, + PM_QOS_DEFAULT_VALUE); + return 0; + +do_free_mem: + destroy_workqueue(ctx->post_msg_wq); + return ret; +} + +void sst_context_cleanup(struct intel_sst_drv *ctx) +{ + pm_runtime_get_noresume(ctx->dev); + pm_runtime_forbid(ctx->dev); + sst_unregister(ctx->dev); + sst_set_fw_state_locked(ctx, SST_SHUTDOWN); + flush_scheduled_work(); + destroy_workqueue(ctx->post_msg_wq); + pm_qos_remove_request(ctx->qos); + kfree(ctx->fw_sg_list.src); + kfree(ctx->fw_sg_list.dst); + ctx->fw_sg_list.list_len = 0; + kfree(ctx->fw_in_mem); + ctx->fw_in_mem = NULL; + sst_memcpy_free_resources(ctx); + ctx = NULL; +} /* * intel_sst_probe - PCI probe function @@ -254,9 +342,8 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, static int intel_sst_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - int i, ret = 0; + int ret = 0; struct intel_sst_drv *sst_drv_ctx; - struct intel_sst_ops *ops; struct sst_platform_info *sst_pdata = pci->dev.platform_data; int ddr_base; @@ -266,43 +353,12 @@ static int intel_sst_probe(struct pci_dev *pci, if (ret < 0) return ret; - if (!sst_pdata) - return -EINVAL; - sst_drv_ctx->pdata = sst_pdata; - if (!sst_drv_ctx->pdata->probe_data) - return -EINVAL; - - memcpy(&sst_drv_ctx->info, sst_drv_ctx->pdata->probe_data, - sizeof(sst_drv_ctx->info)); - - if (0 != sst_driver_ops(sst_drv_ctx)) - return -EINVAL; - - ops = sst_drv_ctx->ops; - sst_init_locks(sst_drv_ctx); - - /* pvt_id 0 reserved for async messages */ - sst_drv_ctx->pvt_id = 1; - sst_drv_ctx->stream_cnt = 0; - sst_drv_ctx->fw_in_mem = NULL; - - /* we use memcpy, so set to 0 */ - sst_drv_ctx->use_dma = 0; - sst_drv_ctx->use_lli = 0; - - if (sst_workqueue_init(sst_drv_ctx)) - return -EINVAL; - dev_info(sst_drv_ctx->dev, "Got drv data max stream %d\n", - sst_drv_ctx->info.max_streams); - for (i = 1; i <= sst_drv_ctx->info.max_streams; i++) { - struct stream_info *stream = &sst_drv_ctx->streams[i]; + ret = sst_context_init(sst_drv_ctx); + if (ret < 0) + goto do_free_drv_ctx; - memset(stream, 0, sizeof(*stream)); - stream->pipe_id = PIPE_RSVD; - mutex_init(&stream->lock); - } /* Init the device */ ret = pcim_enable_device(pci); @@ -402,18 +458,6 @@ static int intel_sst_probe(struct pci_dev *pci, } sst_drv_ctx->irq_num = pci->irq; - /* Register the ISR */ - ret = devm_request_threaded_irq(&pci->dev, pci->irq, - sst_drv_ctx->ops->interrupt, - sst_drv_ctx->ops->irq_thread, 0, SST_DRV_NAME, - sst_drv_ctx); - if (ret) - goto do_release_regions; - dev_dbg(sst_drv_ctx->dev, "Registered IRQ 0x%x\n", pci->irq); - - /* default intr are unmasked so set this as masked */ - if (sst_drv_ctx->dev_id == SST_MRFLD_PCI_ID) - sst_shim_write64(sst_drv_ctx->shim, SST_IMRX, 0xFFFF0038); pci_set_drvdata(pci, sst_drv_ctx); pm_runtime_set_autosuspend_delay(sst_drv_ctx->dev, SST_SUSPEND_DELAY); @@ -421,14 +465,6 @@ static int intel_sst_probe(struct pci_dev *pci, pm_runtime_allow(sst_drv_ctx->dev); pm_runtime_put_noidle(sst_drv_ctx->dev); sst_register(sst_drv_ctx->dev); - sst_drv_ctx->qos = devm_kzalloc(&pci->dev, - sizeof(struct pm_qos_request), GFP_KERNEL); - if (!sst_drv_ctx->qos) { - ret = -ENOMEM; - goto do_release_regions; - } - pm_qos_add_request(sst_drv_ctx->qos, PM_QOS_CPU_DMA_LATENCY, - PM_QOS_DEFAULT_VALUE); return ret; @@ -436,6 +472,7 @@ do_release_regions: pci_release_regions(pci); do_free_mem: destroy_workqueue(sst_drv_ctx->post_msg_wq); +do_free_drv_ctx: dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); return ret; } @@ -452,22 +489,8 @@ static void intel_sst_remove(struct pci_dev *pci) { struct intel_sst_drv *sst_drv_ctx = pci_get_drvdata(pci); - pm_runtime_get_noresume(sst_drv_ctx->dev); - pm_runtime_forbid(sst_drv_ctx->dev); - sst_unregister(sst_drv_ctx->dev); + sst_context_cleanup(sst_drv_ctx); pci_dev_put(sst_drv_ctx->pci); - sst_set_fw_state_locked(sst_drv_ctx, SST_SHUTDOWN); - - flush_scheduled_work(); - destroy_workqueue(sst_drv_ctx->post_msg_wq); - pm_qos_remove_request(sst_drv_ctx->qos); - kfree(sst_drv_ctx->fw_sg_list.src); - kfree(sst_drv_ctx->fw_sg_list.dst); - sst_drv_ctx->fw_sg_list.list_len = 0; - kfree(sst_drv_ctx->fw_in_mem); - sst_drv_ctx->fw_in_mem = NULL; - sst_memcpy_free_resources(sst_drv_ctx); - sst_drv_ctx = NULL; pci_release_regions(pci); pci_set_drvdata(pci, NULL); } -- cgit v1.1 From 7fb73c74ffee65bda3b6d3b00c1b841f557a1191 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 30 Oct 2014 16:21:49 +0530 Subject: ASoC: Intel: more probe modularization for sst Move the PCI BAR and resource initialization to a separate routine Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 158 +++++++++++++++++++++++++--------------------- 1 file changed, 86 insertions(+), 72 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 09d367a..2bfb404 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -332,114 +332,132 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) ctx = NULL; } -/* -* intel_sst_probe - PCI probe function -* -* @pci: PCI device structure -* @pci_id: PCI device ID structure -* -*/ -static int intel_sst_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { - int ret = 0; - struct intel_sst_drv *sst_drv_ctx; - struct sst_platform_info *sst_pdata = pci->dev.platform_data; - int ddr_base; - - dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); - - ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); - if (ret < 0) - return ret; - - sst_drv_ctx->pdata = sst_pdata; - - ret = sst_context_init(sst_drv_ctx); - if (ret < 0) - goto do_free_drv_ctx; - + pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); + pm_runtime_use_autosuspend(ctx->dev); + pm_runtime_allow(ctx->dev); + pm_runtime_put_noidle(ctx->dev); +} - /* Init the device */ - ret = pcim_enable_device(pci); - if (ret) { - dev_err(sst_drv_ctx->dev, - "device can't be enabled. Returned err: %d\n", ret); - goto do_free_mem; - } - sst_drv_ctx->pci = pci_dev_get(pci); +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + int ddr_base, ret = 0; + struct pci_dev *pci = ctx->pci; ret = pci_request_regions(pci, SST_DRV_NAME); if (ret) - goto do_free_mem; + return ret; /* map registers */ /* DDR base */ - if (sst_drv_ctx->dev_id == SST_MRFLD_PCI_ID) { - sst_drv_ctx->ddr_base = pci_resource_start(pci, 0); + if (ctx->dev_id == SST_MRFLD_PCI_ID) { + ctx->ddr_base = pci_resource_start(pci, 0); /* check that the relocated IMR base matches with FW Binary */ - ddr_base = relocate_imr_addr_mrfld(sst_drv_ctx->ddr_base); - if (!sst_drv_ctx->pdata->lib_info) { - dev_err(sst_drv_ctx->dev, "lib_info pointer NULL\n"); + ddr_base = relocate_imr_addr_mrfld(ctx->ddr_base); + if (!ctx->pdata->lib_info) { + dev_err(ctx->dev, "lib_info pointer NULL\n"); ret = -EINVAL; goto do_release_regions; } - if (ddr_base != sst_drv_ctx->pdata->lib_info->mod_base) { - dev_err(sst_drv_ctx->dev, + if (ddr_base != ctx->pdata->lib_info->mod_base) { + dev_err(ctx->dev, "FW LSP DDR BASE does not match with IFWI\n"); ret = -EINVAL; goto do_release_regions; } - sst_drv_ctx->ddr_end = pci_resource_end(pci, 0); + ctx->ddr_end = pci_resource_end(pci, 0); - sst_drv_ctx->ddr = pcim_iomap(pci, 0, + ctx->ddr = pcim_iomap(pci, 0, pci_resource_len(pci, 0)); - if (!sst_drv_ctx->ddr) { + if (!ctx->ddr) { ret = -EINVAL; goto do_release_regions; } - dev_dbg(sst_drv_ctx->dev, "sst: DDR Ptr %p\n", sst_drv_ctx->ddr); + dev_dbg(ctx->dev, "sst: DDR Ptr %p\n", ctx->ddr); } else { - sst_drv_ctx->ddr = NULL; + ctx->ddr = NULL; } - /* SHIM */ - sst_drv_ctx->shim_phy_add = pci_resource_start(pci, 1); - sst_drv_ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); - if (!sst_drv_ctx->shim) { + ctx->shim_phy_add = pci_resource_start(pci, 1); + ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); + if (!ctx->shim) { ret = -EINVAL; goto do_release_regions; } - dev_dbg(sst_drv_ctx->dev, "SST Shim Ptr %p\n", sst_drv_ctx->shim); + dev_dbg(ctx->dev, "SST Shim Ptr %p\n", ctx->shim); /* Shared SRAM */ - sst_drv_ctx->mailbox_add = pci_resource_start(pci, 2); - sst_drv_ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); - if (!sst_drv_ctx->mailbox) { + ctx->mailbox_add = pci_resource_start(pci, 2); + ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); + if (!ctx->mailbox) { ret = -EINVAL; goto do_release_regions; } - dev_dbg(sst_drv_ctx->dev, "SRAM Ptr %p\n", sst_drv_ctx->mailbox); + dev_dbg(ctx->dev, "SRAM Ptr %p\n", ctx->mailbox); /* IRAM */ - sst_drv_ctx->iram_end = pci_resource_end(pci, 3); - sst_drv_ctx->iram_base = pci_resource_start(pci, 3); - sst_drv_ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); - if (!sst_drv_ctx->iram) { + ctx->iram_end = pci_resource_end(pci, 3); + ctx->iram_base = pci_resource_start(pci, 3); + ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); + if (!ctx->iram) { ret = -EINVAL; goto do_release_regions; } - dev_dbg(sst_drv_ctx->dev, "IRAM Ptr %p\n", sst_drv_ctx->iram); + dev_dbg(ctx->dev, "IRAM Ptr %p\n", ctx->iram); /* DRAM */ - sst_drv_ctx->dram_end = pci_resource_end(pci, 4); - sst_drv_ctx->dram_base = pci_resource_start(pci, 4); - sst_drv_ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); - if (!sst_drv_ctx->dram) { + ctx->dram_end = pci_resource_end(pci, 4); + ctx->dram_base = pci_resource_start(pci, 4); + ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); + if (!ctx->dram) { ret = -EINVAL; goto do_release_regions; } - dev_dbg(sst_drv_ctx->dev, "DRAM Ptr %p\n", sst_drv_ctx->dram); + dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); +do_release_regions: + pci_release_regions(pci); + return 0; +} +/* +* intel_sst_probe - PCI probe function +* +* @pci: PCI device structure +* @pci_id: PCI device ID structure +* +*/ +static int intel_sst_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret = 0; + struct intel_sst_drv *sst_drv_ctx; + struct sst_platform_info *sst_pdata = pci->dev.platform_data; + + dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); + + ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); + if (ret < 0) + return ret; + + sst_drv_ctx->pdata = sst_pdata; + sst_drv_ctx->irq_num = pci->irq; + + ret = sst_context_init(sst_drv_ctx); + if (ret < 0) + goto do_free_drv_ctx; + + + /* Init the device */ + ret = pcim_enable_device(pci); + if (ret) { + dev_err(sst_drv_ctx->dev, + "device can't be enabled. Returned err: %d\n", ret); + goto do_destroy_wq; + } + sst_drv_ctx->pci = pci_dev_get(pci); + ret = sst_platform_get_resources(sst_drv_ctx); + if (ret < 0) + goto do_destroy_wq; sst_set_fw_state_locked(sst_drv_ctx, SST_RESET); snprintf(sst_drv_ctx->firmware_name, sizeof(sst_drv_ctx->firmware_name), @@ -457,20 +475,16 @@ static int intel_sst_probe(struct pci_dev *pci, goto do_release_regions; } - sst_drv_ctx->irq_num = pci->irq; pci_set_drvdata(pci, sst_drv_ctx); - pm_runtime_set_autosuspend_delay(sst_drv_ctx->dev, SST_SUSPEND_DELAY); - pm_runtime_use_autosuspend(sst_drv_ctx->dev); - pm_runtime_allow(sst_drv_ctx->dev); - pm_runtime_put_noidle(sst_drv_ctx->dev); + sst_configure_runtime_pm(sst_drv_ctx); sst_register(sst_drv_ctx->dev); return ret; do_release_regions: pci_release_regions(pci); -do_free_mem: +do_destroy_wq: destroy_workqueue(sst_drv_ctx->post_msg_wq); do_free_drv_ctx: dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); -- cgit v1.1 From c1e99c913be4294e63b5e74b197b8a8c86e6e67b Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 30 Oct 2014 21:16:23 +0800 Subject: ASoC: Intel: Add jack detection for Broadwell Add jack dectection and event reporting for Broadwell. It use combo jack on BDW platform, which including Mic Jack pin and Headphone jack pin. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 49 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 46 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 0e550f1..52cb764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include "sst-dsp.h" @@ -26,8 +27,26 @@ #include "../codecs/rt286.h" +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_MIC("DMIC1", NULL), @@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"Speaker", NULL, "SPOL"}, /* HP jack connectors - unknown if we have jack deteck */ - {"Headphones", NULL, "HPO Pin"}, + {"Headphone Jack", NULL, "HPO Pin"}, /* other jacks */ {"MIC1", NULL, "Mic Jack"}, @@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); + + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&broadwell_headset, + ARRAY_SIZE(broadwell_headset_pins), + broadwell_headset_pins); + if (ret) + return ret; + + rt286_mic_detect(codec, &broadwell_headset); + return 0; +} + + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) } /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line Jack"); @@ -196,6 +236,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .no_pcm = 1, .codec_name = "i2c-INT343A:00", .codec_dai_name = "rt286-aif1", + .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_suspend = 1, @@ -213,6 +254,8 @@ static struct snd_soc_card broadwell_rt286 = { .owner = THIS_MODULE, .dai_link = broadwell_rt286_dais, .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), .dapm_widgets = broadwell_widgets, .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), .dapm_routes = broadwell_rt286_map, -- cgit v1.1 From 16af0ee16ca9391ef82e1c74c362d80551e769fe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:00:58 +0100 Subject: ASoC: ad1980: Remove unused header The constants defined in the ad1980 header are not used. So remove the file. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1980.c | 2 -- sound/soc/codecs/ad1980.c | 2 -- sound/soc/codecs/ad1980.h | 26 -------------------------- 3 files changed, 30 deletions(-) delete mode 100644 sound/soc/codecs/ad1980.h diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index 3450e8f..0fa81a5 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -46,8 +46,6 @@ #include #include -#include "../codecs/ad1980.h" - #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 304d300..cc28dba 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -30,8 +30,6 @@ #include #include -#include "ad1980.h" - /* * AD1980 register cache */ diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h deleted file mode 100644 index eb0af44..0000000 --- a/sound/soc/codecs/ad1980.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * ad1980.h -- ad1980 Soc Audio driver - * - * WARNING: - * - * Because Analog Devices Inc. discontinued the ad1980 sound chip since - * Sep. 2009, this ad1980 driver is not maintained, tested and supported - * by ADI now. - */ - -#ifndef _AD1980_H -#define _AD1980_H -/* Bit definition of Power-Down Control/Status Register */ -#define ADC 0x0001 -#define DAC 0x0002 -#define ANL 0x0004 -#define REF 0x0008 -#define PR0 0x0100 -#define PR1 0x0200 -#define PR2 0x0400 -#define PR3 0x0800 -#define PR4 0x1000 -#define PR5 0x2000 -#define PR6 0x4000 - -#endif -- cgit v1.1 From e5adb6cddb17f8e76be404f23a2e0db102ee1bd1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:00:59 +0100 Subject: ASoC: ad1980: Cleanup printk usage Use dev_err()/dev_warn() instead of printk(KERN_ERR/KERN_WARNING. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Also remove the info message that is printed when the driver is probed, this is just noise in bootlog. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index cc28dba..5f076c2 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -209,7 +209,8 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) return 0; } while (retry_cnt++ < 10); - printk(KERN_ERR "AD1980 AC97 reset failed\n"); + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); + return -EIO; } @@ -219,19 +220,15 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) u16 vendor_id2; u16 ext_status; - printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { - printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); + dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } ret = ad1980_reset(codec, 0); - if (ret < 0) { - printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n"); + if (ret < 0) goto reset_err; - } /* Read out vendor ID to make sure it is ad1980 */ if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) { @@ -246,9 +243,8 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) ret = -ENODEV; goto reset_err; } else { - printk(KERN_WARNING "ad1980: " - "Found AD1981 - only 2/2 IN/OUT Channels " - "supported\n"); + dev_warn(codec->dev, + "Found AD1981 - only 2/2 IN/OUT Channels supported\n"); } } -- cgit v1.1 From 6ce13d61dc6cfc3cf6be6bd12faf75bfbc12ea91 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:00 +0100 Subject: ASoC: ad1980: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 5f076c2..9ed4e12 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -259,9 +259,6 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls, - ARRAY_SIZE(ad1980_snd_ac97_controls)); - return 0; reset_err: @@ -285,6 +282,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .write = ac97_write, .read = ac97_read, + .controls = ad1980_snd_ac97_controls, + .num_controls = ARRAY_SIZE(ad1980_snd_ac97_controls), .dapm_widgets = ad1980_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), .dapm_routes = ad1980_dapm_routes, -- cgit v1.1 From 93932abaa3c84c2d76ce713bbbad08bad9162483 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:01 +0100 Subject: ASoC: stac9766: Cleanup printk usage Use dev_err() instead of printk(KERN_ERR. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 9878534..e88d9ac 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -262,7 +262,7 @@ static int stac9766_codec_resume(struct snd_soc_codec *codec) /* give the codec an AC97 warm reset to start the link */ reset: if (reset > 5) { - printk(KERN_ERR "stac9766 failed to resume"); + dev_err(codec->dev, "Failed to resume\n"); return -EIO; } codec->ac97->bus->ops->warm_reset(codec->ac97); @@ -338,7 +338,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) stac9766_reset(codec, 0); ret = stac9766_reset(codec, 1); if (ret < 0) { - printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); goto codec_err; } -- cgit v1.1 From 8865051d9941de905432f59f7a88662e824d5df9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:02 +0100 Subject: ASoC: stac9766: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e88d9ac..6c62d29 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -342,9 +342,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) goto codec_err; } - snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, - ARRAY_SIZE(stac9766_snd_ac97_controls)); - return 0; codec_err: @@ -359,6 +356,8 @@ static int stac9766_codec_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_stac9766 = { + .controls = stac9766_snd_ac97_controls, + .num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls), .write = stac9766_ac97_write, .read = stac9766_ac97_read, .set_bias_level = stac9766_set_bias_level, -- cgit v1.1 From 9cf766f666cc4518e22f185159f285f4e3183230 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:03 +0100 Subject: ASoC: wm9705: Cleanup printk usage Use dev_err() instead of printk(KERN_ERR. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index c0b7f45..355b28d 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -300,6 +300,8 @@ static int wm9705_reset(struct snd_soc_codec *codec) return 0; /* Success */ } + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); + return -EIO; } @@ -317,10 +319,8 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; ret = wm9705_reset(codec); - if (ret < 0) { - printk(KERN_ERR "could not reset AC97 codec\n"); + if (ret < 0) return ret; - } for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); @@ -339,7 +339,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { - printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); + dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } -- cgit v1.1 From d7cabb08ba23c87757fb3be01e82f755aad426d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:04 +0100 Subject: ASoC: wm9705: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 355b28d..1650195 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -347,9 +347,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) if (ret) goto reset_err; - snd_soc_add_codec_controls(codec, wm9705_snd_ac97_controls, - ARRAY_SIZE(wm9705_snd_ac97_controls)); - return 0; reset_err: @@ -374,6 +371,9 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9705 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9705_reg, + + .controls = wm9705_snd_ac97_controls, + .num_controls = ARRAY_SIZE(wm9705_snd_ac97_controls), .dapm_widgets = wm9705_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9705_dapm_widgets), .dapm_routes = wm9705_audio_map, -- cgit v1.1 From 12ced338ab8858d11ef5b11b65c3dc612d9551c9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:05 +0100 Subject: ASoC: wm9712: Cleanup printk usage Use dev_err() instead of printk(KERN_ERR. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Also avoid printing two error messages when the reset fails. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index c5eb746..c389e56 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -595,7 +595,7 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) return 0; err: - printk(KERN_ERR "WM9712 AC97 reset failed\n"); + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); return -EIO; } @@ -611,10 +611,8 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; ret = wm9712_reset(codec, 1); - if (ret < 0) { - printk(KERN_ERR "could not reset AC97 codec\n"); + if (ret < 0) return ret; - } wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -637,15 +635,13 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) { - printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); + dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } ret = wm9712_reset(codec, 0); - if (ret < 0) { - printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n"); + if (ret < 0) goto reset_err; - } /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); -- cgit v1.1 From 9a812c6b7a2092e20b4b78ed0ec6614a89e96dfd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:06 +0100 Subject: ASoC: wm9712: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index c389e56..f3aab6e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -647,8 +647,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, wm9712_snd_ac97_controls, - ARRAY_SIZE(wm9712_snd_ac97_controls)); return 0; @@ -675,6 +673,9 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9712_reg, + + .controls = wm9712_snd_ac97_controls, + .num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls), .dapm_widgets = wm9712_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets), .dapm_routes = wm9712_audio_map, -- cgit v1.1 From a6c2b07f11beaf5719f03c70a9c9597534b297a5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:07 +0100 Subject: ASoC: wm9713: Cleanup printk usage Use dev_err()/dev_warn() instead of printk(KERN_ERR/KERN_WARNING. This is common practice and makes it easy to find out which device generated the message. While we are at it also align the error messages with the other AC'97 drivers. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index bddee30..38e17d45 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -689,7 +689,8 @@ struct _pll_div { * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 22) * 10) -static void pll_factors(struct _pll_div *pll_div, unsigned int source) +static void pll_factors(struct snd_soc_codec *codec, + struct _pll_div *pll_div, unsigned int source) { u64 Kpart; unsigned int K, Ndiv, Nmod, target; @@ -724,7 +725,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source) Ndiv = target / source; if ((Ndiv < 5) || (Ndiv > 12)) - printk(KERN_WARNING + dev_warn(codec->dev, "WM9713 PLL N value %u out of recommended range!\n", Ndiv); @@ -768,7 +769,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } - pll_factors(&pll_div, freq_in); + pll_factors(codec, &pll_div, freq_in); if (pll_div.k == 0) { reg = (pll_div.n << 12) | (pll_div.lf << 11) | @@ -1104,8 +1105,11 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) soc_ac97_ops->reset(codec->ac97); if (soc_ac97_ops->warm_reset) soc_ac97_ops->warm_reset(codec->ac97); - if (ac97_read(codec, 0) != wm9713_reg[0]) + if (ac97_read(codec, 0) != wm9713_reg[0]) { + dev_err(codec->dev, "Failed to reset: AC97 link error\n"); return -EIO; + } + return 0; } EXPORT_SYMBOL_GPL(wm9713_reset); @@ -1163,10 +1167,8 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) u16 *cache = codec->reg_cache; ret = wm9713_reset(codec, 1); - if (ret < 0) { - printk(KERN_ERR "could not reset AC97 codec\n"); + if (ret < 0) return ret; - } wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1205,10 +1207,8 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) * a warm reset followed by an optional cold reset for codec */ wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); - if (ret < 0) { - printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n"); + if (ret < 0) goto reset_err; - } wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.1 From c1359ca303ee5125827c0d2a65f0c86d491dc993 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:08 +0100 Subject: ASoC: wm9713: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 38e17d45..ba8c276 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1216,9 +1216,6 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - snd_soc_add_codec_controls(codec, wm9713_snd_ac97_controls, - ARRAY_SIZE(wm9713_snd_ac97_controls)); - return 0; reset_err: @@ -1248,6 +1245,9 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9713_reg, + + .controls = wm9713_snd_ac97_controls, + .num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls), .dapm_widgets = wm9713_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9713_dapm_widgets), .dapm_routes = wm9713_audio_map, -- cgit v1.1 From 5efe89d9525f24f607079307d2d9510e30ba8590 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:09 +0100 Subject: ASoC: wm9713: Move driver state struct allocation to driver probe Resources for the device should be allocated in the device driver probe callback, rather than in the ASoC CODEC probe callback. E.g. one advantage is that we can use device managed allocations. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ba8c276..2704783 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1191,17 +1191,11 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { - struct wm9713_priv *wm9713; int ret = 0, reg; - wm9713 = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL); - if (wm9713 == NULL) - return -ENOMEM; - snd_soc_codec_set_drvdata(codec, wm9713); - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); if (ret < 0) - goto codec_err; + return ret; /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ @@ -1220,16 +1214,12 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) reset_err: snd_soc_free_ac97_codec(codec); -codec_err: - kfree(wm9713); return ret; } static int wm9713_soc_remove(struct snd_soc_codec *codec) { - struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); snd_soc_free_ac97_codec(codec); - kfree(wm9713); return 0; } @@ -1256,6 +1246,14 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { static int wm9713_probe(struct platform_device *pdev) { + struct wm9713_priv *wm9713; + + wm9713 = devm_kzalloc(&pdev->dev, sizeof(*wm9713), GFP_KERNEL); + if (wm9713 == NULL) + return -ENOMEM; + + platform_set_drvdata(pdev, wm9713); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm9713, wm9713_dai, ARRAY_SIZE(wm9713_dai)); } -- cgit v1.1 From 5bc39b50fd3f9e3585e0cb1cf7d7da979a063848 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 30 Oct 2014 21:01:10 +0100 Subject: ASoC: wm9713: Use virtual control instead of virtual register The wm9713 currently implements the virtual control for the Mic B Source MUX using a virtual register. Replace this by using SOC_ENUM_SINGLE_VIRT(). Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2704783..ac13fc8 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -59,13 +59,12 @@ static const u16 wm9713_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0006, 0x0001, 0x0000, 0x574d, 0x4c13, - 0x0000, 0x0000, 0x0000 + 0x0000, 0x0000 }; /* virtual HP mixers regs */ #define HPL_MIXER 0x80 #define HPR_MIXER 0x82 -#define MICB_MUX 0x82 static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; @@ -110,7 +109,7 @@ SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ -SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ +SOC_ENUM_SINGLE_VIRT(2, wm9713_micb_select), /* mic selection 19 */ }; static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0); -- cgit v1.1 From e894beb8183dd9e3834983440900ceb632823676 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Fri, 31 Oct 2014 10:54:23 -0700 Subject: ASoC: cs42l51: depends on I2C Fix build errors when CONFIG_I2C is not enabled by making the driver depend on I2C. ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(cs42l51_i2c_driver); ^ ../sound/soc/codecs/cs42l51-i2c.c:55:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/codecs/cs42l51-i2c.c:45:26: warning: 'cs42l51_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver cs42l51_i2c_driver = { ^ Signed-off-by: Randy Dunlap Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f4fb12f..02a36b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -337,6 +337,7 @@ config SND_SOC_CS42L51 config SND_SOC_CS42L51_I2C tristate "Cirrus Logic CS42L51 CODEC (I2C)" + depends on I2C select SND_SOC_CS42L51 config SND_SOC_CS42L52 -- cgit v1.1 From 22a236b4d07b5c5cfdc5db9e87d479d32281cfe6 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Sun, 2 Nov 2014 12:04:41 +0530 Subject: ASoC: Intel: fix missing mutex on error in block prepare, we were returning the error code while still holding the mutex. We are releasing the mutex in this patch before return. Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index c451398..4a5bde9 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -1120,6 +1120,7 @@ int sst_block_alloc_scratch(struct sst_dsp *dsp) ret = block_list_prepare(dsp, &dsp->scratch_block_list); if (ret < 0) { dev_err(dsp->dev, "error: scratch block prepare failed\n"); + mutex_unlock(&dsp->mutex); return ret; } -- cgit v1.1 From bf9706fe958469e7dfc6a9e16d9240892f055e62 Mon Sep 17 00:00:00 2001 From: Max Filippov Date: Mon, 3 Nov 2014 13:10:53 +0300 Subject: ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER This fixes build errors in configurations with I2C/SPI master disabled. Reported-by: Fengguang Wu Signed-off-by: Max Filippov Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7881b3c..1e7a417 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -582,10 +582,12 @@ config SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_I2C tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" + depends on I2C select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI_MASTER select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- cgit v1.1 From 9ce63dbd5d2671a209a859ee90f7dc6b8f22f28e Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 1 Nov 2014 10:58:18 +0800 Subject: ASoC: rockchip: i2s: add text after tristate for SND_SOC_ROCKCHIP_I2S For SND_SOC_ROCKCHIP_I2S, adding some text after the tristate to make this directly user selectable. Signed-off-by: Jianqun Signed-off-by: Mark Brown --- sound/soc/rockchip/Kconfig | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 78fc159..b1fc0ca 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -8,4 +8,9 @@ config SND_SOC_ROCKCHIP select the audio interfaces to support below. config SND_SOC_ROCKCHIP_I2S - tristate + tristate "Rockchip I2S Device Driver" + depends on CLKDEV_LOOKUP + help + Say Y or M if you want to add support for I2S driver for + Rockchip I2S device. The device supports upto maximum of + 8 channels each for play and record. -- cgit v1.1 From dd63a9c2952ed142c64fd68c1a74d0d6fcac586f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 3 Nov 2014 10:31:47 +0100 Subject: ASoC: Remove snd_soc_platform_driver suspend/resume callbacks Those are unused and new drivers should use device driver suspend/resume. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 10 ---------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ad47e96..edbb07b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -857,8 +857,6 @@ struct snd_soc_platform_driver { int (*probe)(struct snd_soc_platform *); int (*remove)(struct snd_soc_platform *); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); struct snd_soc_component_driver component_driver; /* pcm creation and destruction */ @@ -891,8 +889,6 @@ struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - struct list_head list; struct snd_soc_component component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a2b51ed..0509d72 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -592,17 +592,12 @@ int snd_soc_suspend(struct device *dev) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); - if (platform->driver->suspend && !platform->suspended) { - platform->driver->suspend(cpu_dai); - platform->suspended = 1; - } } /* close any waiting streams and save state */ @@ -775,17 +770,12 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) cpu_dai->driver->resume(cpu_dai); - if (platform->driver->resume && platform->suspended) { - platform->driver->resume(cpu_dai); - platform->suspended = 0; - } } if (card->resume_post) -- cgit v1.1 From 2a374b78f5c2b5f31d35f8a7cd004989d6936756 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 3 Nov 2014 10:31:48 +0100 Subject: ASoC: Remove platform field from snd_soc_dai Typically a DAI does not need direct access to the platform. Currently the only user of this field is in a platform driver where we have a more direct way of getting a pointer to the platform. This patch updates the driver to use the more direct way and then removes the platform field from the snd_soc_dai struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - sound/soc/soc-core.c | 2 -- sound/soc/txx9/txx9aclc.c | 2 +- 3 files changed, 1 insertion(+), 4 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e8b3080..45d0fa1 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -268,7 +268,6 @@ struct snd_soc_dai { unsigned int sample_bits; /* parent platform/codec */ - struct snd_soc_platform *platform; struct snd_soc_codec *codec; struct snd_soc_component *component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0509d72..e20bb65 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1309,7 +1309,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int i, ret; @@ -1317,7 +1316,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) card->name, num, order); /* config components */ - cpu_dai->platform = platform; cpu_dai->card = card; for (i = 0; i < rtd->num_codecs; i++) rtd->codec_dais[i]->card = card; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index cd71fd8..00b7e2d 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -292,7 +292,7 @@ static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; - struct platform_device *pdev = to_platform_device(dai->platform->dev); + struct platform_device *pdev = to_platform_device(rtd->platform->dev); struct txx9aclc_soc_device *dev; struct resource *r; int i; -- cgit v1.1 From 4476159f0b73e58e8c4d750ce03843d70c13994c Mon Sep 17 00:00:00 2001 From: Jianqun Date: Sat, 1 Nov 2014 11:22:18 +0800 Subject: ASoC: simple-card: add "invert" property for detect GPIOs Since hardware may invert detect GPIO of headphone or mic, add one property to support software invert. Signed-off-by: Jianqun Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index cac95d7..cd49d50 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,7 +29,9 @@ struct simple_card_data { } *dai_props; unsigned int mclk_fs; int gpio_hp_det; + int gpio_hp_det_invert; int gpio_mic_det; + int gpio_mic_det_invert; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -148,6 +150,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) simple_card_hp_jack_pins); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); } @@ -159,6 +162,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(simple_card_mic_jack_pins), simple_card_mic_jack_pins); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); } @@ -374,6 +378,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + enum of_gpio_flags flags; u32 val; int ret; @@ -429,13 +434,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } - priv->gpio_hp_det = of_get_named_gpio(node, - "simple-audio-card,hp-det-gpio", 0); + priv->gpio_hp_det = of_get_named_gpio_flags(node, + "simple-audio-card,hp-det-gpio", 0, &flags); + priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; - priv->gpio_mic_det = of_get_named_gpio(node, - "simple-audio-card,mic-det-gpio", 0); + priv->gpio_mic_det = of_get_named_gpio_flags(node, + "simple-audio-card,mic-det-gpio", 0, &flags); + priv->gpio_mic_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_mic_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- cgit v1.1 From eb58960e9ebf15cde8ca1248e15ecb0c8f3a28bd Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Fri, 24 Oct 2014 21:25:59 +0200 Subject: ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in half-duplex scenarios, which is needed when using the OSS API. Thus, only lock the divider if there is a stream in the other direction. Fixes the below program to not fail with the atmel ssc dai in master mode. int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22025; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..b1cc2a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) -- cgit v1.1 From e369086968157415aeb11af3b57cd998c6721603 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 3 Nov 2014 16:04:12 +0530 Subject: ALSA: echoaudio: add reference of struct echoaudio added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- cgit v1.1 From b5b4a41b392960010fccf1f9ccf8334d612bd450 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 3 Nov 2014 16:04:13 +0530 Subject: ALSA: echoaudio: remove all snd_printk removed all references of snd_printk with the standard dev_* macro. [a few places degraded to dev_dbg(), too -- tiwai] Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 7 +- sound/pci/echoaudio/darla24_dsp.c | 15 ++-- sound/pci/echoaudio/echo3g_dsp.c | 7 +- sound/pci/echoaudio/echoaudio.c | 137 ++++++++++++++++--------------- sound/pci/echoaudio/echoaudio.h | 28 ------- sound/pci/echoaudio/echoaudio_3g.c | 32 ++++---- sound/pci/echoaudio/echoaudio_dsp.c | 111 +++++++++++++++---------- sound/pci/echoaudio/echoaudio_gml.c | 11 +-- sound/pci/echoaudio/gina20_dsp.c | 15 ++-- sound/pci/echoaudio/gina24_dsp.c | 39 +++++---- sound/pci/echoaudio/indigo_dsp.c | 13 +-- sound/pci/echoaudio/indigo_express_dsp.c | 6 +- sound/pci/echoaudio/indigodj_dsp.c | 13 +-- sound/pci/echoaudio/indigodjx_dsp.c | 7 +- sound/pci/echoaudio/indigoio_dsp.c | 10 ++- sound/pci/echoaudio/indigoiox_dsp.c | 7 +- sound/pci/echoaudio/layla20_dsp.c | 37 +++++---- sound/pci/echoaudio/layla24_dsp.c | 34 ++++---- sound/pci/echoaudio/mia_dsp.c | 17 ++-- sound/pci/echoaudio/midi.c | 29 +++---- sound/pci/echoaudio/mona_dsp.c | 35 ++++---- 21 files changed, 327 insertions(+), 283 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index 20c7cbc..c94e92e 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,12 +33,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Darla20\n")); + dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw: could not initialize DSP comm page\n"); return err; } @@ -57,7 +58,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 6da6663..b1272f88 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,12 +33,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Darla24\n")); + dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw: could not initialize DSP comm page\n"); return err; } @@ -56,7 +57,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } @@ -128,15 +129,17 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) clock = GD24_8000; break; default: - DE_ACT(("set_sample_rate: Error, invalid sample rate %d\n", - rate)); + dev_err(chip->card->dev, + "set_sample_rate: Error, invalid sample rate %d\n", + rate); return -EINVAL; } if (wait_handshake(chip)) return -EIO; - DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); + dev_dbg(chip->card->dev, + "set_sample_rate: %d clock %d\n", rate, clock); chip->sample_rate = rate; /* Override the sample rate if this card is set to Echo sync. */ diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 3cdc2ee..bc37168 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,12 +46,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - DE_INIT(("init_hw() - Echo3G\n")); + dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -98,7 +99,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index db1b247..1ef29e5 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -48,13 +48,16 @@ static int get_firmware(const struct firmware **fw_entry, #ifdef CONFIG_PM_SLEEP if (chip->fw_cache[fw_index]) { - DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + dev_dbg(chip->card->dev, + "firmware requested: %s is cached\n", + card_fw[fw_index].data); *fw_entry = chip->fw_cache[fw_index]; return 0; } #endif - DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + dev_dbg(chip->card->dev, + "firmware requested: %s\n", card_fw[fw_index].data); snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); err = request_firmware(fw_entry, name, pci_device(chip)); if (err < 0) @@ -73,10 +76,10 @@ static void free_firmware(const struct firmware *fw_entry, struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP - DE_ACT(("firmware not released (kept in cache)\n")); + dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); #else release_firmware(fw_entry); - DE_ACT(("firmware released\n")); + dev_dbg(chip->card->dev, "firmware released\n"); #endif } @@ -90,10 +93,10 @@ static void free_firmware_cache(struct echoaudio *chip) for (i = 0; i < 8 ; i++) if (chip->fw_cache[i]) { release_firmware(chip->fw_cache[i]); - DE_ACT(("release_firmware(%d)\n", i)); + dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - DE_ACT(("firmware_cache released\n")); + dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -287,7 +290,7 @@ static int pcm_open(struct snd_pcm_substream *substream, /* Set up hw capabilities and contraints */ memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware)); - DE_HWP(("max_channels=%d\n", max_channels)); + dev_dbg(chip->card->dev, "max_channels=%d\n", max_channels); pipe->constr.list = channels_list; pipe->constr.mask = 0; for (i = 0; channels_list[i] <= max_channels; i++); @@ -337,7 +340,7 @@ static int pcm_open(struct snd_pcm_substream *substream, if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), PAGE_SIZE, &pipe->sgpage)) < 0) { - DE_HWP(("s-g list allocation failed\n")); + dev_err(chip->card->dev, "s-g list allocation failed\n"); return err; } @@ -351,7 +354,7 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - DE_ACT(("pcm_analog_in_open\n")); + dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -368,9 +371,9 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) atomic_inc(&chip->opencount); if (atomic_read(&chip->opencount) > 1 && chip->rate_set) chip->can_set_rate=0; - DE_HWP(("pcm_analog_in_open cs=%d oc=%d r=%d\n", + dev_dbg(chip->card->dev, "pcm_analog_in_open cs=%d oc=%d r=%d\n", chip->can_set_rate, atomic_read(&chip->opencount), - chip->sample_rate)); + chip->sample_rate); return 0; } @@ -386,7 +389,7 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - DE_ACT(("pcm_analog_out_open\n")); + dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -404,9 +407,9 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) atomic_inc(&chip->opencount); if (atomic_read(&chip->opencount) > 1 && chip->rate_set) chip->can_set_rate=0; - DE_HWP(("pcm_analog_out_open cs=%d oc=%d r=%d\n", + dev_dbg(chip->card->dev, "pcm_analog_out_open cs=%d oc=%d r=%d\n", chip->can_set_rate, atomic_read(&chip->opencount), - chip->sample_rate)); + chip->sample_rate); return 0; } @@ -419,7 +422,7 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - DE_ACT(("pcm_digital_in_open\n")); + dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -461,7 +464,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - DE_ACT(("pcm_digital_out_open\n")); + dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -508,18 +511,18 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - DE_ACT(("pcm_close\n")); + dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); - DE_ACT(("pcm_close oc=%d cs=%d rs=%d\n", oc, - chip->can_set_rate, chip->rate_set)); + dev_dbg(chip->card->dev, "pcm_close oc=%d cs=%d rs=%d\n", oc, + chip->can_set_rate, chip->rate_set); if (oc < 2) chip->can_set_rate = 1; if (oc == 0) chip->rate_set = 0; - DE_ACT(("pcm_close2 oc=%d cs=%d rs=%d\n", oc, - chip->can_set_rate,chip->rate_set)); + dev_dbg(chip->card->dev, "pcm_close2 oc=%d cs=%d rs=%d\n", oc, + chip->can_set_rate, chip->rate_set); return 0; } @@ -543,7 +546,7 @@ static int init_engine(struct snd_pcm_substream *substream, */ spin_lock_irq(&chip->lock); if (pipe->index >= 0) { - DE_HWP(("hwp_ie free(%d)\n", pipe->index)); + dev_dbg(chip->card->dev, "hwp_ie free(%d)\n", pipe->index); err = free_pipes(chip, pipe); snd_BUG_ON(err); chip->substream[pipe->index] = NULL; @@ -552,16 +555,17 @@ static int init_engine(struct snd_pcm_substream *substream, err = allocate_pipes(chip, pipe, pipe_index, interleave); if (err < 0) { spin_unlock_irq(&chip->lock); - DE_ACT((KERN_NOTICE "allocate_pipes(%d) err=%d\n", - pipe_index, err)); + dev_err(chip->card->dev, "allocate_pipes(%d) err=%d\n", + pipe_index, err); return err; } spin_unlock_irq(&chip->lock); - DE_ACT((KERN_NOTICE "allocate_pipes()=%d\n", pipe_index)); + dev_dbg(chip->card->dev, "allocate_pipes()=%d\n", pipe_index); - DE_HWP(("pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n", + dev_dbg(chip->card->dev, + "pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n", params_buffer_bytes(hw_params), params_periods(hw_params), - params_period_bytes(hw_params))); + params_period_bytes(hw_params)); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) { @@ -616,7 +620,7 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - DE_HWP(("pcm_hw_params ok\n")); + dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -680,14 +684,14 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); if (pipe->index >= 0) { - DE_HWP(("pcm_hw_free(%d)\n", pipe->index)); + dev_dbg(chip->card->dev, "pcm_hw_free(%d)\n", pipe->index); free_pipes(chip, pipe); chip->substream[pipe->index] = NULL; pipe->index = -1; } spin_unlock_irq(&chip->lock); - DE_HWP(("pcm_hw_freed\n")); + dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -701,8 +705,8 @@ static int pcm_prepare(struct snd_pcm_substream *substream) struct audioformat format; int pipe_index = ((struct audiopipe *)runtime->private_data)->index; - DE_HWP(("Prepare rate=%d format=%d channels=%d\n", - runtime->rate, runtime->format, runtime->channels)); + dev_dbg(chip->card->dev, "Prepare rate=%d format=%d channels=%d\n", + runtime->rate, runtime->format, runtime->channels); format.interleave = runtime->channels; format.data_are_bigendian = 0; format.mono_to_stereo = 0; @@ -722,8 +726,9 @@ static int pcm_prepare(struct snd_pcm_substream *substream) format.bits_per_sample = 32; break; default: - DE_HWP(("Prepare error: unsupported format %d\n", - runtime->format)); + dev_err(chip->card->dev, + "Prepare error: unsupported format %d\n", + runtime->format); return -EINVAL; } @@ -758,10 +763,10 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - DE_ACT(("pcm_trigger resume\n")); + dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - DE_ACT(("pcm_trigger start\n")); + dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -783,9 +788,9 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - DE_ACT(("pcm_trigger suspend\n")); + dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - DE_ACT(("pcm_trigger stop\n")); + dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -795,7 +800,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - DE_ACT(("pcm_trigger pause\n")); + dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -932,7 +937,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - DE_INIT(("Analog PCM ok\n")); + dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -945,7 +950,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - DE_INIT(("Digital PCM ok\n")); + dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -967,7 +972,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - DE_INIT(("Analog PCM ok\n")); + dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -982,7 +987,7 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - DE_INIT(("Digital PCM ok\n")); + dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1475,7 +1480,8 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol, snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->clock_src_ctl->id); - DE_ACT(("SDM() =%d\n", changed)); + dev_dbg(chip->card->dev, + "SDM() =%d\n", changed); } if (changed >= 0) changed = 1; /* No errors */ @@ -1602,7 +1608,8 @@ static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol, } if (changed < 0) - DE_ACT(("seticlk val%d err 0x%x\n", dclock, changed)); + dev_dbg(chip->card->dev, + "seticlk val%d err 0x%x\n", dclock, changed); return changed; } @@ -1859,7 +1866,7 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) #ifdef ECHOCARD_HAS_MIDI if (st > 0 && chip->midi_in) { snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st); - DE_MID(("rawmidi_iread=%d\n", st)); + dev_dbg(chip->card->dev, "rawmidi_iread=%d\n", st); } #endif return IRQ_HANDLED; @@ -1874,10 +1881,10 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - DE_INIT(("Stop DSP...\n")); + dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - DE_INIT(("Stopped.\n")); + dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1891,14 +1898,14 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - DE_INIT(("MMIO freed.\n")); + dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - DE_INIT(("Chip freed.\n")); + dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1908,7 +1915,7 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - DE_INIT(("snd_echo_dev_free()...\n")); + dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -1941,7 +1948,7 @@ static int snd_echo_create(struct snd_card *card, pci_disable_device(pci); return -ENOMEM; } - DE_INIT(("chip=%p\n", chip)); + dev_dbg(card->dev, "chip=%p\n", chip); spin_lock_init(&chip->lock); chip->card = card; chip->pci = pci; @@ -1978,8 +1985,8 @@ static int snd_echo_create(struct snd_card *card, return -EBUSY; } chip->irq = pci->irq; - DE_INIT(("pci=%p irq=%d subdev=%04x Init hardware...\n", - chip->pci, chip->irq, chip->pci->subsystem_device)); + dev_dbg(card->dev, "pci=%p irq=%d subdev=%04x Init hardware...\n", + chip->pci, chip->irq, chip->pci->subsystem_device); /* Create the DSP comm page - this is the area of memory used for most of the communication with the DSP, which accesses it via bus mastering */ @@ -1997,11 +2004,11 @@ static int snd_echo_create(struct snd_card *card, if (err >= 0) err = set_mixer_defaults(chip); if (err < 0) { - DE_INIT(("init_hw err=%d\n", err)); + dev_err(card->dev, "init_hw err=%d\n", err); snd_echo_free(chip); return err; } - DE_INIT(("Card init OK\n")); + dev_dbg(card->dev, "Card init OK\n"); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_echo_free(chip); @@ -2031,7 +2038,7 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - DE_INIT(("Echoaudio driver starting...\n")); + dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2184,7 +2191,7 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - DE_INIT(("suspend start\n")); + dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2211,7 +2218,7 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - DE_INIT(("suspend done\n")); + dev_dbg(dev, "suspend done\n"); return 0; } @@ -2225,7 +2232,7 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - DE_INIT(("resume start\n")); + dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2236,11 +2243,11 @@ static int snd_echo_resume(struct device *dev) err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { kfree(commpage_bak); - DE_INIT(("resume init_hw err=%d\n", err)); + dev_err(dev, "resume init_hw err=%d\n", err); snd_echo_free(chip); return err; } - DE_INIT(("resume init OK\n")); + dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2253,7 +2260,7 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - DE_INIT(("resume restore OK\n")); + dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2270,7 +2277,7 @@ static int snd_echo_resume(struct device *dev) return -EBUSY; } chip->irq = pci->irq; - DE_INIT(("resume irq=%d\n", chip->irq)); + dev_dbg(dev, "resume irq=%d\n", chip->irq); #ifdef ECHOCARD_HAS_MIDI if (chip->midi_input_enabled) @@ -2279,7 +2286,7 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - DE_INIT(("resume done\n")); + dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index a4f112a..3251522 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -295,34 +295,6 @@ #define PIPE_STATE_PENDING 3 /* Pipe has pending start */ -/* Debug initialization */ -#ifdef CONFIG_SND_DEBUG -#define DE_INIT(x) snd_printk x -#else -#define DE_INIT(x) -#endif - -/* Debug hw_params callbacks */ -#ifdef CONFIG_SND_DEBUG -#define DE_HWP(x) snd_printk x -#else -#define DE_HWP(x) -#endif - -/* Debug normal activity (open, start, stop...) */ -#ifdef CONFIG_SND_DEBUG -#define DE_ACT(x) snd_printk x -#else -#define DE_ACT(x) -#endif - -/* Debug midi activity */ -#ifdef CONFIG_SND_DEBUG -#define DE_MID(x) snd_printk x -#else -#define DE_MID(x) -#endif - struct audiopipe { volatile u32 *dma_counter; /* Commpage register that contains diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 658db44..50a2169 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -51,7 +51,7 @@ static int check_asic_status(struct echoaudio *chip) } box_status = le32_to_cpu(chip->comm_page->ext_box_status); - DE_INIT(("box_status=%x\n", box_status)); + dev_dbg(chip->card->dev, "box_status=%x\n", box_status); if (box_status == E3G_ASIC_NOT_LOADED) return -ENODEV; @@ -76,7 +76,8 @@ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq, if (wait_handshake(chip)) return -EIO; - DE_ACT(("WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq)); + dev_dbg(chip->card->dev, + "WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq); ctl = cpu_to_le32(ctl); frq = cpu_to_le32(frq); @@ -89,7 +90,7 @@ static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq, return send_vector(chip, DSP_VC_WRITE_CONTROL_REG); } - DE_ACT(("WriteControlReg: not written, no change\n")); + dev_dbg(chip->card->dev, "WriteControlReg: not written, no change\n"); return 0; } @@ -258,8 +259,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) /* Only set the clock for internal mode. */ if (chip->input_clock != ECHO_CLOCK_INTERNAL) { - DE_ACT(("set_sample_rate: Cannot set sample rate - " - "clock not set to CLK_CLOCKININTERNAL\n")); + dev_warn(chip->card->dev, + "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n"); /* Save the rate anyhow */ chip->comm_page->sample_rate = cpu_to_le32(rate); chip->sample_rate = rate; @@ -313,7 +314,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */ chip->sample_rate = rate; - DE_ACT(("SetSampleRate: %d clock %x\n", rate, control_reg)); + dev_dbg(chip->card->dev, + "SetSampleRate: %d clock %x\n", rate, control_reg); /* Tell the DSP about it - DSP reads both control reg & freq reg */ return write_control_reg(chip, control_reg, frq_reg, 0); @@ -326,7 +328,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - DE_ACT(("set_input_clock:\n")); + dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -335,13 +337,13 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - DE_ACT(("Set Echo3G clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - DE_ACT(("Set Echo3G clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n"); control_reg |= E3G_SPDIF_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -351,12 +353,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - DE_ACT(("Set Echo3G clock to ADAT\n")); + dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n"); control_reg |= E3G_ADAT_CLOCK; control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - DE_ACT(("Set Echo3G clock to WORD\n")); + dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n"); control_reg |= E3G_WORD_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -364,7 +366,8 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; default: - DE_ACT(("Input clock 0x%x not supported for Echo3G\n", clock)); + dev_err(chip->card->dev, + "Input clock 0x%x not supported for Echo3G\n", clock); return -EINVAL; } @@ -392,7 +395,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) incompatible_clock = TRUE; break; default: - DE_ACT(("Digital mode not supported: %d\n", mode)); + dev_err(chip->card->dev, + "Digital mode not supported: %d\n", mode); return -EINVAL; } @@ -427,6 +431,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) return err; chip->digital_mode = mode; - DE_ACT(("set_digital_mode(%d)\n", chip->digital_mode)); + dev_dbg(chip->card->dev, "set_digital_mode(%d)\n", chip->digital_mode); return incompatible_clock; } diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 977b2bd..ba9d4f1 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -80,7 +80,7 @@ static int send_vector(struct echoaudio *chip, u32 command) udelay(1); } - DE_ACT((KERN_ERR "timeout on send_vector\n")); + dev_err(chip->card->dev, "timeout on send_vector\n"); return -EBUSY; } @@ -104,7 +104,7 @@ static int write_dsp(struct echoaudio *chip, u32 data) } chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - DE_ACT((KERN_ERR "write_dsp: Set bad_board to TRUE\n")); + dev_dbg(chip->card->dev, "write_dsp: Set bad_board to TRUE\n"); return -EIO; } @@ -127,7 +127,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data) } chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - DE_INIT((KERN_ERR "read_dsp: Set bad_board to TRUE\n")); + dev_err(chip->card->dev, "read_dsp: Set bad_board to TRUE\n"); return -EIO; } @@ -154,8 +154,9 @@ static int read_sn(struct echoaudio *chip) return -EIO; } } - DE_INIT(("Read serial number %08x %08x %08x %08x %08x\n", - sn[0], sn[1], sn[2], sn[3], sn[4])); + dev_dbg(chip->card->dev, + "Read serial number %08x %08x %08x %08x %08x\n", + sn[0], sn[1], sn[2], sn[3], sn[4]); return 0; } @@ -205,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - DE_INIT(("ASIC loaded\n")); + dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; la_error: - DE_INIT(("failed on write_dsp\n")); + dev_err(chip->card->dev, "failed on write_dsp\n"); free_firmware(fw, chip); return -EIO; } @@ -241,8 +242,9 @@ static int install_resident_loader(struct echoaudio *chip) loader is already installed, host flag 5 will be on. */ status = get_dsp_register(chip, CHI32_STATUS_REG); if (status & CHI32_STATUS_REG_HF5) { - DE_INIT(("Resident loader already installed; status is 0x%x\n", - status)); + dev_dbg(chip->card->dev, + "Resident loader already installed; status is 0x%x\n", + status); return 0; } @@ -283,12 +285,14 @@ static int install_resident_loader(struct echoaudio *chip) /* Write the count to the DSP */ if (write_dsp(chip, words)) { - DE_INIT(("install_resident_loader: Failed to write word count!\n")); + dev_err(chip->card->dev, + "install_resident_loader: Failed to write word count!\n"); goto irl_error; } /* Write the DSP address */ if (write_dsp(chip, address)) { - DE_INIT(("install_resident_loader: Failed to write DSP address!\n")); + dev_err(chip->card->dev, + "install_resident_loader: Failed to write DSP address!\n"); goto irl_error; } /* Write out this block of code to the DSP */ @@ -297,7 +301,8 @@ static int install_resident_loader(struct echoaudio *chip) data = ((u32)code[index] << 16) + code[index + 1]; if (write_dsp(chip, data)) { - DE_INIT(("install_resident_loader: Failed to write DSP code\n")); + dev_err(chip->card->dev, + "install_resident_loader: Failed to write DSP code\n"); goto irl_error; } index += 2; @@ -312,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } if (i == 200) { - DE_INIT(("Resident loader failed to set HF5\n")); + dev_err(chip->card->dev, "Resident loader failed to set HF5\n"); goto irl_error; } - DE_INIT(("Resident loader successfully installed\n")); + dev_dbg(chip->card->dev, "Resident loader successfully installed\n"); free_firmware(fw, chip); return 0; @@ -334,14 +339,14 @@ static int load_dsp(struct echoaudio *chip, u16 *code) int index, words, i; if (chip->dsp_code == code) { - DE_INIT(("DSP is already loaded!\n")); + dev_warn(chip->card->dev, "DSP is already loaded!\n"); return 0; } chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */ - DE_INIT(("load_dsp: Set bad_board to TRUE\n")); + dev_dbg(chip->card->dev, "load_dsp: Set bad_board to TRUE\n"); /* If this board requires a resident loader, install it. */ #ifdef DSP_56361 @@ -351,7 +356,8 @@ static int load_dsp(struct echoaudio *chip, u16 *code) /* Send software reset command */ if (send_vector(chip, DSP_VC_RESET) < 0) { - DE_INIT(("LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n")); + dev_err(chip->card->dev, + "LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n"); return -EIO; } /* Delay 10us */ @@ -366,7 +372,8 @@ static int load_dsp(struct echoaudio *chip, u16 *code) } if (i == 1000) { - DE_INIT(("load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n")); + dev_err(chip->card->dev, + "load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n"); return -EIO; } @@ -403,29 +410,34 @@ static int load_dsp(struct echoaudio *chip, u16 *code) index += 2; if (write_dsp(chip, words) < 0) { - DE_INIT(("load_dsp: failed to write number of DSP words\n")); + dev_err(chip->card->dev, + "load_dsp: failed to write number of DSP words\n"); return -EIO; } if (write_dsp(chip, address) < 0) { - DE_INIT(("load_dsp: failed to write DSP address\n")); + dev_err(chip->card->dev, + "load_dsp: failed to write DSP address\n"); return -EIO; } if (write_dsp(chip, mem_type) < 0) { - DE_INIT(("load_dsp: failed to write DSP memory type\n")); + dev_err(chip->card->dev, + "load_dsp: failed to write DSP memory type\n"); return -EIO; } /* Code */ for (i = 0; i < words; i++, index+=2) { data = ((u32)code[index] << 16) + code[index + 1]; if (write_dsp(chip, data) < 0) { - DE_INIT(("load_dsp: failed to write DSP data\n")); + dev_err(chip->card->dev, + "load_dsp: failed to write DSP data\n"); return -EIO; } } } if (write_dsp(chip, 0) < 0) { /* We're done!!! */ - DE_INIT(("load_dsp: Failed to write final zero\n")); + dev_err(chip->card->dev, + "load_dsp: Failed to write final zero\n"); return -EIO; } udelay(10); @@ -438,12 +450,14 @@ static int load_dsp(struct echoaudio *chip, u16 *code) get_dsp_register(chip, CHI32_CONTROL_REG) & ~0x1b00); if (write_dsp(chip, DSP_FNC_SET_COMMPAGE_ADDR) < 0) { - DE_INIT(("load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n")); + dev_err(chip->card->dev, + "load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n"); return -EIO; } if (write_dsp(chip, chip->comm_page_phys) < 0) { - DE_INIT(("load_dsp: Failed to write comm page address\n")); + dev_err(chip->card->dev, + "load_dsp: Failed to write comm page address\n"); return -EIO; } @@ -452,19 +466,21 @@ static int load_dsp(struct echoaudio *chip, u16 *code) We don't actually use the serial number but we have to get it as part of the DSP init voodoo. */ if (read_sn(chip) < 0) { - DE_INIT(("load_dsp: Failed to read serial number\n")); + dev_err(chip->card->dev, + "load_dsp: Failed to read serial number\n"); return -EIO; } chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - DE_INIT(("load_dsp: OK!\n")); + dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); } - DE_INIT(("load_dsp: DSP load timed out waiting for HF4\n")); + dev_err(chip->card->dev, + "load_dsp: DSP load timed out waiting for HF4\n"); return -EIO; } @@ -658,7 +674,7 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - DE_INIT(("restore_dsp_settings\n")); + dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -755,7 +771,7 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - DE_INIT(("restore_dsp_rettings done\n")); + dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -835,7 +851,8 @@ static void set_audio_format(struct echoaudio *chip, u16 pipe_index, break; } } - DE_ACT(("set_audio_format[%d] = %x\n", pipe_index, dsp_format)); + dev_dbg(chip->card->dev, + "set_audio_format[%d] = %x\n", pipe_index, dsp_format); chip->comm_page->audio_format[pipe_index] = cpu_to_le16(dsp_format); } @@ -848,7 +865,7 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - DE_ACT(("start_transport %x\n", channel_mask)); + dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -866,7 +883,7 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, return 0; } - DE_ACT(("start_transport: No pipes to start!\n")); + dev_err(chip->card->dev, "start_transport: No pipes to start!\n"); return -EINVAL; } @@ -874,7 +891,7 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - DE_ACT(("pause_transport %x\n", channel_mask)); + dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -893,7 +910,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) return 0; } - DE_ACT(("pause_transport: No pipes to stop!\n")); + dev_warn(chip->card->dev, "pause_transport: No pipes to stop!\n"); return 0; } @@ -901,7 +918,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - DE_ACT(("stop_transport %x\n", channel_mask)); + dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -920,7 +937,7 @@ static int stop_transport(struct echoaudio *chip, u32 channel_mask) return 0; } - DE_ACT(("stop_transport: No pipes to stop!\n")); + dev_warn(chip->card->dev, "stop_transport: No pipes to stop!\n"); return 0; } @@ -937,7 +954,8 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - DE_ACT(("rest_in_peace() open=%x\n", chip->pipe_alloc_mask)); + dev_dbg(chip->card->dev, + "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -965,7 +983,8 @@ static int init_dsp_comm_page(struct echoaudio *chip) { /* Check if the compiler added extra padding inside the structure */ if (offsetof(struct comm_page, midi_output) != 0xbe0) { - DE_INIT(("init_dsp_comm_page() - Invalid struct comm_page structure\n")); + dev_err(chip->card->dev, + "init_dsp_comm_page() - Invalid struct comm_page structure\n"); return -EPERM; } @@ -999,7 +1018,7 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - DE_INIT(("init_line_levels\n")); + dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1051,7 +1070,8 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, u32 channel_mask; char is_cyclic; - DE_ACT(("allocate_pipes: ch=%d int=%d\n", pipe_index, interleave)); + dev_dbg(chip->card->dev, + "allocate_pipes: ch=%d int=%d\n", pipe_index, interleave); if (chip->bad_board) return -EIO; @@ -1061,7 +1081,8 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, for (channel_mask = i = 0; i < interleave; i++) channel_mask |= 1 << (pipe_index + i); if (chip->pipe_alloc_mask & channel_mask) { - DE_ACT(("allocate_pipes: channel already open\n")); + dev_err(chip->card->dev, + "allocate_pipes: channel already open\n"); return -EAGAIN; } @@ -1078,7 +1099,7 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - DE_ACT(("allocate_pipes: ok\n")); + dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1089,7 +1110,7 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - DE_ACT(("free_pipes: Pipe %d\n", pipe->index)); + dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) @@ -1131,7 +1152,7 @@ static int sglist_add_mapping(struct echoaudio *chip, struct audiopipe *pipe, list[head].size = cpu_to_le32(length); pipe->sglist_head++; } else { - DE_ACT(("SGlist: too many fragments\n")); + dev_err(chip->card->dev, "SGlist: too many fragments\n"); return -ENOMEM; } return 0; diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c index afa2733..23a0994 100644 --- a/sound/pci/echoaudio/echoaudio_gml.c +++ b/sound/pci/echoaudio/echoaudio_gml.c @@ -46,7 +46,8 @@ static int check_asic_status(struct echoaudio *chip) /* The DSP will return a value to indicate whether or not the ASIC is currently loaded */ if (read_dsp(chip, &asic_status) < 0) { - DE_INIT(("check_asic_status: failed on read_dsp\n")); + dev_err(chip->card->dev, + "check_asic_status: failed on read_dsp\n"); chip->asic_loaded = FALSE; return -EIO; } @@ -68,7 +69,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force) else value &= ~GML_DIGITAL_IN_AUTO_MUTE; - DE_ACT(("write_control_reg: 0x%x\n", value)); + dev_dbg(chip->card->dev, "write_control_reg: 0x%x\n", value); /* Write the control register */ value = cpu_to_le32(value); @@ -91,7 +92,7 @@ If the auto-mute is disabled, the digital inputs are enabled regardless of what the input clock is set or what is connected. */ static int set_input_auto_mute(struct echoaudio *chip, int automute) { - DE_ACT(("set_input_auto_mute %d\n", automute)); + dev_dbg(chip->card->dev, "set_input_auto_mute %d\n", automute); chip->digital_in_automute = automute; @@ -194,7 +195,7 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) if ((err = write_control_reg(chip, control_reg, FALSE))) return err; chip->professional_spdif = prof; - DE_ACT(("set_professional_spdif to %s\n", - prof ? "Professional" : "Consumer")); + dev_dbg(chip->card->dev, "set_professional_spdif to %s\n", + prof ? "Professional" : "Consumer"); return 0; } diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index d1615a0..a959eae 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,12 +37,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Gina20\n")); + dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -62,7 +63,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -149,7 +150,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - DE_ACT(("set_input_clock:\n")); + dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -158,7 +159,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) chip->spdif_status = GD_SPDIF_STATUS_UNDEF; set_sample_rate(chip, chip->sample_rate); chip->input_clock = clock; - DE_ACT(("Set Gina clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n"); break; case ECHO_CLOCK_SPDIF: chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN; @@ -166,7 +167,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) clear_handshake(chip); send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE); chip->clock_state = GD_CLOCK_SPDIFIN; - DE_ACT(("Set Gina20 clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n"); chip->input_clock = clock; break; default: @@ -208,7 +209,7 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - DE_ACT(("set_professional_spdif %d\n", prof)); + dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 98f7cfa..c8ea576 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Gina24\n")); + dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -78,7 +79,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -155,7 +156,7 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - DE_INIT(("load_asic() done\n")); + dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -171,8 +172,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) /* Only set the clock for internal mode. */ if (chip->input_clock != ECHO_CLOCK_INTERNAL) { - DE_ACT(("set_sample_rate: Cannot set sample rate - " - "clock not set to CLK_CLOCKININTERNAL\n")); + dev_warn(chip->card->dev, + "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n"); /* Save the rate anyhow */ chip->comm_page->sample_rate = cpu_to_le32(rate); chip->sample_rate = rate; @@ -217,7 +218,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) clock = GML_8KHZ; break; default: - DE_ACT(("set_sample_rate: %d invalid!\n", rate)); + dev_err(chip->card->dev, + "set_sample_rate: %d invalid!\n", rate); return -EINVAL; } @@ -225,7 +227,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */ chip->sample_rate = rate; - DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); + dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, clock); return write_control_reg(chip, control_reg, FALSE); } @@ -236,7 +238,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - DE_ACT(("set_input_clock:\n")); + dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -245,13 +247,13 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - DE_ACT(("Set Gina24 clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - DE_ACT(("Set Gina24 clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -261,21 +263,22 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - DE_ACT(("Set Gina24 clock to ADAT\n")); + dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n"); control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC: - DE_ACT(("Set Gina24 clock to ESYNC\n")); + dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n"); control_reg |= GML_ESYNC_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC96: - DE_ACT(("Set Gina24 clock to ESYNC96\n")); + dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n"); control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE; break; default: - DE_ACT(("Input clock 0x%x not supported for Gina24\n", clock)); + dev_err(chip->card->dev, + "Input clock 0x%x not supported for Gina24\n", clock); return -EINVAL; } @@ -304,7 +307,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) incompatible_clock = TRUE; break; default: - DE_ACT(("Digital mode not supported: %d\n", mode)); + dev_err(chip->card->dev, + "Digital mode not supported: %d\n", mode); return -EINVAL; } @@ -344,6 +348,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) return err; chip->digital_mode = mode; - DE_ACT(("set_digital_mode to %d\n", chip->digital_mode)); + dev_dbg(chip->card->dev, + "set_digital_mode to %d\n", chip->digital_mode); return incompatible_clock; } diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 5e85f14..cdeb073 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Indigo\n")); + dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -109,7 +110,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg = MIA_32000; break; default: - DE_ACT(("set_sample_rate: %d invalid!\n", rate)); + dev_err(chip->card->dev, + "set_sample_rate: %d invalid!\n", rate); return -EINVAL; } @@ -147,7 +149,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, index = output * num_pipes_out(chip) + pipe; chip->comm_page->vmixer[index] = gain; - DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + dev_dbg(chip->card->dev, + "set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain); return 0; } diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c index 2e4ab3e..ceda2d7 100644 --- a/sound/pci/echoaudio/indigo_express_dsp.c +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -61,7 +61,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg |= clock; if (control_reg != old_control_reg) { - DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); + dev_dbg(chip->card->dev, + "set_sample_rate: %d clock %d\n", rate, clock); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); @@ -89,7 +90,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, index = output * num_pipes_out(chip) + pipe; chip->comm_page->vmixer[index] = gain; - DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + dev_dbg(chip->card->dev, + "set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain); return 0; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 68f3c8c..133915c 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Indigo DJ\n")); + dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -109,7 +110,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg = MIA_32000; break; default: - DE_ACT(("set_sample_rate: %d invalid!\n", rate)); + dev_err(chip->card->dev, + "set_sample_rate: %d invalid!\n", rate); return -EINVAL; } @@ -147,7 +149,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, index = output * num_pipes_out(chip) + pipe; chip->comm_page->vmixer[index] = gain; - DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + dev_dbg(chip->card->dev, + "set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain); return 0; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index bb9632c7..26cdfcf 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,13 +35,14 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Indigo DJx\n")); + dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; err = init_dsp_comm_page(chip); if (err < 0) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -59,7 +60,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index beb9a5b6..5e6df7c 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,12 +38,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Indigo IO\n")); + dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -60,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +119,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, index = output * num_pipes_out(chip) + pipe; chip->comm_page->vmixer[index] = gain; - DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + dev_dbg(chip->card->dev, + "set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain); return 0; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 394c6e76..90cdd27 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,13 +35,14 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Indigo IOx\n")); + dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; err = init_dsp_comm_page(chip); if (err < 0) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -59,7 +60,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 53ce946..7f0f6ea 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,12 +40,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Layla20\n")); + dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -64,7 +65,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -121,7 +122,8 @@ static int check_asic_status(struct echoaudio *chip) /* The DSP will return a value to indicate whether or not the ASIC is currently loaded */ if (read_dsp(chip, &asic_status) < 0) { - DE_ACT(("check_asic_status: failed on read_dsp\n")); + dev_err(chip->card->dev, + "check_asic_status: failed on read_dsp\n"); return -EIO; } @@ -164,8 +166,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) /* Only set the clock for internal mode. Do not return failure, simply treat it as a non-event. */ if (chip->input_clock != ECHO_CLOCK_INTERNAL) { - DE_ACT(("set_sample_rate: Cannot set sample rate - " - "clock not set to CLK_CLOCKININTERNAL\n")); + dev_warn(chip->card->dev, + "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n"); chip->comm_page->sample_rate = cpu_to_le32(rate); chip->sample_rate = rate; return 0; @@ -174,7 +176,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) if (wait_handshake(chip)) return -EIO; - DE_ACT(("set_sample_rate(%d)\n", rate)); + dev_dbg(chip->card->dev, "set_sample_rate(%d)\n", rate); chip->sample_rate = rate; chip->comm_page->sample_rate = cpu_to_le32(rate); clear_handshake(chip); @@ -188,29 +190,30 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - DE_ACT(("set_input_clock:\n")); + dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: - DE_ACT(("Set Layla20 clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n"); rate = chip->sample_rate; clock = LAYLA20_CLOCK_INTERNAL; break; case ECHO_CLOCK_SPDIF: - DE_ACT(("Set Layla20 clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n"); clock = LAYLA20_CLOCK_SPDIF; break; case ECHO_CLOCK_WORD: - DE_ACT(("Set Layla20 clock to WORD\n")); + dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n"); clock = LAYLA20_CLOCK_WORD; break; case ECHO_CLOCK_SUPER: - DE_ACT(("Set Layla20 clock to SUPER\n")); + dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n"); clock = LAYLA20_CLOCK_SUPER; break; default: - DE_ACT(("Input clock 0x%x not supported for Layla24\n", - clock_source)); + dev_err(chip->card->dev, + "Input clock 0x%x not supported for Layla24\n", + clock_source); return -EINVAL; } chip->input_clock = clock_source; @@ -229,7 +232,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - DE_ACT(("set_output_clock: %d\n", clock)); + dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -238,7 +241,7 @@ static int set_output_clock(struct echoaudio *chip, u16 clock) clock = LAYLA20_OUTPUT_CLOCK_WORD; break; default: - DE_ACT(("set_output_clock wrong clock\n")); + dev_err(chip->card->dev, "set_output_clock wrong clock\n"); return -EINVAL; } @@ -283,7 +286,7 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - DE_ACT(("set_professional_spdif %d\n", prof)); + dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 8c04164..eb8f218 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,12 +40,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Layla24\n")); + dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -69,7 +70,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -117,7 +118,7 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - DE_INIT(("load_asic\n")); + dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -151,7 +152,7 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - DE_INIT(("load_asic() done\n")); + dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -167,8 +168,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) /* Only set the clock for internal mode. */ if (chip->input_clock != ECHO_CLOCK_INTERNAL) { - DE_ACT(("set_sample_rate: Cannot set sample rate - " - "clock not set to CLK_CLOCKININTERNAL\n")); + dev_warn(chip->card->dev, + "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n"); /* Save the rate anyhow */ chip->comm_page->sample_rate = cpu_to_le32(rate); chip->sample_rate = rate; @@ -241,7 +242,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP ? */ chip->sample_rate = rate; - DE_ACT(("set_sample_rate: %d clock %d\n", rate, control_reg)); + dev_dbg(chip->card->dev, + "set_sample_rate: %d clock %d\n", rate, control_reg); return write_control_reg(chip, control_reg, FALSE); } @@ -260,7 +262,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Pick the new clock */ switch (clock) { case ECHO_CLOCK_INTERNAL: - DE_ACT(("Set Layla24 clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -269,7 +271,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_SPDIF_CLOCK; /* Layla24 doesn't support 96KHz S/PDIF */ control_reg &= ~GML_DOUBLE_SPEED_MODE; - DE_ACT(("Set Layla24 clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n"); break; case ECHO_CLOCK_WORD: control_reg |= GML_WORD_CLOCK; @@ -277,17 +279,18 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_DOUBLE_SPEED_MODE; else control_reg &= ~GML_DOUBLE_SPEED_MODE; - DE_ACT(("Set Layla24 clock to WORD\n")); + dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n"); break; case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; - DE_ACT(("Set Layla24 clock to ADAT\n")); + dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n"); break; default: - DE_ACT(("Input clock 0x%x not supported for Layla24\n", clock)); + dev_err(chip->card->dev, + "Input clock 0x%x not supported for Layla24\n", clock); return -EINVAL; } @@ -353,7 +356,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) asic = FW_LAYLA24_2A_ASIC; break; default: - DE_ACT(("Digital mode not supported: %d\n", mode)); + dev_err(chip->card->dev, + "Digital mode not supported: %d\n", mode); return -EINVAL; } @@ -393,6 +397,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) return err; chip->digital_mode = mode; - DE_ACT(("set_digital_mode to %d\n", mode)); + dev_dbg(chip->card->dev, "set_digital_mode to %d\n", mode); return incompatible_clock; } diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 6ebfa6e..ed2f21d 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Mia\n")); + dev_dbg(chip->card->dev, "init_hw() - Mia\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -66,7 +67,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -126,7 +127,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg = MIA_32000; break; default: - DE_ACT(("set_sample_rate: %d invalid!\n", rate)); + dev_err(chip->card->dev, + "set_sample_rate: %d invalid!\n", rate); return -EINVAL; } @@ -153,7 +155,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - DE_ACT(("set_input_clock(%d)\n", clock)); + dev_dbg(chip->card->dev, "set_input_clock(%d)\n", clock); if (snd_BUG_ON(clock != ECHO_CLOCK_INTERNAL && clock != ECHO_CLOCK_SPDIF)) return -EINVAL; @@ -181,7 +183,8 @@ static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, index = output * num_pipes_out(chip) + pipe; chip->comm_page->vmixer[index] = gain; - DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + dev_dbg(chip->card->dev, + "set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain); return 0; } @@ -211,7 +214,7 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - DE_ACT(("set_professional_spdif %d\n", prof)); + dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 7f4dfae..8d43c5a 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -36,7 +36,7 @@ /* Start and stop Midi input */ static int enable_midi_input(struct echoaudio *chip, char enable) { - DE_MID(("enable_midi_input(%d)\n", enable)); + dev_dbg(chip->card->dev, "enable_midi_input(%d)\n", enable); if (wait_handshake(chip)) return -EIO; @@ -74,7 +74,7 @@ static int write_midi(struct echoaudio *chip, u8 *data, int bytes) chip->comm_page->midi_out_free_count = 0; clear_handshake(chip); send_vector(chip, DSP_VC_MIDI_WRITE); - DE_MID(("write_midi: %d\n", bytes)); + dev_dbg(chip->card->dev, "write_midi: %d\n", bytes); return bytes; } @@ -157,7 +157,7 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = substream; - DE_MID(("rawmidi_iopen\n")); + dev_dbg(chip->card->dev, "rawmidi_iopen\n"); return 0; } @@ -183,7 +183,7 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = NULL; - DE_MID(("rawmidi_iclose\n")); + dev_dbg(chip->card->dev, "rawmidi_iclose\n"); return 0; } @@ -196,7 +196,7 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) chip->tinuse = 0; chip->midi_full = 0; chip->midi_out = substream; - DE_MID(("rawmidi_oopen\n")); + dev_dbg(chip->card->dev, "rawmidi_open\n"); return 0; } @@ -209,7 +209,7 @@ static void snd_echo_midi_output_write(unsigned long data) int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; - DE_MID(("snd_echo_midi_output_write\n")); + dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n"); /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ sent = bytes = 0; @@ -218,7 +218,7 @@ static void snd_echo_midi_output_write(unsigned long data) if (!snd_rawmidi_transmit_empty(chip->midi_out)) { bytes = snd_rawmidi_transmit_peek(chip->midi_out, buf, MIDI_OUT_BUFFER_SIZE - 1); - DE_MID(("Try to send %d bytes...\n", bytes)); + dev_dbg(chip->card->dev, "Try to send %d bytes...\n", bytes); sent = write_midi(chip, buf, bytes); if (sent < 0) { dev_err(chip->card->dev, @@ -227,12 +227,12 @@ static void snd_echo_midi_output_write(unsigned long data) sent = 9000; chip->midi_full = 1; } else if (sent > 0) { - DE_MID(("%d bytes sent\n", sent)); + dev_dbg(chip->card->dev, "%d bytes sent\n", sent); snd_rawmidi_transmit_ack(chip->midi_out, sent); } else { /* Buffer is full. DSP's internal buffer is 64 (128 ?) bytes long. Let's wait until half of them are sent */ - DE_MID(("Full\n")); + dev_dbg(chip->card->dev, "Full\n"); sent = 32; chip->midi_full = 1; } @@ -244,7 +244,8 @@ static void snd_echo_midi_output_write(unsigned long data) sent */ time = (sent << 3) / 25 + 1; /* 8/25=0.32ms to send a byte */ mod_timer(&chip->timer, jiffies + (time * HZ + 999) / 1000); - DE_MID(("Timer armed(%d)\n", ((time * HZ + 999) / 1000))); + dev_dbg(chip->card->dev, + "Timer armed(%d)\n", ((time * HZ + 999) / 1000)); } spin_unlock_irqrestore(&chip->lock, flags); } @@ -256,7 +257,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream { struct echoaudio *chip = substream->rmidi->private_data; - DE_MID(("snd_echo_midi_output_trigger(%d)\n", up)); + dev_dbg(chip->card->dev, "snd_echo_midi_output_trigger(%d)\n", up); spin_lock_irq(&chip->lock); if (up) { if (!chip->tinuse) { @@ -270,7 +271,7 @@ static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream chip->tinuse = 0; spin_unlock_irq(&chip->lock); del_timer_sync(&chip->timer); - DE_MID(("Timer removed\n")); + dev_dbg(chip->card->dev, "Timer removed\n"); return; } } @@ -287,7 +288,7 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_out = NULL; - DE_MID(("rawmidi_oclose\n")); + dev_dbg(chip->card->dev, "rawmidi_oclose\n"); return 0; } @@ -327,6 +328,6 @@ static int snd_echo_midi_create(struct snd_card *card, chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; - DE_INIT(("MIDI ok\n")); + dev_dbg(chip->card->dev, "MIDI ok\n"); return 0; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index 6e6a7eb..cc46a8c 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,12 +41,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - DE_INIT(("init_hw() - Mona\n")); + dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; if ((err = init_dsp_comm_page(chip))) { - DE_INIT(("init_hw - could not initialize DSP comm page\n")); + dev_err(chip->card->dev, + "init_hw - could not initialize DSP comm page\n"); return err; } @@ -71,7 +72,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - DE_INIT(("init_hw done\n")); + dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -202,8 +203,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) /* Only set the clock for internal mode. */ if (chip->input_clock != ECHO_CLOCK_INTERNAL) { - DE_ACT(("set_sample_rate: Cannot set sample rate - " - "clock not set to CLK_CLOCKININTERNAL\n")); + dev_dbg(chip->card->dev, + "Cannot set sample rate - clock not set to CLK_CLOCKININTERNAL\n"); /* Save the rate anyhow */ chip->comm_page->sample_rate = cpu_to_le32(rate); chip->sample_rate = rate; @@ -279,7 +280,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) clock = GML_8KHZ; break; default: - DE_ACT(("set_sample_rate: %d invalid!\n", rate)); + dev_err(chip->card->dev, + "set_sample_rate: %d invalid!\n", rate); return -EINVAL; } @@ -287,7 +289,8 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */ chip->sample_rate = rate; - DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); + dev_dbg(chip->card->dev, + "set_sample_rate: %d clock %d\n", rate, clock); return write_control_reg(chip, control_reg, force_write); } @@ -299,7 +302,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - DE_ACT(("set_input_clock:\n")); + dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) @@ -312,7 +315,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - DE_ACT(("Set Mona clock to INTERNAL\n")); + dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -324,7 +327,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) spin_lock_irq(&chip->lock); if (err < 0) return err; - DE_ACT(("Set Mona clock to SPDIF\n")); + dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -332,7 +335,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - DE_ACT(("Set Mona clock to WORD\n")); + dev_dbg(chip->card->dev, "Set Mona clock to WORD\n"); spin_unlock_irq(&chip->lock); err = switch_asic(chip, clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96); @@ -346,14 +349,15 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ADAT: - DE_ACT(("Set Mona clock to ADAT\n")); + dev_dbg(chip->card->dev, "Set Mona clock to ADAT\n"); if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; default: - DE_ACT(("Input clock 0x%x not supported for Mona\n", clock)); + dev_err(chip->card->dev, + "Input clock 0x%x not supported for Mona\n", clock); return -EINVAL; } @@ -381,7 +385,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) incompatible_clock = TRUE; break; default: - DE_ACT(("Digital mode not supported: %d\n", mode)); + dev_err(chip->card->dev, + "Digital mode not supported: %d\n", mode); return -EINVAL; } @@ -422,6 +427,6 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) return err; chip->digital_mode = mode; - DE_ACT(("set_digital_mode to %d\n", mode)); + dev_dbg(chip->card->dev, "set_digital_mode to %d\n", mode); return incompatible_clock; } -- cgit v1.1 From 31604d35db18c1382c7ee9fa836ff9ab0b4d2751 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 3 Nov 2014 14:54:36 +0100 Subject: ALSA: emu10k1: Deletion of unnecessary checks before three function calls The functions kfree(), release_firmware() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 9 +++------ sound/pci/emu10k1/emufx.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2292697..b4458a6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1289,10 +1289,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); - if (emu->firmware) - release_firmware(emu->firmware); - if (emu->dock_fw) - release_firmware(emu->dock_fw); + release_firmware(emu->firmware); + release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); /* remove reserved page */ @@ -1301,8 +1299,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) (struct snd_util_memblk *)emu->reserved_page); emu->reserved_page = NULL; } - if (emu->memhdr) - snd_util_memhdr_free(emu->memhdr); + snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) snd_dma_free_pages(&emu->silent_page); if (emu->ptb_pages.area) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 745f062..eb5c0ab 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -777,8 +777,7 @@ static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl) kctl->private_value = 0; list_del(&ctl->list); kfree(ctl); - if (kctl->tlv.p) - kfree(kctl->tlv.p); + kfree(kctl->tlv.p); } static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, -- cgit v1.1 From f74e2c9cb03076d11e807088d2120a8a381a6f3c Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Mon, 3 Nov 2014 21:59:37 +0800 Subject: ASoC: Intel: Correct a macro for FW message For the broadwell official released FW(Since 8.4.1.43), the macro SST_HSW_NO_CHANNELS is changed and fixed to 4, so here change it to 4. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index afd0ae1..387511f 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -21,7 +21,7 @@ #include #include -#define SST_HSW_NO_CHANNELS 2 +#define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) -- cgit v1.1 From 2603fe21b764eb7412598c8c6cd6199fb8b1d9c5 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 3 Nov 2014 21:35:48 +0100 Subject: ALSA: es18xx: Add GPO controls Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 52 ++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 42 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 63e7323..b481bb8 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1120,11 +1121,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1143,10 +1147,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1159,7 +1167,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1167,6 +1176,15 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + unsigned char cur = inb(chip->port + ES18XX_PM); + + if ((cur & mask) == val) + return 0; + outb((cur & ~mask) | val, chip->port + ES18XX_PM); + return 1; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1288,7 +1306,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1347,6 +1365,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1613,10 +1636,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1626,7 +1649,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1928,6 +1951,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- cgit v1.1 From 0b2e4959ceacb26eb586698d9ceecc0a6bd30f72 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 4 Nov 2014 13:15:10 +0800 Subject: ASoC: rt5645: make bias level more reasonale This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 57ba742..1423cb2 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2069,8 +2069,8 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { - case SND_SOC_BIAS_STANDBY: - if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -2085,15 +2085,24 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + break; + case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); - snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000); - snd_soc_write(codec, RT5645_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2 | + RT5645_PWR_FV1 | RT5645_PWR_FV2, 0x0); break; default: -- cgit v1.1 From e648f6add20d1cfb5945e24b5bffe5843476645b Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 4 Nov 2014 14:45:24 +0800 Subject: ASoC: Intel: Fix the driver data not set issue The priv_data is allocated again here wrongly, and it is not set to the driver data after assignment. This make the pdata->dev is NULL and oops occurs on the first call to hsw_volume_put. The resource has been allocated in driver probe callback hsw_pcm_dev_probe, so here just remove this sencond allocation is OK. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index cd54dd9..093b939 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -868,7 +868,6 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) dev = platform->dev; dma_dev = pdata->dma_dev; - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; priv_data->dev = platform->dev; priv_data->pm_state = HSW_PM_STATE_D0; -- cgit v1.1 From 4539441690cd31ae7d42e6f080033911a1788440 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 3 Nov 2014 19:33:02 +0100 Subject: ASoC: mioa701_wm9713: Don't opencode CODEC register access Properly use snd_soc_update_bits() instead of manually calling the CODEC driver's read and write callbacks. The later will stop working once the wm9713 driver has been converted to regmap. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/mioa701_wm9713.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 595eee3..a6b2be2 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -127,15 +127,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - unsigned short reg; /* Prepare GPIO8 for rear speaker amplifier */ - reg = codec->driver->read(codec, AC97_GPIO_CFG); - codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100); + snd_soc_update_bits(codec, AC97_GPIO_CFG, 0x100, 0x100); /* Prepare MIC input */ - reg = codec->driver->read(codec, AC97_3D_CONTROL); - codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000); + snd_soc_update_bits(codec, AC97_3D_CONTROL, 0xc000, 0xc000); return 0; } -- cgit v1.1 From 313665b983fe30af9d0eb274f7e03276e05a1bbf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Nov 2014 11:30:58 +0100 Subject: ASoC: Remove card field from snd_soc_dai struct The card field of the snd_soc_dai field is very rarely used. We can use dai->component->card instead and remove the card field from the snd_soc_dai struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 -- sound/soc/soc-core.c | 7 +------ sound/soc/soc-dapm.c | 2 +- 3 files changed, 2 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 45d0fa1..373d177 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -275,8 +275,6 @@ struct snd_soc_dai { unsigned int tx_mask; unsigned int rx_mask; - struct snd_soc_card *card; - struct list_head list; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ea1df20..f3216fc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1315,11 +1315,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); - /* config components */ - cpu_dai->card = card; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->card = card; - /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -2314,7 +2309,7 @@ EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = dai->card->snd_card; + struct snd_card *card = dai->component->card->snd_card; return snd_soc_add_controls(card, dai->dev, controls, num_controls, NULL, dai); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bf2c97..c5136bb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1043,7 +1043,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list) { - struct snd_soc_card *card = dai->card; + struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; int paths; -- cgit v1.1 From 8e2be56273666614e24756d7ee551203b8a86809 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Nov 2014 11:30:59 +0100 Subject: ASoC: Consolidate CPU and CODEC DAI probe CPU and CODEC DAI probe are performed in exactly the same way. Which means we can reuse the snd_soc_codec_dai_probe() for probing CPU DAIs as well. While we are at it also drop the unused card parameter form the function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 38 ++++++++++++-------------------------- 1 file changed, 12 insertions(+), 26 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f3216fc..406925c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1241,25 +1241,22 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, return 0; } -static int soc_probe_codec_dai(struct snd_soc_card *card, - struct snd_soc_dai *codec_dai, - int order) +static int soc_probe_dai(struct snd_soc_dai *dai, int order) { int ret; - if (!codec_dai->probed && codec_dai->driver->probe_order == order) { - if (codec_dai->driver->probe) { - ret = codec_dai->driver->probe(codec_dai); + if (!dai->probed && dai->driver->probe_order == order) { + if (dai->driver->probe) { + ret = dai->driver->probe(dai); if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: failed to probe CODEC DAI %s: %d\n", - codec_dai->name, ret); + dev_err(dai->dev, + "ASoC: failed to probe DAI %s: %d\n", + dai->name, ret); return ret; } } - /* mark codec_dai as probed and add to card dai list */ - codec_dai->probed = 1; + dai->probed = 1; } return 0; @@ -1318,24 +1315,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* set default power off timeout */ rtd->pmdown_time = pmdown_time; - /* probe the cpu_dai */ - if (!cpu_dai->probed && - cpu_dai->driver->probe_order == order) { - if (cpu_dai->driver->probe) { - ret = cpu_dai->driver->probe(cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, - "ASoC: failed to probe CPU DAI %s: %d\n", - cpu_dai->name, ret); - return ret; - } - } - cpu_dai->probed = 1; - } + ret = soc_probe_dai(cpu_dai, order); + if (ret) + return ret; /* probe the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) { - ret = soc_probe_codec_dai(card, rtd->codec_dais[i], order); + ret = soc_probe_dai(rtd->codec_dais[i], order); if (ret) return ret; } -- cgit v1.1 From d507941beb1ef98c19e2902007aee4faf36f854f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 4 Nov 2014 13:57:15 +0100 Subject: ALSA: pcm: Correct PCM BUG error message While converting to dev_*(), the message showing the invalid PCM position was wrongly tagged as if an XRUN although it's actually a BUG. This patch corrects the message again. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index dfc2854..7b9e2fb 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "XRUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- cgit v1.1 From f5914908a5b7b2338f210e56827a1ef35585dc6d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 4 Nov 2014 12:45:59 +0100 Subject: ALSA: pcm: Replace PCM hwptr tracking with tracepoints ALSA PCM core has a mechanism tracking the PCM hwptr updates for analyzing XRUNs. But its log is limited (up to 10) and its log output is a kernel message, which is hard to handle. In this patch, the hwptr logging is moved to the tracing infrastructure instead of its own. Not only the hwptr updates but also XRUN and hwptr errors are recorded on the trace log, so that user can see such events at the exact timing. The new "snd_pcm" entry will appear in the tracing events: # ls -F /sys/kernel/debug/tracing/events/snd_pcm enable filter hw_ptr_error/ hwptr/ xrun/ The hwptr is for the regular hwptr update events. An event trace looks like: aplay-26187 [004] d..3 4012.834761: hwptr: pcmC0D0p/sub0: POS: pos=488, old=0, base=0, period=1024, buf=16384 "POS" shows the hwptr update by the explicit position update call and "IRQ" means the hwptr update by the interrupt, i.e. snd_pcm_period_elapsed() call. The "pos" is the passed ring-buffer offset by the caller, "old" is the previous hwptr, "base" is the hwptr base position, "period" and "buf" are period- and buffer-size of the target PCM substream. (Note that the hwptr position displayed here isn't the ring-buffer offset. It increments up to the PCM position boundary.) The XRUN event appears similarly, but without "pos" field. The hwptr error events appear with the PCM identifier and its reason string, such as "Lost interrupt?". The XRUN and hwptr error reports on kernel message are still left, can be turned on/off via xrun_debug proc like before. But the bit 3, 4, 5 and 6 bits of xrun_debug proc are dropped by this patch. Also, along with the change, the message strings have been reformatted to be a bit more consistent. Last but not least, the hwptr reporting is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is set. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Procfile.txt | 13 --- sound/core/Makefile | 3 + sound/core/pcm_lib.c | 146 ++++++---------------------------- sound/core/pcm_trace.h | 110 +++++++++++++++++++++++++ 4 files changed, 137 insertions(+), 135 deletions(-) create mode 100644 sound/core/pcm_trace.h diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index 7fcd1ad..cfc4956 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -101,10 +101,6 @@ card*/pcm*/xrun_debug bit 0 = Enable XRUN/jiffies debug messages bit 1 = Show stack trace at XRUN / jiffies check bit 2 = Enable additional jiffies check - bit 3 = Log hwptr update at each period interrupt - bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr() - bit 5 = Show last 10 positions on error - bit 6 = Do above only once When the bit 0 is set, the driver will show the messages to kernel log when an xrun is detected. The debug message is @@ -121,15 +117,6 @@ card*/pcm*/xrun_debug buggy) hardware that doesn't give smooth pointer updates. This feature is enabled via the bit 2. - Bits 3 and 4 are for logging the hwptr records. Note that - these will give flood of kernel messages. - - When bit 5 is set, the driver logs the last 10 xrun errors and - the proc file shows each jiffies, position, period_size, - buffer_size, old_hw_ptr, and hw_ptr_base values. - - When bit 6 is set, the full xrun log is shown only once. - card*/pcm*/sub*/info The general information of this PCM sub-stream. diff --git a/sound/core/Makefile b/sound/core/Makefile index 394a389..4daf2f5 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -14,6 +14,9 @@ snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o +# for trace-points +CFLAGS_pcm_lib.o := -I$(src) + snd-pcm-dmaengine-objs := pcm_dmaengine.o snd-rawmidi-objs := rawmidi.o diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7b9e2fb..ec9e786 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -32,6 +32,15 @@ #include #include +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define CREATE_TRACE_POINTS +#include "pcm_trace.h" +#else +#define trace_hwptr(substream, pos, in_interrupt) +#define trace_xrun(substream) +#define trace_hw_ptr_error(substream, reason) +#endif + /* * fill ring buffer with silence * runtime->silence_start: starting pointer to silence area @@ -146,10 +155,6 @@ EXPORT_SYMBOL(snd_pcm_debug_name); #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ #define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ -#define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ -#define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ -#define XRUN_DEBUG_LOG (1<<5) /* show last 10 positions on err */ -#define XRUN_DEBUG_LOGONCE (1<<6) /* do above only once */ #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -168,6 +173,7 @@ static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + trace_xrun(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); @@ -180,97 +186,19 @@ static void xrun(struct snd_pcm_substream *substream) } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -#define hw_ptr_error(substream, fmt, args...) \ +#define hw_ptr_error(substream, in_interrupt, reason, fmt, args...) \ do { \ + trace_hw_ptr_error(substream, reason); \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ - xrun_log_show(substream); \ - pr_err_ratelimited("ALSA: PCM: " fmt, ##args); \ + pr_err_ratelimited("ALSA: PCM: [%c] " reason ": " fmt, \ + (in_interrupt) ? 'Q' : 'P', ##args); \ dump_stack_on_xrun(substream); \ } \ } while (0) -#define XRUN_LOG_CNT 10 - -struct hwptr_log_entry { - unsigned int in_interrupt; - unsigned long jiffies; - snd_pcm_uframes_t pos; - snd_pcm_uframes_t period_size; - snd_pcm_uframes_t buffer_size; - snd_pcm_uframes_t old_hw_ptr; - snd_pcm_uframes_t hw_ptr_base; -}; - -struct snd_pcm_hwptr_log { - unsigned int idx; - unsigned int hit: 1; - struct hwptr_log_entry entries[XRUN_LOG_CNT]; -}; - -static void xrun_log(struct snd_pcm_substream *substream, - snd_pcm_uframes_t pos, int in_interrupt) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_pcm_hwptr_log *log = runtime->hwptr_log; - struct hwptr_log_entry *entry; - - if (log == NULL) { - log = kzalloc(sizeof(*log), GFP_ATOMIC); - if (log == NULL) - return; - runtime->hwptr_log = log; - } else { - if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) - return; - } - entry = &log->entries[log->idx]; - entry->in_interrupt = in_interrupt; - entry->jiffies = jiffies; - entry->pos = pos; - entry->period_size = runtime->period_size; - entry->buffer_size = runtime->buffer_size; - entry->old_hw_ptr = runtime->status->hw_ptr; - entry->hw_ptr_base = runtime->hw_ptr_base; - log->idx = (log->idx + 1) % XRUN_LOG_CNT; -} - -static void xrun_log_show(struct snd_pcm_substream *substream) -{ - struct snd_pcm_hwptr_log *log = substream->runtime->hwptr_log; - struct hwptr_log_entry *entry; - char name[16]; - unsigned int idx; - int cnt; - - if (log == NULL) - return; - if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) - return; - snd_pcm_debug_name(substream, name, sizeof(name)); - for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { - entry = &log->entries[idx]; - if (entry->period_size == 0) - break; - pr_info("hwptr log: %s: %sj=%lu, pos=%ld/%ld/%ld, " - "hwptr=%ld/%ld\n", - name, entry->in_interrupt ? "[Q] " : "", - entry->jiffies, - (unsigned long)entry->pos, - (unsigned long)entry->period_size, - (unsigned long)entry->buffer_size, - (unsigned long)entry->old_hw_ptr, - (unsigned long)entry->hw_ptr_base); - idx++; - idx %= XRUN_LOG_CNT; - } - log->hit = 1; -} - #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ #define hw_ptr_error(substream, fmt, args...) do { } while (0) -#define xrun_log(substream, pos, in_interrupt) do { } while (0) -#define xrun_log_show(substream) do { } while (0) #endif @@ -343,7 +271,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (printk_ratelimit()) { char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); - xrun_log_show(substream); pcm_err(substream->pcm, "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, @@ -352,8 +279,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, pos = 0; } pos -= pos % runtime->min_align; - if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos, in_interrupt); + trace_hwptr(substream, pos, in_interrupt); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; if (in_interrupt) { @@ -388,22 +314,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta = new_hw_ptr - old_hw_ptr; if (delta < 0) delta += runtime->boundary; - if (xrun_debug(substream, in_interrupt ? - XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { - char name[16]; - snd_pcm_debug_name(substream, name, sizeof(name)); - pcm_dbg(substream->pcm, - "%s_update: %s: pos=%u/%u/%u, hwptr=%ld/%ld/%ld/%ld\n", - in_interrupt ? "period" : "hwptr", - name, - (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)delta, - (unsigned long)old_hw_ptr, - (unsigned long)new_hw_ptr, - (unsigned long)runtime->hw_ptr_base); - } if (runtime->no_period_wakeup) { snd_pcm_sframes_t xrun_threshold; @@ -431,13 +341,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* something must be really wrong */ if (delta >= runtime->buffer_size + runtime->period_size) { - hw_ptr_error(substream, - "Unexpected hw_pointer value %s" - "(stream=%i, pos=%ld, new_hw_ptr=%ld, " - "old_hw_ptr=%ld)\n", - in_interrupt ? "[Q] " : "[P]", - substream->stream, (long)pos, - (long)new_hw_ptr, (long)old_hw_ptr); + hw_ptr_error(substream, in_interrupt, "Unexpected hw_ptr", + "(stream=%i, pos=%ld, new_hw_ptr=%ld, old_hw_ptr=%ld)\n", + substream->stream, (long)pos, + (long)new_hw_ptr, (long)old_hw_ptr); return 0; } @@ -474,11 +381,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta--; } /* align hw_base to buffer_size */ - hw_ptr_error(substream, - "hw_ptr skipping! %s" - "(pos=%ld, delta=%ld, period=%ld, " - "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", - in_interrupt ? "[Q] " : "", + hw_ptr_error(substream, in_interrupt, "hw_ptr skipping", + "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, ((hdelta * HZ) / runtime->rate), hw_base, @@ -490,11 +394,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { - hw_ptr_error(substream, - "Lost interrupts? %s" - "(stream=%i, delta=%ld, new_hw_ptr=%ld, " - "old_hw_ptr=%ld)\n", - in_interrupt ? "[Q] " : "", + hw_ptr_error(substream, in_interrupt, + "Lost interrupts?", + "(stream=%i, delta=%ld, new_hw_ptr=%ld, old_hw_ptr=%ld)\n", substream->stream, (long)delta, (long)new_hw_ptr, (long)old_hw_ptr); diff --git a/sound/core/pcm_trace.h b/sound/core/pcm_trace.h new file mode 100644 index 0000000..b63b654 --- /dev/null +++ b/sound/core/pcm_trace.h @@ -0,0 +1,110 @@ +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_pcm +#define TRACE_INCLUDE_FILE pcm_trace + +#if !defined(_PCM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _PCM_TRACE_H + +#include + +TRACE_EVENT(hwptr, + TP_PROTO(struct snd_pcm_substream *substream, snd_pcm_uframes_t pos, bool irq), + TP_ARGS(substream, pos, irq), + TP_STRUCT__entry( + __field( bool, in_interrupt ) + __field( unsigned int, card ) + __field( unsigned int, device ) + __field( unsigned int, number ) + __field( unsigned int, stream ) + __field( snd_pcm_uframes_t, pos ) + __field( snd_pcm_uframes_t, period_size ) + __field( snd_pcm_uframes_t, buffer_size ) + __field( snd_pcm_uframes_t, old_hw_ptr ) + __field( snd_pcm_uframes_t, hw_ptr_base ) + ), + TP_fast_assign( + __entry->in_interrupt = (irq); + __entry->card = (substream)->pcm->card->number; + __entry->device = (substream)->pcm->device; + __entry->number = (substream)->number; + __entry->stream = (substream)->stream; + __entry->pos = (pos); + __entry->period_size = (substream)->runtime->period_size; + __entry->buffer_size = (substream)->runtime->buffer_size; + __entry->old_hw_ptr = (substream)->runtime->status->hw_ptr; + __entry->hw_ptr_base = (substream)->runtime->hw_ptr_base; + ), + TP_printk("pcmC%dD%d%c/sub%d: %s: pos=%lu, old=%lu, base=%lu, period=%lu, buf=%lu", + __entry->card, __entry->device, + __entry->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c', + __entry->number, + __entry->in_interrupt ? "IRQ" : "POS", + (unsigned long)__entry->pos, + (unsigned long)__entry->old_hw_ptr, + (unsigned long)__entry->hw_ptr_base, + (unsigned long)__entry->period_size, + (unsigned long)__entry->buffer_size) +); + +TRACE_EVENT(xrun, + TP_PROTO(struct snd_pcm_substream *substream), + TP_ARGS(substream), + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( unsigned int, device ) + __field( unsigned int, number ) + __field( unsigned int, stream ) + __field( snd_pcm_uframes_t, period_size ) + __field( snd_pcm_uframes_t, buffer_size ) + __field( snd_pcm_uframes_t, old_hw_ptr ) + __field( snd_pcm_uframes_t, hw_ptr_base ) + ), + TP_fast_assign( + __entry->card = (substream)->pcm->card->number; + __entry->device = (substream)->pcm->device; + __entry->number = (substream)->number; + __entry->stream = (substream)->stream; + __entry->period_size = (substream)->runtime->period_size; + __entry->buffer_size = (substream)->runtime->buffer_size; + __entry->old_hw_ptr = (substream)->runtime->status->hw_ptr; + __entry->hw_ptr_base = (substream)->runtime->hw_ptr_base; + ), + TP_printk("pcmC%dD%d%c/sub%d: XRUN: old=%lu, base=%lu, period=%lu, buf=%lu", + __entry->card, __entry->device, + __entry->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c', + __entry->number, + (unsigned long)__entry->old_hw_ptr, + (unsigned long)__entry->hw_ptr_base, + (unsigned long)__entry->period_size, + (unsigned long)__entry->buffer_size) +); + +TRACE_EVENT(hw_ptr_error, + TP_PROTO(struct snd_pcm_substream *substream, const char *why), + TP_ARGS(substream, why), + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( unsigned int, device ) + __field( unsigned int, number ) + __field( unsigned int, stream ) + __field( const char *, reason ) + ), + TP_fast_assign( + __entry->card = (substream)->pcm->card->number; + __entry->device = (substream)->pcm->device; + __entry->number = (substream)->number; + __entry->stream = (substream)->stream; + __entry->reason = (why); + ), + TP_printk("pcmC%dD%d%c/sub%d: ERROR: %s", + __entry->card, __entry->device, + __entry->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c', + __entry->number, __entry->reason) +); + +#endif /* _PCM_TRACE_H */ + +/* This part must be outside protection */ +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#include -- cgit v1.1 From 2b30d411dbc6eddfb5b4f9afd5a2c57b6f4dd96c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 4 Nov 2014 14:02:40 +0100 Subject: ALSA: pcm: Add xrun_injection proc entry This patch adds a new proc entry for PCM substreams to inject an XRUN. When a PCM substream is running and any value is written to its xrun_injection proc file, the driver triggers XRUN. This is a useful feature for debugging XRUN and error handling code paths. Note that this entry is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is set. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Procfile.txt | 4 ++++ include/sound/pcm.h | 3 +++ sound/core/pcm.c | 33 +++++++++++++++++++++++++++++++++ 3 files changed, 40 insertions(+) diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index cfc4956..7f8a0d3 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -133,6 +133,10 @@ card*/pcm*/sub*/sw_params card*/pcm*/sub*/prealloc The buffer pre-allocation information. +card*/pcm*/sub*/xrun_injection + Triggers an XRUN to the running stream when any value is + written to this proc file. Used for fault injection. + This entry is write-only. AC97 Codec Information ---------------------- diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 29eb09ef..0b8daee 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -416,7 +416,10 @@ struct snd_pcm_substream { struct snd_info_entry *proc_status_entry; struct snd_info_entry *proc_prealloc_entry; struct snd_info_entry *proc_prealloc_max_entry; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + struct snd_info_entry *proc_xrun_injection_entry; #endif +#endif /* CONFIG_SND_VERBOSE_PROCFS */ /* misc flags */ unsigned int hw_opened: 1; }; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 31acc3d..8f624b7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -483,6 +483,19 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, } #ifdef CONFIG_SND_PCM_XRUN_DEBUG +static void snd_pcm_xrun_injection_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcm_substream *substream = entry->private_data; + struct snd_pcm_runtime *runtime; + + snd_pcm_stream_lock_irq(substream); + runtime = substream->runtime; + if (runtime && runtime->status->state == SNDRV_PCM_STATE_RUNNING) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); +} + static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -614,6 +627,22 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) } substream->proc_status_entry = entry; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + entry = snd_info_create_card_entry(card, "xrun_injection", + substream->proc_root); + if (entry) { + entry->private_data = substream; + entry->c.text.read = NULL; + entry->c.text.write = snd_pcm_xrun_injection_write; + entry->mode = S_IFREG | S_IWUSR; + if (snd_info_register(entry) < 0) { + snd_info_free_entry(entry); + entry = NULL; + } + } + substream->proc_xrun_injection_entry = entry; +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ + return 0; } @@ -627,6 +656,10 @@ static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) substream->proc_sw_params_entry = NULL; snd_info_free_entry(substream->proc_status_entry); substream->proc_status_entry = NULL; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + snd_info_free_entry(substream->proc_xrun_injection_entry); + substream->proc_xrun_injection_entry = NULL; +#endif snd_info_free_entry(substream->proc_root); substream->proc_root = NULL; return 0; -- cgit v1.1 From 4c8c3a4fcc021677c9a363b4e77f61dd09dbfd1a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Oct 2014 11:00:23 +0100 Subject: ALSA: usb-audio: Flatten probe and disconnect functions The usb-audio probe and disconnect functions have been split just for adapting the (new!) API at 2.5 kernel time. We left them until now, partly because we wanted to build with the pretty old kernels in the external alsa-driver tree. But the support of such old kernels has been longly stopped, so it's good time to clean up this mess. One good point by this cleanup is that now the probe function returns a proper error code instead of only -EIO. Signed-off-by: Takashi Iwai --- sound/usb/card.c | 75 +++++++++++++++++++++++--------------------------------- 1 file changed, 30 insertions(+), 45 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 7ecd0e8..be16bdc 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -112,7 +112,7 @@ static struct usb_driver usb_audio_driver; /* * disconnect streams - * called from snd_usb_audio_disconnect() + * called from usb_audio_disconnect() */ static void snd_usb_stream_disconnect(struct list_head *head) { @@ -475,14 +475,14 @@ static int snd_usb_audio_create(struct usb_interface *intf, * only at the first time. the successive calls of this function will * append the pcm interface to the corresponding card. */ -static struct snd_usb_audio * -snd_usb_audio_probe(struct usb_device *dev, - struct usb_interface *intf, - const struct usb_device_id *usb_id) +static int usb_audio_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) { - const struct snd_usb_audio_quirk *quirk = (const struct snd_usb_audio_quirk *)usb_id->driver_info; - int i, err; + struct usb_device *dev = interface_to_usbdev(intf); + const struct snd_usb_audio_quirk *quirk = + (const struct snd_usb_audio_quirk *)usb_id->driver_info; struct snd_usb_audio *chip; + int i, err; struct usb_host_interface *alts; int ifnum; u32 id; @@ -492,10 +492,11 @@ snd_usb_audio_probe(struct usb_device *dev, id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) - goto __err_val; + return -ENXIO; - if (snd_usb_apply_boot_quirk(dev, intf, quirk) < 0) - goto __err_val; + err = snd_usb_apply_boot_quirk(dev, intf, quirk); + if (err < 0) + return err; /* * found a config. now register to ALSA @@ -508,6 +509,7 @@ snd_usb_audio_probe(struct usb_device *dev, if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { dev_err(&dev->dev, "USB device is in the shutdown state, cannot create a card instance\n"); + err = -EIO; goto __error; } chip = usb_chip[i]; @@ -523,15 +525,16 @@ snd_usb_audio_probe(struct usb_device *dev, if (enable[i] && ! usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - if (snd_usb_audio_create(intf, dev, i, quirk, - &chip) < 0) { + err = snd_usb_audio_create(intf, dev, i, quirk, + &chip); + if (err < 0) goto __error; - } chip->pm_intf = intf; break; } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); + err = -ENODEV; goto __error; } } @@ -548,28 +551,32 @@ snd_usb_audio_probe(struct usb_device *dev, err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ - if ((err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk)) < 0) + err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); + if (err < 0) goto __error; } if (err > 0) { /* create normal USB audio interfaces */ - if (snd_usb_create_streams(chip, ifnum) < 0 || - snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { + err = snd_usb_create_streams(chip, ifnum); + if (err < 0) + goto __error; + err = snd_usb_create_mixer(chip, ifnum, ignore_ctl_error); + if (err < 0) goto __error; - } } /* we are allowed to call snd_card_register() many times */ - if (snd_card_register(chip->card) < 0) { + err = snd_card_register(chip->card); + if (err < 0) goto __error; - } usb_chip[chip->index] = chip; chip->num_interfaces++; chip->probing = 0; + usb_set_intfdata(intf, chip); mutex_unlock(®ister_mutex); - return chip; + return 0; __error: if (chip) { @@ -578,17 +585,16 @@ snd_usb_audio_probe(struct usb_device *dev, chip->probing = 0; } mutex_unlock(®ister_mutex); - __err_val: - return NULL; + return err; } /* * we need to take care of counter, since disconnection can be called also * many times as well as usb_audio_probe(). */ -static void snd_usb_audio_disconnect(struct usb_device *dev, - struct snd_usb_audio *chip) +static void usb_audio_disconnect(struct usb_interface *intf) { + struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_card *card; struct list_head *p; @@ -630,27 +636,6 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, } } -/* - * new 2.5 USB kernel API - */ -static int usb_audio_probe(struct usb_interface *intf, - const struct usb_device_id *id) -{ - struct snd_usb_audio *chip; - chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id); - if (chip) { - usb_set_intfdata(intf, chip); - return 0; - } else - return -EIO; -} - -static void usb_audio_disconnect(struct usb_interface *intf) -{ - snd_usb_audio_disconnect(interface_to_usbdev(intf), - usb_get_intfdata(intf)); -} - #ifdef CONFIG_PM int snd_usb_autoresume(struct snd_usb_audio *chip) -- cgit v1.1 From a6cece9d81990e729c1f9da2a5bff2d29f7df649 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Oct 2014 11:24:32 +0100 Subject: ALSA: usb-audio: Pass direct struct pointer instead of list_head Some functions in mixer.c and endpoint.c receive list_head instead of the object itself. This is not obvious and rather error-prone. Let's pass the proper object directly instead. The functions in midi.c still receive list_head and this can't be changed since the object definition isn't exposed to the outside of midi.c, so left as is. Signed-off-by: Takashi Iwai --- sound/usb/card.c | 20 ++++++++++---------- sound/usb/endpoint.c | 7 ++----- sound/usb/endpoint.h | 2 +- sound/usb/mixer.c | 5 +---- sound/usb/mixer.h | 2 +- 5 files changed, 15 insertions(+), 21 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index be16bdc..fa6c097 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -114,13 +114,11 @@ static struct usb_driver usb_audio_driver; * disconnect streams * called from usb_audio_disconnect() */ -static void snd_usb_stream_disconnect(struct list_head *head) +static void snd_usb_stream_disconnect(struct snd_usb_stream *as) { int idx; - struct snd_usb_stream *as; struct snd_usb_substream *subs; - as = list_entry(head, struct snd_usb_stream, list); for (idx = 0; idx < 2; idx++) { subs = &as->substream[idx]; if (!subs->num_formats) @@ -307,10 +305,10 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) static int snd_usb_audio_free(struct snd_usb_audio *chip) { - struct list_head *p, *n; + struct snd_usb_endpoint *ep, *n; - list_for_each_safe(p, n, &chip->ep_list) - snd_usb_endpoint_free(p); + list_for_each_entry_safe(ep, n, &chip->ep_list, list) + snd_usb_endpoint_free(ep); mutex_destroy(&chip->mutex); kfree(chip); @@ -609,12 +607,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) mutex_lock(®ister_mutex); chip->num_interfaces--; if (chip->num_interfaces <= 0) { + struct snd_usb_stream *as; struct snd_usb_endpoint *ep; + struct usb_mixer_interface *mixer; snd_card_disconnect(card); /* release the pcm resources */ - list_for_each(p, &chip->pcm_list) { - snd_usb_stream_disconnect(p); + list_for_each_entry(as, &chip->pcm_list, list) { + snd_usb_stream_disconnect(as); } /* release the endpoint resources */ list_for_each_entry(ep, &chip->ep_list, list) { @@ -625,8 +625,8 @@ static void usb_audio_disconnect(struct usb_interface *intf) snd_usbmidi_disconnect(p); } /* release mixer resources */ - list_for_each(p, &chip->mixer_list) { - snd_usb_mixer_disconnect(p); + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_usb_mixer_disconnect(mixer); } usb_chip[chip->index] = NULL; mutex_unlock(®ister_mutex); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 114e3e7..167d0c1 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1002,15 +1002,12 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep) /** * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint * - * @ep: the list header of the endpoint to free + * @ep: the endpoint to free * * This free all resources of the given ep. */ -void snd_usb_endpoint_free(struct list_head *head) +void snd_usb_endpoint_free(struct snd_usb_endpoint *ep) { - struct snd_usb_endpoint *ep; - - ep = list_entry(head, struct snd_usb_endpoint, list); kfree(ep); } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index e61ee5c..6428392 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -24,7 +24,7 @@ void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); -void snd_usb_endpoint_free(struct list_head *head); +void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 63a8adb..e4aaa21 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2483,11 +2483,8 @@ _error: return err; } -void snd_usb_mixer_disconnect(struct list_head *p) +void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer) { - struct usb_mixer_interface *mixer; - - mixer = list_entry(p, struct usb_mixer_interface, list); usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 73b1f64..2c7b9c9 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -57,7 +57,7 @@ struct usb_mixer_elem_info { int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error); -void snd_usb_mixer_disconnect(struct list_head *p); +void snd_usb_mixer_disconnect(struct usb_mixer_interface *mixer); void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); -- cgit v1.1 From ae366c2049b48b54e5250cb57438bbebd1dbe610 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Oct 2014 11:32:19 +0100 Subject: ALSA: usb-audio: Use strim() instead of open code Signed-off-by: Takashi Iwai --- sound/usb/card.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fa6c097..69725d5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -321,16 +321,6 @@ static int snd_usb_audio_dev_free(struct snd_device *device) return snd_usb_audio_free(chip); } -static void remove_trailing_spaces(char *str) -{ - char *p; - - if (!*str) - return; - for (p = str + strlen(str) - 1; p >= str && isspace(*p); p--) - *p = 0; -} - /* * create a chip instance and set its names. */ @@ -414,7 +404,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, USB_ID_PRODUCT(chip->usb_id)); } } - remove_trailing_spaces(card->shortname); + strim(card->shortname); /* retrieve the vendor and device strings as longname */ if (quirk && quirk->vendor_name && *quirk->vendor_name) { @@ -428,7 +418,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, /* we don't really care if there isn't any vendor string */ } if (len > 0) { - remove_trailing_spaces(card->longname); + strim(card->longname); if (*card->longname) strlcat(card->longname, " ", sizeof(card->longname)); } -- cgit v1.1 From 7b8ef67a0b1edb37957a2aa71a5c0bbbcf2694e9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 4 Nov 2014 13:27:45 +0000 Subject: ASoC: Intel: Fix build with CONFIG_SLEEP enabled. Fix the following build error when CONFIG_SLEEP is enabled and CONFIG_RUNTIME is disabled. The BDW ADSP sleep PM functionality depends on the runtime pm calls for context save/restore. All error/warnings: >> ERROR: "snd_soc_suspend" undefined! >> ERROR: "snd_soc_resume" undefined! Reported-by: kbuild test robot Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 093b939..e7a3b6a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1003,7 +1003,6 @@ static int hsw_pcm_dev_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM #ifdef CONFIG_PM_RUNTIME static int hsw_pcm_runtime_idle(struct device *dev) @@ -1063,6 +1062,8 @@ static int hsw_pcm_runtime_resume(struct device *dev) #define hsw_pcm_runtime_resume NULL #endif +#if defined(CONFIG_PM_SLEEP) && defined(CONFIG_PM_RUNTIME) + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1158,6 +1159,11 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } +#else +#define hsw_pcm_prepare NULL +#define hsw_pcm_complete NULL +#endif + static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, @@ -1165,9 +1171,6 @@ static const struct dev_pm_ops hsw_pcm_pm = { .prepare = hsw_pcm_prepare, .complete = hsw_pcm_complete, }; -#else -#define hsw_pcm_pm NULL -#endif static struct platform_driver hsw_pcm_driver = { .driver = { -- cgit v1.1 From defcd98b16461e123cb4a6cb6ef24a1d0085c1b2 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Mon, 3 Nov 2014 10:28:57 -0800 Subject: ASoC: max98090: Different comp tables for different pclks In addtion expand the table to handle other values of sysclk. Instead of making the table 3D, expand it to a more descriptive struct. The divisors are specified in Table 19 of the 98090 data sheet version 0p94. The dmic frequency was previously assumed. Instead make it explicit and configurable through device tree. This now handles independently set pclk and dmic frequency. Based on downstream work by Ralph Birt. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/max98090.txt | 2 + sound/soc/codecs/max98090.c | 189 +++++++++++++++++---- sound/soc/codecs/max98090.h | 8 + 3 files changed, 163 insertions(+), 36 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index c454e67..aa802a2 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -16,6 +16,8 @@ Optional properties: - clock-names: Should be "mclk" +- maxim,dmic-freq: Frequency at which to clock DMIC + Pins on the device (for linking into audio routes): * MIC1 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1229554..a65861c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1826,27 +1826,155 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, return 0; } -static const int comp_pclk_rates[] = { - 11289600, 12288000, 12000000, 13000000, 19200000 -}; - -static const int dmic_micclk[] = { - 2, 2, 2, 2, 4, 2 -}; +static const int dmic_divisors[] = { 2, 3, 4, 5, 6, 8 }; static const int comp_lrclk_rates[] = { 8000, 16000, 32000, 44100, 48000, 96000 }; -static const int dmic_comp[6][6] = { - {7, 8, 3, 3, 3, 3}, - {7, 8, 3, 3, 3, 3}, - {7, 8, 3, 3, 3, 3}, - {7, 8, 3, 1, 1, 1}, - {7, 8, 3, 1, 2, 2}, - {7, 8, 3, 3, 3, 3} +struct dmic_table { + int pclk; + struct { + int freq; + int comp[6]; /* One each for 8, 16, 32, 44.1, 48, and 96 kHz */ + } settings[6]; /* One for each dmic divisor. */ }; +static const struct dmic_table dmic_table[] = { /* One for each pclk freq. */ + { + .pclk = 11289600, + .settings = { + { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + }, + }, + { + .pclk = 12000000, + .settings = { + { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + } + }, + { + .pclk = 12288000, + .settings = { + { .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } }, + } + }, + { + .pclk = 13000000, + .settings = { + { .freq = 2, .comp = { 7, 8, 1, 1, 1, 1 } }, + { .freq = 1, .comp = { 7, 8, 0, 0, 0, 0 } }, + { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } }, + { .freq = 0, .comp = { 7, 8, 4, 4, 5, 5 } }, + { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } }, + { .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } }, + } + }, + { + .pclk = 19200000, + .settings = { + { .freq = 2, .comp = { 0, 0, 0, 0, 0, 0 } }, + { .freq = 1, .comp = { 7, 8, 1, 1, 1, 1 } }, + { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } }, + { .freq = 0, .comp = { 7, 8, 2, 2, 3, 3 } }, + { .freq = 0, .comp = { 7, 8, 1, 1, 2, 2 } }, + { .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } }, + } + }, +}; + +static int max98090_find_divisor(int target_freq, int pclk) +{ + int current_diff = INT_MAX; + int test_diff = INT_MAX; + int divisor_index = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(dmic_divisors); i++) { + test_diff = abs(target_freq - (pclk / dmic_divisors[i])); + if (test_diff < current_diff) { + current_diff = test_diff; + divisor_index = i; + } + } + + return divisor_index; +} + +static int max98090_find_closest_pclk(int pclk) +{ + int m1; + int m2; + int i; + + for (i = 0; i < ARRAY_SIZE(dmic_table); i++) { + if (pclk == dmic_table[i].pclk) + return i; + if (pclk < dmic_table[i].pclk) { + if (i == 0) + return i; + m1 = pclk - dmic_table[i-1].pclk; + m2 = dmic_table[i].pclk - pclk; + if (m1 < m2) + return i - 1; + else + return i; + } + } + + return -EINVAL; +} + +static int max98090_configure_dmic(struct max98090_priv *max98090, + int target_dmic_clk, int pclk, int fs) +{ + int micclk_index; + int pclk_index; + int dmic_freq; + int dmic_comp; + int i; + + pclk_index = max98090_find_closest_pclk(pclk); + if (pclk_index < 0) + return pclk_index; + + micclk_index = max98090_find_divisor(target_dmic_clk, pclk); + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (fs <= (comp_lrclk_rates[i] + comp_lrclk_rates[i+1]) / 2) + break; + } + + dmic_freq = dmic_table[pclk_index].settings[micclk_index].freq; + dmic_comp = dmic_table[pclk_index].settings[micclk_index].comp[i]; + + regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_ENABLE, + M98090_MICCLK_MASK, + micclk_index << M98090_MICCLK_SHIFT); + + regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_MASK | M98090_DMIC_FREQ_MASK, + dmic_comp << M98090_DMIC_COMP_SHIFT | + dmic_freq << M98090_DMIC_FREQ_SHIFT); + + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1854,7 +1982,6 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; - int i, j; cdata = &max98090->dai[0]; max98090->bclk = snd_soc_params_to_bclk(params); @@ -1893,27 +2020,8 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, M98090_DHF_MASK, M98090_DHF_MASK); - /* Check for supported PCLK to LRCLK ratios */ - for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { - if (comp_pclk_rates[j] == max98090->sysclk) { - break; - } - } - - for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { - if (max98090->lrclk <= (comp_lrclk_rates[i] + - comp_lrclk_rates[i + 1]) / 2) { - break; - } - } - - snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE, - M98090_MICCLK_MASK, - dmic_micclk[j] << M98090_MICCLK_SHIFT); - - snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG, - M98090_DMIC_COMP_MASK, - dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + max98090_configure_dmic(max98090, max98090->dmic_freq, max98090->pclk, + max98090->lrclk); return 0; } @@ -1944,12 +2052,15 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, if ((freq >= 10000000) && (freq <= 20000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV1); + max98090->pclk = freq; } else if ((freq > 20000000) && (freq <= 40000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV2); + max98090->pclk = freq >> 1; } else if ((freq > 40000000) && (freq <= 60000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV4); + max98090->pclk = freq >> 2; } else { dev_err(codec->dev, "Invalid master clock frequency\n"); return -EINVAL; @@ -2324,6 +2435,7 @@ static int max98090_probe(struct snd_soc_codec *codec) /* Initialize private data */ max98090->sysclk = (unsigned)-1; + max98090->pclk = (unsigned)-1; max98090->master = false; cdata = &max98090->dai[0]; @@ -2463,6 +2575,11 @@ static int max98090_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, max98090); max98090->pdata = i2c->dev.platform_data; + ret = of_property_read_u32(i2c->dev.of_node, "maxim,dmic-freq", + &max98090->dmic_freq); + if (ret < 0) + max98090->dmic_freq = MAX98090_DEFAULT_DMIC_FREQ; + max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { ret = PTR_ERR(max98090->regmap); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index a5f6bad..21ff743 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -12,6 +12,12 @@ #define _MAX98090_H /* + * The default operating frequency for a DMIC attached to the codec. + * This can be overridden by a device tree property. + */ +#define MAX98090_DEFAULT_DMIC_FREQ 2500000 + +/* * MAX98090 Register Definitions */ @@ -1518,8 +1524,10 @@ struct max98090_priv { struct max98090_pdata *pdata; struct clk *mclk; unsigned int sysclk; + unsigned int pclk; unsigned int bclk; unsigned int lrclk; + u32 dmic_freq; struct max98090_cdata dai[1]; int jack_state; struct delayed_work jack_work; -- cgit v1.1 From 9161bd0d1cf375492f0a6aa86b3e4c28b070fb7c Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Wed, 5 Nov 2014 19:51:56 +0530 Subject: ALSA: echoaudio: cleanup of unnecessary messages commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function, and which switch-case it is executing. we can easily get from ftrace the information about the entry and exit from a function. Reported-by: Dan Carpenter Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 31 ------------------------------- sound/pci/echoaudio/echoaudio_3g.c | 5 ----- sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 6 ------ sound/pci/echoaudio/gina24_dsp.c | 9 --------- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 9 --------- sound/pci/echoaudio/layla24_dsp.c | 8 -------- sound/pci/echoaudio/mia_dsp.c | 2 -- sound/pci/echoaudio/midi.c | 6 ------ sound/pci/echoaudio/mona_dsp.c | 6 ------ 18 files changed, 110 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..60e4003 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -79,7 +79,6 @@ static void free_firmware(const struct firmware *fw_entry, dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); #else release_firmware(fw_entry); - dev_dbg(chip->card->dev, "firmware released\n"); #endif } @@ -96,7 +95,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +352,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +386,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +418,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +459,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +505,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +613,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +683,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +754,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +777,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +787,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -937,7 +923,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -950,7 +935,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -972,7 +956,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -987,7 +970,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1881,10 +1863,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1878,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1893,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2008,7 +1985,6 @@ static int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - dev_dbg(card->dev, "Card init OK\n"); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_echo_free(chip); @@ -2038,7 +2014,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2166,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2192,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2205,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2219,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2231,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2256,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..2fa66dc 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -337,13 +336,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n"); control_reg |= E3G_SPDIF_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -353,12 +350,10 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n"); control_reg |= E3G_ADAT_CLOCK; control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n"); control_reg |= E3G_WORD_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96) control_reg |= E3G_DOUBLE_SPEED_MODE; diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..5dafe92 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -159,7 +156,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) chip->spdif_status = GD_SPDIF_STATUS_UNDEF; set_sample_rate(chip, chip->sample_rate); chip->input_clock = clock; - dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n"); break; case ECHO_CLOCK_SPDIF: chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN; @@ -167,7 +163,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) clear_handshake(chip); send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE); chip->clock_state = GD_CLOCK_SPDIFIN; - dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n"); chip->input_clock = clock; break; default: @@ -209,7 +204,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..6971766 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -247,13 +243,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -263,17 +257,14 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n"); control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n"); control_reg |= GML_ESYNC_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC96: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n"); control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE; break; default: diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..f2024a3 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,24 +188,19 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n"); rate = chip->sample_rate; clock = LAYLA20_CLOCK_INTERNAL; break; case ECHO_CLOCK_SPDIF: - dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n"); clock = LAYLA20_CLOCK_SPDIF; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n"); clock = LAYLA20_CLOCK_WORD; break; case ECHO_CLOCK_SUPER: - dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n"); clock = LAYLA20_CLOCK_SUPER; break; default: @@ -232,7 +225,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +278,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..4f11e81 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -262,7 +258,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Pick the new clock */ switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -271,7 +266,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_SPDIF_CLOCK; /* Layla24 doesn't support 96KHz S/PDIF */ control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n"); break; case ECHO_CLOCK_WORD: control_reg |= GML_WORD_CLOCK; @@ -279,14 +273,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_DOUBLE_SPEED_MODE; else control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n"); break; case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n"); break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index ed2f21d..fdad079 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mia\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA)) return -ENODEV; @@ -67,7 +66,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8d43c5a..d913749 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -157,7 +157,6 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = substream; - dev_dbg(chip->card->dev, "rawmidi_iopen\n"); return 0; } @@ -183,7 +182,6 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = NULL; - dev_dbg(chip->card->dev, "rawmidi_iclose\n"); return 0; } @@ -196,7 +194,6 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) chip->tinuse = 0; chip->midi_full = 0; chip->midi_out = substream; - dev_dbg(chip->card->dev, "rawmidi_open\n"); return 0; } @@ -209,7 +206,6 @@ static void snd_echo_midi_output_write(unsigned long data) int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; - dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n"); /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ sent = bytes = 0; @@ -288,7 +284,6 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_out = NULL; - dev_dbg(chip->card->dev, "rawmidi_oclose\n"); return 0; } @@ -328,6 +323,5 @@ static int snd_echo_midi_create(struct snd_card *card, chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; - dev_dbg(chip->card->dev, "MIDI ok\n"); return 0; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..843c7a5 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) @@ -315,7 +312,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -327,7 +323,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) spin_lock_irq(&chip->lock); if (err < 0) return err; - dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -335,7 +330,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Mona clock to WORD\n"); spin_unlock_irq(&chip->lock); err = switch_asic(chip, clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96); -- cgit v1.1 From 0c239fa6ebd20dd55d8978502d78b7c17441351a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:31 +0100 Subject: ASoC: sn95031: Use table based control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index cf8fa40..6167c59 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -867,9 +867,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_codec_controls(codec, sn95031_snd_controls, - ARRAY_SIZE(sn95031_snd_controls)); - return 0; } @@ -886,6 +883,9 @@ static struct snd_soc_codec_driver sn95031_codec = { .remove = sn95031_codec_remove, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, + + .controls = sn95031_snd_controls, + .num_controls = ARRAY_SIZE(sn95031_snd_controls), .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, -- cgit v1.1 From e3f1ff318e78990977dae91f7f17f02e9af38e7d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:32 +0100 Subject: ASoC: tas2552: Use table based DAPM setup Makes the code a bit cleaner and shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index f039dc8..b505212 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -345,7 +345,6 @@ static const struct reg_default tas2552_init_regs[] = { static int tas2552_codec_probe(struct snd_soc_codec *codec) { struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; tas2552->codec = codec; @@ -390,11 +389,6 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TAS2552_CFG_2, TAS2552_BOOST_EN | TAS2552_APT_EN | TAS2552_LIM_EN); - snd_soc_dapm_new_controls(dapm, tas2552_dapm_widgets, - ARRAY_SIZE(tas2552_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, tas2552_audio_map, - ARRAY_SIZE(tas2552_audio_map)); - return 0; patch_fail: @@ -462,6 +456,10 @@ static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { .resume = tas2552_resume, .controls = tas2552_snd_controls, .num_controls = ARRAY_SIZE(tas2552_snd_controls), + .dapm_widgets = tas2552_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2552_dapm_widgets), + .dapm_routes = tas2552_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2552_audio_map), }; static const struct regmap_config tas2552_regmap_config = { -- cgit v1.1 From c8b5d089d6fd614dfc8a04e3cf087c97486898fb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:33 +0100 Subject: ASoC: wl1273: Use table based control setup Makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index f3d4e88..00aea41 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -452,7 +452,6 @@ static int wl1273_probe(struct snd_soc_codec *codec) { struct wl1273_core **core = codec->dev->platform_data; struct wl1273_priv *wl1273; - int r; dev_dbg(codec->dev, "%s.\n", __func__); @@ -470,12 +469,7 @@ static int wl1273_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, wl1273); - r = snd_soc_add_codec_controls(codec, wl1273_controls, - ARRAY_SIZE(wl1273_controls)); - if (r) - kfree(wl1273); - - return r; + return 0; } static int wl1273_remove(struct snd_soc_codec *codec) @@ -492,6 +486,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + .controls = wl1273_controls, + .num_controls = ARRAY_SIZE(wl1273_controls), .dapm_widgets = wl1273_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), .dapm_routes = wl1273_dapm_routes, -- cgit v1.1 From a6bf30698825718f22a689a54ea023cdf51a4c76 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:34 +0100 Subject: ASoC: wm8737: Use table based DAPM and control setup Makes the code a bit cleaner and shorter. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 22 +++++++--------------- 1 file changed, 7 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 744a422..fe41dd2 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -277,17 +277,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF", NULL, "ADCR" }, }; -static int wm8737_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8737_dapm_widgets, - ARRAY_SIZE(wm8737_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - /* codec mclk clock divider coefficients */ static const struct { u32 mclk; @@ -593,10 +582,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - snd_soc_add_codec_controls(codec, wm8737_snd_controls, - ARRAY_SIZE(wm8737_snd_controls)); - wm8737_add_widgets(codec); - return 0; err_enable: @@ -617,6 +602,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .suspend = wm8737_suspend, .resume = wm8737_resume, .set_bias_level = wm8737_set_bias_level, + + .controls = wm8737_snd_controls, + .num_controls = ARRAY_SIZE(wm8737_snd_controls), + .dapm_widgets = wm8737_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8737_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static const struct of_device_id wm8737_of_match[] = { -- cgit v1.1 From c4f50dbc56580bc5fc84667860e973ca24291697 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:35 +0100 Subject: ASoC: wm8961: Use table based DAPM and control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 41d23e9..e077bb2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -835,7 +835,6 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; u16 reg; /* Enable class W */ @@ -873,12 +872,6 @@ static int wm8961_probe(struct snd_soc_codec *codec) wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, wm8961_snd_controls, - ARRAY_SIZE(wm8961_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets, - ARRAY_SIZE(wm8961_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return 0; } @@ -915,6 +908,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, + + .controls = wm8961_snd_controls, + .num_controls = ARRAY_SIZE(wm8961_snd_controls), + .dapm_widgets = wm8961_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8961_dapm_widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), }; static const struct regmap_config wm8961_regmap = { -- cgit v1.1 From b131c02e99b9da672a2b0cf96bad48d74c39572e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 10:46:36 +0100 Subject: ASoC: wm8995: Use table based DAPM and control setup Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 1288ede..e40c8a6 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2102,13 +2102,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995_update_class_w(codec); - snd_soc_add_codec_controls(codec, wm8995_snd_controls, - ARRAY_SIZE(wm8995_snd_controls)); - snd_soc_dapm_new_controls(&codec->dapm, wm8995_dapm_widgets, - ARRAY_SIZE(wm8995_dapm_widgets)); - snd_soc_dapm_add_routes(&codec->dapm, wm8995_intercon, - ARRAY_SIZE(wm8995_intercon)); - return 0; err_reg_enable: @@ -2205,6 +2198,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .remove = wm8995_remove, .set_bias_level = wm8995_set_bias_level, .idle_bias_off = true, + + .controls = wm8995_snd_controls, + .num_controls = ARRAY_SIZE(wm8995_snd_controls), + .dapm_widgets = wm8995_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8995_dapm_widgets), + .dapm_routes = wm8995_intercon, + .num_dapm_routes = ARRAY_SIZE(wm8995_intercon), }; static struct regmap_config wm8995_regmap = { -- cgit v1.1 From d65fd3a42e00d322448f2518db6a3f0eb12ce1bd Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 5 Nov 2014 13:42:52 +0800 Subject: ASoC: rt5677: Minor coding style and typo fix Minor coding style and typo fix Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677-spi.h | 4 ++-- sound/soc/codecs/rt5677.c | 14 +++++++------- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index 7528bfd..ec41b2b 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -9,8 +9,8 @@ * published by the Free Software Foundation. */ -#ifndef __RT5671_SPI_H__ -#define __RT5671_SPI_H__ +#ifndef __RT5677_SPI_H__ +#define __RT5677_SPI_H__ #define RT5677_SPI_BUF_LEN 240 #define RT5677_SPI_CMD_BURST_WRITE 0x05 diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ca264f8..0d24dc4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1353,7 +1353,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum); -/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ +/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ static const char * const rt5677_stereo_adc2_src[] = { "DD MIX1", "DMIC", "Stereo DAC MIX" }; @@ -1438,7 +1438,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum); -/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ +/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ static const char * const rt5677_stereo_adc1_src[] = { "DD MIX1", "ADC1/2", "Stereo DAC MIX" }; @@ -1710,7 +1710,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_pdm2_r_mux = SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum); -/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ +/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0] */ static const char * const rt5677_if12_adc1_src[] = { "STO1 ADC MIX", "OB01", "VAD ADC" }; @@ -1788,7 +1788,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc3_mux = SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum); -/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ +/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ static const char * const rt5677_if12_adc4_src[] = { "STO4 ADC MIX", "OB67", "OB01" }; @@ -1814,7 +1814,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc4_mux = SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum); -/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ +/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4] */ static const char * const rt5677_if34_adc_src[] = { "STO1 ADC MIX", "STO2 ADC MIX", "STO3 ADC MIX", "STO4 ADC MIX", "MONO ADC MIX", "OB01", "OB23", "VAD ADC" @@ -1895,7 +1895,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); -/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ static const char * const rt5677_if1_adc_tdm_swap_src[] = { "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", "3/1/2/4", "3/4/1/2" @@ -2442,7 +2442,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { rt5677_ob_7_mix, ARRAY_SIZE(rt5677_ob_7_mix)), /* Output Side */ - /* DAC mixer before sound effect */ + /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5677_dac_l_mix, ARRAY_SIZE(rt5677_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, -- cgit v1.1 From 1b86a3fa4eb3c7a6d738fa21475b92493f8952b1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Nov 2014 17:19:53 +0100 Subject: ASoC: ad193x: Keep DAC output stage active in idle Setting the DAC power-down bit for the ad193x will also disable the DAC output amplifier. This will cause audible clicks and pops when starting or stopping playback. To prevent this a new widget is introduced that controls the DAC power-down bit. This widget is connected to both the DAC and a newly introduced VMID widget. This makes sure that the DAC power-down bit is not set as long as a audio sink is connected to the DAC output. At the same time the PLL and SYSCLK will still be disabled when no playback or capture stream is active. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 6844d0b..387530b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -72,11 +72,13 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), + SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), @@ -87,13 +89,15 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, + { "DAC Output", NULL, "DAC" }, + { "DAC Output", NULL, "VMID" }, { "ADC", NULL, "SYSCLK" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", NULL, "DAC" }, - { "DAC2OUT", NULL, "DAC" }, - { "DAC3OUT", NULL, "DAC" }, - { "DAC4OUT", NULL, "DAC" }, + { "DAC1OUT", NULL, "DAC Output" }, + { "DAC2OUT", NULL, "DAC Output" }, + { "DAC3OUT", NULL, "DAC Output" }, + { "DAC4OUT", NULL, "DAC Output" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, { "SYSCLK", NULL, "PLL_PWR" }, -- cgit v1.1 From 6c67cde2aa88bb06cd039aa0f61b26df887075d7 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 6 Nov 2014 09:59:59 +0800 Subject: ASoC: rt286: set combo jack by dmi This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..d4acb64 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id force_combo_jack_table[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(force_combo_jack_table)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- cgit v1.1 From f8c101bc357d509291f6accb6f62b8439158a203 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 6 Nov 2014 10:00:00 +0800 Subject: ASoC: rt286: fix comment style Adds spaces around the /* */. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index d4acb64..2e818aa 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -191,7 +191,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) u8 data[4]; int ret, i; - /*handle index registers*/ + /* handle index registers */ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); for (i = 0; i < INDEX_CACHE_SIZE; i++) { @@ -234,7 +234,7 @@ static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) __be32 be_reg; unsigned int index, vid, buf = 0x0; - /*handle index registers*/ + /* handle index registers */ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); reg = RT286_PROC_COEF; @@ -1281,11 +1281,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); - /*Power down LDO, VREF*/ + /* Power down LDO, VREF */ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); - /*Set depop parameter*/ + /* Set depop parameter */ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); -- cgit v1.1 From 9bc889b4ba88a3f2d81e4b799d47d71d7381573a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Nov 2014 12:15:25 +0100 Subject: ALSA: pcm: Update the state properly before notification Some state changes (e.g. snd_pcm_stop()) sets the runtime state after calling snd_timer_notify(). This is basically racy, since the notification may wakes up the user even before the state change. Although the possibility is low, we should set the state before the notifications. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ca224fa..dfb5031 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1052,10 +1052,10 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); + runtime->status->state = state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, &runtime->trigger_tstamp); - runtime->status->state = state; } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1204,11 +1204,11 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->suspended_state = runtime->status->state; + runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, &runtime->trigger_tstamp); - runtime->status->suspended_state = runtime->status->state; - runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1311,10 +1311,10 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->state = runtime->status->suspended_state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, &runtime->trigger_tstamp); - runtime->status->state = runtime->status->suspended_state; } static struct action_ops snd_pcm_action_resume = { -- cgit v1.1 From 67e225009bb15403341d313f51326113c61af7df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Nov 2014 13:04:49 +0100 Subject: ALSA: usb-audio: Trigger PCM XRUN at XRUN The usb-audio driver detects XRUN at its complete callback, but the actual code to trigger PCM XRUN is commented out because it caused deadlock in the past. This patch revives the PCM trigger properly. It resulted in more than just enabling snd_pcm_stop(), but it had to deduce the PCM substream with proper NULL checks and holds the stream lock around the call. Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 167d0c1..a467991 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -348,6 +348,8 @@ static void snd_complete_urb(struct urb *urb) { struct snd_urb_ctx *ctx = urb->context; struct snd_usb_endpoint *ep = ctx->ep; + struct snd_pcm_substream *substream; + unsigned long flags; int err; if (unlikely(urb->status == -ENOENT || /* unlinked */ @@ -364,8 +366,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; if (snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned long flags; - spin_lock_irqsave(&ep->lock, flags); list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); spin_unlock_irqrestore(&ep->lock, flags); @@ -389,7 +389,12 @@ static void snd_complete_urb(struct urb *urb) return; usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); - //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + if (ep->data_subs && ep->data_subs->pcm_substream) { + substream = ep->data_subs->pcm_substream; + snd_pcm_stream_lock_irqsave(substream, flags); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + } exit_clear: clear_bit(ctx->index, &ep->active_mask); -- cgit v1.1 From bb656add19764c7a3cf28b2b330ec0a189fe4f48 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 5 Nov 2014 15:02:08 +0800 Subject: ASoC: rt5645: Add JD function support rt5645 codec support jack detection function. The patch will set related registers if JD function is used. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5645.h | 3 +++ sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt5645.h | 5 +++++ 3 files changed, 28 insertions(+) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a535271..937f421 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,9 @@ struct rt5645_platform_data { unsigned int hp_det_gpio; bool gpio_hp_det_active_high; + + /* true if codec's jd function is used */ + bool en_jd_func; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1423cb2..286438d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2203,6 +2203,13 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + /* for JD function */ + if (rt5645->pdata.en_jd_func) { + snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_sync(&codec->dapm); + } + return 0; } @@ -2436,6 +2443,19 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } + if (rt5645->pdata.en_jd_func) { + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); + regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 5ec2520..82f681b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1348,6 +1348,8 @@ #define RT5645_PWR_CLK25M_SFT 4 #define RT5645_PWR_CLK25M_PD (0x0 << 4) #define RT5645_PWR_CLK25M_PU (0x1 << 4) +#define RT5645_IRQ_CLK_MCLK (0x0 << 3) +#define RT5645_IRQ_CLK_INT (0x1 << 3) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -2116,6 +2118,9 @@ enum { #define RT5645_RXDP2_SEL_ADC (0x1 << 3) #define RT5645_RXDP2_SEL_SFT (3) +/* General Control3 (0xfc) */ +#define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) +#define RT5645_MICINDET_MANU (0x1 << 7) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- cgit v1.1 From 29e1812d761183a6dd27c53d1259169e9e7ba4e2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Nov 2014 16:25:15 +0530 Subject: ASoC: Intel: mrfld - remove unnecessary check for pointer the 'platform' pointer in sst_map_modules_to_pipe() is deref in caller function so we need to check for it in this function Reported-by: Dan Carpenter Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c index 309a8f3..90aa5c0 100644 --- a/sound/soc/intel/sst-atom-controls.c +++ b/sound/soc/intel/sst-atom-controls.c @@ -1351,7 +1351,7 @@ static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) int ret = 0; list_for_each_entry(w, &platform->component.card->widgets, list) { - if (platform && is_sst_dapm_widget(w) && (w->priv)) { + if (is_sst_dapm_widget(w) && (w->priv)) { struct sst_ids *ids = w->priv; dev_dbg(platform->dev, "widget type=%d name=%s\n", -- cgit v1.1 From f533a035e4da2fdd5e7b0100c84b62fd73ecd6c7 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Nov 2014 16:25:16 +0530 Subject: ASoC: Intel: mrfld - create separate module for pci part Now the SST_IPC will support both ACPI and PCI, separate into core module and PCI module. This also move probe function into PCI module and exports the required symbols from core module Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 6 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 212 ++++-------------------------------------- sound/soc/intel/sst/sst.h | 6 ++ sound/soc/intel/sst/sst_pci.c | 209 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 1 + 6 files changed, 243 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/sst/sst_pci.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index ae7f872..c963a5d 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -3,7 +3,7 @@ config SND_MFLD_MACHINE depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_MFLD_PLATFORM - select SND_SST_IPC + select SND_SST_IPC_PCI help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices @@ -16,6 +16,10 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_IPC_PCI + tristate + select SND_SST_IPC + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile index 4d0e79b..b8aa1d3 100644 --- a/sound/soc/intel/sst/Makefile +++ b/sound/soc/intel/sst/Makefile @@ -1,3 +1,7 @@ -snd-intel-sst-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o +snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o +snd-intel-sst-pci-objs += sst_pci.o + + +obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o +obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o -obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst.o diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 2bfb404..8753754 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -20,20 +20,14 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ #include -#include #include #include #include #include #include #include -#include -#include #include -#include #include -#include -#include #include #include "../sst-mfld-platform.h" #include "sst.h" @@ -242,6 +236,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, return 0; } +EXPORT_SYMBOL_GPL(sst_alloc_drv_context); int sst_context_init(struct intel_sst_drv *ctx) { @@ -260,6 +255,7 @@ int sst_context_init(struct intel_sst_drv *ctx) return -EINVAL; sst_init_locks(ctx); + sst_set_fw_state_locked(ctx, SST_RESET); /* pvt_id 0 reserved for async messages */ ctx->pvt_id = 1; @@ -307,12 +303,22 @@ int sst_context_init(struct intel_sst_drv *ctx) } pm_qos_add_request(ctx->qos, PM_QOS_CPU_DMA_LATENCY, PM_QOS_DEFAULT_VALUE); + + dev_dbg(ctx->dev, "Requesting FW %s now...\n", ctx->firmware_name); + ret = request_firmware_nowait(THIS_MODULE, true, ctx->firmware_name, + ctx->dev, GFP_KERNEL, ctx, sst_firmware_load_cb); + if (ret) { + dev_err(ctx->dev, "Firmware download failed:%d\n", ret); + goto do_free_mem; + } + sst_register(ctx->dev); return 0; do_free_mem: destroy_workqueue(ctx->post_msg_wq); return ret; } +EXPORT_SYMBOL_GPL(sst_context_init); void sst_context_cleanup(struct intel_sst_drv *ctx) { @@ -331,6 +337,7 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) sst_memcpy_free_resources(ctx); ctx = NULL; } +EXPORT_SYMBOL_GPL(sst_context_cleanup); void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { @@ -339,175 +346,7 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) pm_runtime_allow(ctx->dev); pm_runtime_put_noidle(ctx->dev); } - -static int sst_platform_get_resources(struct intel_sst_drv *ctx) -{ - int ddr_base, ret = 0; - struct pci_dev *pci = ctx->pci; - ret = pci_request_regions(pci, SST_DRV_NAME); - if (ret) - return ret; - - /* map registers */ - /* DDR base */ - if (ctx->dev_id == SST_MRFLD_PCI_ID) { - ctx->ddr_base = pci_resource_start(pci, 0); - /* check that the relocated IMR base matches with FW Binary */ - ddr_base = relocate_imr_addr_mrfld(ctx->ddr_base); - if (!ctx->pdata->lib_info) { - dev_err(ctx->dev, "lib_info pointer NULL\n"); - ret = -EINVAL; - goto do_release_regions; - } - if (ddr_base != ctx->pdata->lib_info->mod_base) { - dev_err(ctx->dev, - "FW LSP DDR BASE does not match with IFWI\n"); - ret = -EINVAL; - goto do_release_regions; - } - ctx->ddr_end = pci_resource_end(pci, 0); - - ctx->ddr = pcim_iomap(pci, 0, - pci_resource_len(pci, 0)); - if (!ctx->ddr) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "sst: DDR Ptr %p\n", ctx->ddr); - } else { - ctx->ddr = NULL; - } - /* SHIM */ - ctx->shim_phy_add = pci_resource_start(pci, 1); - ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); - if (!ctx->shim) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "SST Shim Ptr %p\n", ctx->shim); - - /* Shared SRAM */ - ctx->mailbox_add = pci_resource_start(pci, 2); - ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); - if (!ctx->mailbox) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "SRAM Ptr %p\n", ctx->mailbox); - - /* IRAM */ - ctx->iram_end = pci_resource_end(pci, 3); - ctx->iram_base = pci_resource_start(pci, 3); - ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); - if (!ctx->iram) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "IRAM Ptr %p\n", ctx->iram); - - /* DRAM */ - ctx->dram_end = pci_resource_end(pci, 4); - ctx->dram_base = pci_resource_start(pci, 4); - ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); - if (!ctx->dram) { - ret = -EINVAL; - goto do_release_regions; - } - dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); -do_release_regions: - pci_release_regions(pci); - return 0; -} -/* -* intel_sst_probe - PCI probe function -* -* @pci: PCI device structure -* @pci_id: PCI device ID structure -* -*/ -static int intel_sst_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) -{ - int ret = 0; - struct intel_sst_drv *sst_drv_ctx; - struct sst_platform_info *sst_pdata = pci->dev.platform_data; - - dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); - - ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); - if (ret < 0) - return ret; - - sst_drv_ctx->pdata = sst_pdata; - sst_drv_ctx->irq_num = pci->irq; - - ret = sst_context_init(sst_drv_ctx); - if (ret < 0) - goto do_free_drv_ctx; - - - /* Init the device */ - ret = pcim_enable_device(pci); - if (ret) { - dev_err(sst_drv_ctx->dev, - "device can't be enabled. Returned err: %d\n", ret); - goto do_destroy_wq; - } - sst_drv_ctx->pci = pci_dev_get(pci); - - ret = sst_platform_get_resources(sst_drv_ctx); - if (ret < 0) - goto do_destroy_wq; - - sst_set_fw_state_locked(sst_drv_ctx, SST_RESET); - snprintf(sst_drv_ctx->firmware_name, sizeof(sst_drv_ctx->firmware_name), - "%s%04x%s", "fw_sst_", - sst_drv_ctx->dev_id, ".bin"); - dev_dbg(sst_drv_ctx->dev, - "Requesting FW %s now...\n", sst_drv_ctx->firmware_name); - ret = request_firmware_nowait(THIS_MODULE, 1, - sst_drv_ctx->firmware_name, sst_drv_ctx->dev, - GFP_KERNEL, sst_drv_ctx, sst_firmware_load_cb); - - if (ret) { - dev_err(sst_drv_ctx->dev, - "Firmware load failed with error: %d\n", ret); - goto do_release_regions; - } - - - pci_set_drvdata(pci, sst_drv_ctx); - sst_configure_runtime_pm(sst_drv_ctx); - sst_register(sst_drv_ctx->dev); - - return ret; - -do_release_regions: - pci_release_regions(pci); -do_destroy_wq: - destroy_workqueue(sst_drv_ctx->post_msg_wq); -do_free_drv_ctx: - dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); - return ret; -} - -/** -* intel_sst_remove - PCI remove function -* -* @pci: PCI device structure -* -* This function is called by OS when a device is unloaded -* This frees the interrupt etc -*/ -static void intel_sst_remove(struct pci_dev *pci) -{ - struct intel_sst_drv *sst_drv_ctx = pci_get_drvdata(pci); - - sst_context_cleanup(sst_drv_ctx); - pci_dev_put(sst_drv_ctx->pci); - pci_release_regions(pci); - pci_set_drvdata(pci, NULL); -} +EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); static int intel_sst_runtime_suspend(struct device *dev) { @@ -546,27 +385,8 @@ static int intel_sst_runtime_resume(struct device *dev) return ret; } -static const struct dev_pm_ops intel_sst_pm = { +const struct dev_pm_ops intel_sst_pm = { .runtime_suspend = intel_sst_runtime_suspend, .runtime_resume = intel_sst_runtime_resume, }; - -/* PCI Routines */ -static struct pci_device_id intel_sst_ids[] = { - { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, - { 0, } -}; - -static struct pci_driver sst_driver = { - .name = SST_DRV_NAME, - .id_table = intel_sst_ids, - .probe = intel_sst_probe, - .remove = intel_sst_remove, -#ifdef CONFIG_PM - .driver = { - .pm = &intel_sst_pm, - }, -#endif -}; - -module_pci_driver(sst_driver); +EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index b65b9c0..3ee555e 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -40,6 +40,7 @@ #define MRFLD_FW_FEATURE_BASE_OFFSET 0x4 #define MRFLD_FW_BSS_RESET_BIT 0 +extern const struct dev_pm_ops intel_sst_pm; enum sst_states { SST_FW_LOADING = 1, SST_FW_RUNNING, @@ -537,4 +538,9 @@ void sst_fill_header_dsp(struct ipc_dsp_hdr *dsp, int msg, int sst_register(struct device *); int sst_unregister(struct device *); +int sst_alloc_drv_context(struct intel_sst_drv **ctx, + struct device *dev, unsigned int dev_id); +int sst_context_init(struct intel_sst_drv *ctx); +void sst_context_cleanup(struct intel_sst_drv *ctx); +void sst_configure_runtime_pm(struct intel_sst_drv *ctx); #endif diff --git a/sound/soc/intel/sst/sst_pci.c b/sound/soc/intel/sst/sst_pci.c new file mode 100644 index 0000000..3a0b3bf --- /dev/null +++ b/sound/soc/intel/sst/sst_pci.c @@ -0,0 +1,209 @@ +/* + * sst_pci.c - SST (LPE) driver init file for pci enumeration. + * + * Copyright (C) 2008-14 Intel Corp + * Authors: Vinod Koul + * Harsha Priya + * Dharageswari R + * KP Jeeja + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "sst.h" + +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + int ddr_base, ret = 0; + struct pci_dev *pci = ctx->pci; + + ret = pci_request_regions(pci, SST_DRV_NAME); + if (ret) + return ret; + + /* map registers */ + /* DDR base */ + if (ctx->dev_id == SST_MRFLD_PCI_ID) { + ctx->ddr_base = pci_resource_start(pci, 0); + /* check that the relocated IMR base matches with FW Binary */ + ddr_base = relocate_imr_addr_mrfld(ctx->ddr_base); + if (!ctx->pdata->lib_info) { + dev_err(ctx->dev, "lib_info pointer NULL\n"); + ret = -EINVAL; + goto do_release_regions; + } + if (ddr_base != ctx->pdata->lib_info->mod_base) { + dev_err(ctx->dev, + "FW LSP DDR BASE does not match with IFWI\n"); + ret = -EINVAL; + goto do_release_regions; + } + ctx->ddr_end = pci_resource_end(pci, 0); + + ctx->ddr = pcim_iomap(pci, 0, + pci_resource_len(pci, 0)); + if (!ctx->ddr) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(ctx->dev, "sst: DDR Ptr %p\n", ctx->ddr); + } else { + ctx->ddr = NULL; + } + /* SHIM */ + ctx->shim_phy_add = pci_resource_start(pci, 1); + ctx->shim = pcim_iomap(pci, 1, pci_resource_len(pci, 1)); + if (!ctx->shim) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(ctx->dev, "SST Shim Ptr %p\n", ctx->shim); + + /* Shared SRAM */ + ctx->mailbox_add = pci_resource_start(pci, 2); + ctx->mailbox = pcim_iomap(pci, 2, pci_resource_len(pci, 2)); + if (!ctx->mailbox) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(ctx->dev, "SRAM Ptr %p\n", ctx->mailbox); + + /* IRAM */ + ctx->iram_end = pci_resource_end(pci, 3); + ctx->iram_base = pci_resource_start(pci, 3); + ctx->iram = pcim_iomap(pci, 3, pci_resource_len(pci, 3)); + if (!ctx->iram) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(ctx->dev, "IRAM Ptr %p\n", ctx->iram); + + /* DRAM */ + ctx->dram_end = pci_resource_end(pci, 4); + ctx->dram_base = pci_resource_start(pci, 4); + ctx->dram = pcim_iomap(pci, 4, pci_resource_len(pci, 4)); + if (!ctx->dram) { + ret = -EINVAL; + goto do_release_regions; + } + dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); +do_release_regions: + pci_release_regions(pci); + return 0; +} + +/* + * intel_sst_probe - PCI probe function + * + * @pci: PCI device structure + * @pci_id: PCI device ID structure + * + */ +static int intel_sst_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + int ret = 0; + struct intel_sst_drv *sst_drv_ctx; + struct sst_platform_info *sst_pdata = pci->dev.platform_data; + + dev_dbg(&pci->dev, "Probe for DID %x\n", pci->device); + ret = sst_alloc_drv_context(&sst_drv_ctx, &pci->dev, pci->device); + if (ret < 0) + return ret; + + sst_drv_ctx->pdata = sst_pdata; + sst_drv_ctx->irq_num = pci->irq; + snprintf(sst_drv_ctx->firmware_name, sizeof(sst_drv_ctx->firmware_name), + "%s%04x%s", "fw_sst_", + sst_drv_ctx->dev_id, ".bin"); + + ret = sst_context_init(sst_drv_ctx); + if (ret < 0) + return ret; + + /* Init the device */ + ret = pcim_enable_device(pci); + if (ret) { + dev_err(sst_drv_ctx->dev, + "device can't be enabled. Returned err: %d\n", ret); + goto do_free_drv_ctx; + } + sst_drv_ctx->pci = pci_dev_get(pci); + ret = sst_platform_get_resources(sst_drv_ctx); + if (ret < 0) + goto do_free_drv_ctx; + + pci_set_drvdata(pci, sst_drv_ctx); + sst_configure_runtime_pm(sst_drv_ctx); + + return ret; + +do_free_drv_ctx: + sst_context_cleanup(sst_drv_ctx); + dev_err(sst_drv_ctx->dev, "Probe failed with %d\n", ret); + return ret; +} + +/** + * intel_sst_remove - PCI remove function + * + * @pci: PCI device structure + * + * This function is called by OS when a device is unloaded + * This frees the interrupt etc + */ +static void intel_sst_remove(struct pci_dev *pci) +{ + struct intel_sst_drv *sst_drv_ctx = pci_get_drvdata(pci); + + sst_context_cleanup(sst_drv_ctx); + pci_dev_put(sst_drv_ctx->pci); + pci_release_regions(pci); + pci_set_drvdata(pci, NULL); +} + +/* PCI Routines */ +static struct pci_device_id intel_sst_ids[] = { + { PCI_VDEVICE(INTEL, SST_MRFLD_PCI_ID), 0}, + { 0, } +}; + +static struct pci_driver sst_driver = { + .name = SST_DRV_NAME, + .id_table = intel_sst_ids, + .probe = intel_sst_probe, + .remove = intel_sst_remove, +#ifdef CONFIG_PM + .driver = { + .pm = &intel_sst_pm, + }, +#endif +}; + +module_pci_driver(sst_driver); + +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine PCI Driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_AUTHOR("Dharageswari R "); +MODULE_AUTHOR("KP Jeeja "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 1c2e081..9a5df19 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -433,6 +433,7 @@ u32 relocate_imr_addr_mrfld(u32 base_addr) base_addr = MRFLD_FW_VIRTUAL_BASE + (base_addr % (512 * 1024 * 1024)); return base_addr; } +EXPORT_SYMBOL_GPL(relocate_imr_addr_mrfld); void sst_add_to_dispatch_list_and_post(struct intel_sst_drv *sst, struct ipc_post *msg) -- cgit v1.1 From b0d94acd634a5cff7fe5fc46131a23997e8d0f60 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Nov 2014 16:25:17 +0530 Subject: ASoC: Intel: mrfld - add shim save restore In ACPI platform we need to save few registers of Shim on suspend and restore them on resume, so add handlers to do this Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8753754..b97c231 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -339,6 +339,34 @@ void sst_context_cleanup(struct intel_sst_drv *ctx) } EXPORT_SYMBOL_GPL(sst_context_cleanup); +static inline void sst_save_shim64(struct intel_sst_drv *ctx, + void __iomem *shim, + struct sst_shim_regs64 *shim_regs) +{ + unsigned long irq_flags; + + spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); + + shim_regs->imrx = sst_shim_read64(shim, SST_IMRX), + + spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); +} + +static inline void sst_restore_shim64(struct intel_sst_drv *ctx, + void __iomem *shim, + struct sst_shim_regs64 *shim_regs) +{ + unsigned long irq_flags; + + /* + * we only need to restore IMRX for this case, rest will be + * initialize by FW or driver when firmware is loaded + */ + spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags); + sst_shim_write64(shim, SST_IMRX, shim_regs->imrx), + spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags); +} + void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); -- cgit v1.1 From feec843d6c4528263724ff3f4c463ea82bf63b4a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:22 +0100 Subject: ASoC: ssm4567: Add DAC high-pass-filter control Add a switch which can be used to enable/disable the DAC high-pass-filter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4b5c17f..e1e33d8 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -145,6 +145,8 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, 0xff, 1, ssm4567_vol_tlv), SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), + SOC_SINGLE("DAC High Pass Filter Switch", SSM4567_REG_DAC_CTRL, + 5, 1, 0), }; static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { -- cgit v1.1 From ead99f89b7cd2b5cfe99601380a6f6f0a1ce7e53 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:23 +0100 Subject: ASoC: ssm4567: Add support for setting the DAI format and TDM configuration The SSM4567 has support for a couple of different DAI formats. In TDM mode it is also possible to select the TDM slot. This patch adds support for this by implementing the set_fmt and set_tdm_slot callbacks. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 119 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 119 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index e1e33d8..2176679 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -69,6 +69,22 @@ #define SSM4567_DAC_FS_64000_96000 0x3 #define SSM4567_DAC_FS_128000_192000 0x4 +/* SAI_CTRL_1 */ +#define SSM4567_SAI_CTRL_1_BCLK BIT(6) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK (0x3 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_32 (0x0 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_48 (0x1 << 4) +#define SSM4567_SAI_CTRL_1_TDM_BLCKS_64 (0x2 << 4) +#define SSM4567_SAI_CTRL_1_FSYNC BIT(3) +#define SSM4567_SAI_CTRL_1_LJ BIT(2) +#define SSM4567_SAI_CTRL_1_TDM BIT(1) +#define SSM4567_SAI_CTRL_1_PDM BIT(0) + +/* SAI_CTRL_2 */ +#define SSM4567_SAI_CTRL_2_AUTO_SLOT BIT(3) +#define SSM4567_SAI_CTRL_2_TDM_SLOT_MASK 0x7 +#define SSM4567_SAI_CTRL_2_TDM_SLOT(x) (x) + struct ssm4567 { struct regmap *regmap; }; @@ -194,6 +210,107 @@ static int ssm4567_mute(struct snd_soc_dai *dai, int mute) SSM4567_DAC_MUTE, val); } +static int ssm4567_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai); + unsigned int blcks; + int slot; + int ret; + + if (tx_mask == 0) + return -EINVAL; + + if (rx_mask && rx_mask != tx_mask) + return -EINVAL; + + slot = __ffs(tx_mask); + if (tx_mask != BIT(slot)) + return -EINVAL; + + switch (width) { + case 32: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_32; + break; + case 48: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_48; + break; + case 64: + blcks = SSM4567_SAI_CTRL_1_TDM_BLCKS_64; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_2, + SSM4567_SAI_CTRL_2_AUTO_SLOT | SSM4567_SAI_CTRL_2_TDM_SLOT_MASK, + SSM4567_SAI_CTRL_2_TDM_SLOT(slot)); + if (ret) + return ret; + + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, + SSM4567_SAI_CTRL_1_TDM_BLCKS_MASK, blcks); +} + +static int ssm4567_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct ssm4567 *ssm4567 = snd_soc_dai_get_drvdata(dai); + unsigned int ctrl1 = 0; + bool invert_fclk; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_fclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1 |= SSM4567_SAI_CTRL_1_BCLK; + invert_fclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; + invert_fclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl1 |= SSM4567_SAI_CTRL_1_BCLK; + invert_fclk = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1 |= SSM4567_SAI_CTRL_1_LJ; + invert_fclk = !invert_fclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= SSM4567_SAI_CTRL_1_TDM; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= SSM4567_SAI_CTRL_1_TDM | SSM4567_SAI_CTRL_1_LJ; + break; + case SND_SOC_DAIFMT_PDM: + ctrl1 |= SSM4567_SAI_CTRL_1_PDM; + break; + default: + return -EINVAL; + } + + if (invert_fclk) + ctrl1 |= SSM4567_SAI_CTRL_1_FSYNC; + + return regmap_write(ssm4567->regmap, SSM4567_REG_SAI_CTRL_1, ctrl1); +} + static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) { int ret = 0; @@ -248,6 +365,8 @@ static int ssm4567_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops ssm4567_dai_ops = { .hw_params = ssm4567_hw_params, .digital_mute = ssm4567_mute, + .set_fmt = ssm4567_set_dai_fmt, + .set_tdm_slot = ssm4567_set_tdm_slot, }; static struct snd_soc_dai_driver ssm4567_dai = { -- cgit v1.1 From 5ad72152b695ba5027f9c6ec9a48a8e1a70f25dc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 6 Nov 2014 15:59:24 +0100 Subject: ASoC: ssm4567: Add support for disabling the boost stage This patch adds a switch to enable/disable boost stage of the output amplifier. Applications that know that they do not need the output amplifier boost stage can disable it to conserve a bit of power. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm4567.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 2176679..a984485 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -165,13 +165,20 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { 5, 1, 0), }; +static const struct snd_kcontrol_new ssm4567_amplifier_boost_control = + SOC_DAPM_SINGLE("Switch", SSM4567_REG_POWER_CTRL, 1, 1, 1); + static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1), + SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, + &ssm4567_amplifier_boost_control), SND_SOC_DAPM_OUTPUT("OUT"), }; static const struct snd_soc_dapm_route ssm4567_routes[] = { + { "OUT", NULL, "Amplifier Boost" }, + { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, }; -- cgit v1.1 From 9b105fe447116ee3cd7fe3c09ca6a6d6a05c736b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 6 Nov 2014 19:30:43 +0530 Subject: ASoC: Intel: mrfld - remove non static definition sst_save_shim64() is defined as static in code but header is non static. Since this is not used other than file where defined remove non static definition Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 3ee555e..2dcbf47 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -508,8 +508,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, size_t mbox_data_len, const void *mbox_data, void **data, bool large, bool fill_dsp, bool sync, bool response); -void sst_save_shim64(struct intel_sst_drv *ctx, void __iomem *shim, - struct sst_shim_regs64 *shim_regs); void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); void sst_init_stream(struct stream_info *stream, -- cgit v1.1 From 1c5d1c988302f324ac396ac13461d59d091be605 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Nov 2014 20:26:53 -0800 Subject: ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() rsnd_dvc_volume_update() is main function to control DVC feature like Digital Volume / Mute / Ramp etc. DVC_DVUCR should be controlled under this function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index deaf0fa..3952237 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -40,6 +40,7 @@ struct rsnd_dvc { static void rsnd_dvc_volume_update(struct rsnd_mod *mod) { struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u32 dvucr = 0; u32 mute = 0; int i; @@ -47,10 +48,18 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) mute |= (!!dvc->mute.val[i]) << i; } + /* Enable Digital Volume */ + dvucr = 0x100; rsnd_mod_write(mod, DVC_VOL0R, dvc->volume.val[0]); rsnd_mod_write(mod, DVC_VOL1R, dvc->volume.val[1]); - rsnd_mod_write(mod, DVC_ZCMCR, mute); + /* Enable Mute */ + if (mute) { + dvucr |= 0x1; + rsnd_mod_write(mod, DVC_ZCMCR, mute); + } + + rsnd_mod_write(mod, DVC_DVUCR, dvucr); } static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, @@ -103,9 +112,6 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod, rsnd_mod_write(dvc_mod, DVC_ADINR, rsnd_get_adinr(dvc_mod)); - /* enable Volume / Mute */ - rsnd_mod_write(dvc_mod, DVC_DVUCR, 0x101); - /* ch0/ch1 Volume */ rsnd_dvc_volume_update(dvc_mod); -- cgit v1.1 From 140bab8961eb4047070b46a6dd50ec87496e0cde Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Nov 2014 20:27:18 -0800 Subject: ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() We need to Enable/Disable DVC_DVUER register if we set DVCp_ZCMCR, DVCp_VRCTR, DVCp_VRPDR, DVCp_VRDBR, DVCp_VOL0R, DVCp_VOL1R, DVCp_VOL2R, DVCp_VOL3R, DVCp_VOL4R, DVCp_VOL5R, DVCp_VOL6R, DVCp_VOL7R and, these are controlled under rsnd_dvc_volume_update(). This patch moves DVC_DVUER settings to it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 3952237..ce1512e 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -48,6 +48,9 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) mute |= (!!dvc->mute.val[i]) << i; } + /* Disable DVC Register access */ + rsnd_mod_write(mod, DVC_DVUER, 0); + /* Enable Digital Volume */ dvucr = 0x100; rsnd_mod_write(mod, DVC_VOL0R, dvc->volume.val[0]); @@ -60,6 +63,9 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) } rsnd_mod_write(mod, DVC_DVUCR, dvucr); + + /* Enable DVC Register access */ + rsnd_mod_write(mod, DVC_DVUER, 1); } static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, @@ -117,8 +123,6 @@ static int rsnd_dvc_init(struct rsnd_mod *dvc_mod, rsnd_mod_write(dvc_mod, DVC_DVUIR, 0); - rsnd_mod_write(dvc_mod, DVC_DVUER, 1); - rsnd_adg_set_cmd_timsel_gen2(rdai, dvc_mod, io); return 0; -- cgit v1.1 From ec14af91a03f7d68b2a72bec20be2ab583d3f63a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Nov 2014 20:27:46 -0800 Subject: ASoC: rsnd: enable multiple DVC valume settings DVC controls some digital volume features. Some of them requests values for "each channels", but, some of them requests values for "feature". Current dvc.c is supporting Mute/Volume, and these have "each channels" settings. This patch adds rsnd_dvc_cfg_m and care about multiple settings for each channels. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 44 ++++++++++++++++++++++++++++++-------------- 1 file changed, 30 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index ce1512e..c729e26 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -17,6 +17,12 @@ struct rsnd_dvc_cfg { unsigned int max; + unsigned int size; + u32 *val; +}; + +struct rsnd_dvc_cfg_m { + struct rsnd_dvc_cfg cfg; u32 val[RSND_DVC_CHANNELS]; }; @@ -24,8 +30,8 @@ struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - struct rsnd_dvc_cfg volume; - struct rsnd_dvc_cfg mute; + struct rsnd_dvc_cfg_m volume; + struct rsnd_dvc_cfg_m mute; }; #define rsnd_mod_to_dvc(_mod) \ @@ -44,9 +50,8 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) u32 mute = 0; int i; - for (i = 0; i < RSND_DVC_CHANNELS; i++) { - mute |= (!!dvc->mute.val[i]) << i; - } + for (i = 0; i < dvc->mute.cfg.size; i++) + mute |= (!!dvc->mute.cfg.val[i]) << i; /* Disable DVC Register access */ rsnd_mod_write(mod, DVC_DVUER, 0); @@ -159,7 +164,7 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, { struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; - uinfo->count = RSND_DVC_CHANNELS; + uinfo->count = cfg->size; uinfo->value.integer.min = 0; uinfo->value.integer.max = cfg->max; @@ -177,7 +182,7 @@ static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; int i; - for (i = 0; i < RSND_DVC_CHANNELS; i++) + for (i = 0; i < cfg->size; i++) ucontrol->value.integer.value[i] = cfg->val[i]; return 0; @@ -190,7 +195,7 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; int i, change = 0; - for (i = 0; i < RSND_DVC_CHANNELS; i++) { + for (i = 0; i < cfg->size; i++) { change |= (ucontrol->value.integer.value[i] != cfg->val[i]); cfg->val[i] = ucontrol->value.integer.value[i]; } @@ -230,6 +235,19 @@ static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod, return 0; } +static int _rsnd_dvc_pcm_new_m(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_dvc_cfg_m *private, + u32 max) +{ + private->cfg.max = max; + private->cfg.size = RSND_DVC_CHANNELS; + private->cfg.val = private->val; + return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); +} + static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd) @@ -239,20 +257,18 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int ret; /* Volume */ - dvc->volume.max = 0x00800000 - 1; - ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + ret = _rsnd_dvc_pcm_new_m(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Playback Volume" : "DVC In Capture Volume", - &dvc->volume); + &dvc->volume, 0x00800000 - 1); if (ret < 0) return ret; /* Mute */ - dvc->mute.max = 1; - ret = __rsnd_dvc_pcm_new(mod, rdai, rtd, + ret = _rsnd_dvc_pcm_new_m(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Mute Switch" : "DVC In Mute Switch", - &dvc->mute); + &dvc->mute, 1); if (ret < 0) return ret; -- cgit v1.1 From ab2e479667507329475c8ef93d61f3dbe654c3c2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Nov 2014 20:28:10 -0800 Subject: ASoC: rsnd: enable single DVC valume settings DVC controls some digital volume features. Some of them requests values for "each channels", but, some of them requests values for "feature". And, Volume Ramp has "feature" settings. This patch adds rsnd_dvc_cfg_s and care about single settings. Compiler will report like below at this point, but, it will be removed if Volume Ramp was supported. warning: '_rsnd_dvc_pcm_new_s' defined but not used Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index c729e26..e7cfc71 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -26,6 +26,11 @@ struct rsnd_dvc_cfg_m { u32 val[RSND_DVC_CHANNELS]; }; +struct rsnd_dvc_cfg_s { + struct rsnd_dvc_cfg cfg; + u32 val; +}; + struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; @@ -248,6 +253,19 @@ static int _rsnd_dvc_pcm_new_m(struct rsnd_mod *mod, return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); } +static int _rsnd_dvc_pcm_new_s(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_dvc_cfg_s *private, + u32 max) +{ + private->cfg.max = max; + private->cfg.size = 1; + private->cfg.val = &private->val; + return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); +} + static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd) -- cgit v1.1 From 018342976ce971944dd4d9309f75e86382079a2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 4 Nov 2014 20:28:50 -0800 Subject: ASoC: rsnd: enable enumerated DVC valume settings DVC controls some digital volume features. Volume Ramp is listed as "XX dB / YY steps", and this enumerated settings are easy for users. This patch adds rsnd_dvc_cfg_e and care about enumerated settings. Compiler will report like below at this point, but, it will be removed if Volume Ramp was supported. warning: '_rsnd_dvc_pcm_new_e' defined but not used Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 55 +++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 44 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index e7cfc71..8504f6b 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -19,6 +19,7 @@ struct rsnd_dvc_cfg { unsigned int max; unsigned int size; u32 *val; + const char * const *texts; }; struct rsnd_dvc_cfg_m { @@ -169,14 +170,23 @@ static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, { struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; - uinfo->count = cfg->size; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = cfg->max; - - if (cfg->max == 1) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + if (cfg->texts) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = cfg->size; + uinfo->value.enumerated.items = cfg->max; + if (uinfo->value.enumerated.item >= cfg->max) + uinfo->value.enumerated.item = cfg->max - 1; + strlcpy(uinfo->value.enumerated.name, + cfg->texts[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); + } else { + uinfo->count = cfg->size; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = cfg->max; + uinfo->type = (cfg->max == 1) ? + SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + } return 0; } @@ -188,7 +198,10 @@ static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, int i; for (i = 0; i < cfg->size; i++) - ucontrol->value.integer.value[i] = cfg->val[i]; + if (cfg->texts) + ucontrol->value.enumerated.item[i] = cfg->val[i]; + else + ucontrol->value.integer.value[i] = cfg->val[i]; return 0; } @@ -201,8 +214,13 @@ static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, int i, change = 0; for (i = 0; i < cfg->size; i++) { - change |= (ucontrol->value.integer.value[i] != cfg->val[i]); - cfg->val[i] = ucontrol->value.integer.value[i]; + if (cfg->texts) { + change |= (ucontrol->value.enumerated.item[i] != cfg->val[i]); + cfg->val[i] = ucontrol->value.enumerated.item[i]; + } else { + change |= (ucontrol->value.integer.value[i] != cfg->val[i]); + cfg->val[i] = ucontrol->value.integer.value[i]; + } } if (change) @@ -266,6 +284,21 @@ static int _rsnd_dvc_pcm_new_s(struct rsnd_mod *mod, return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); } +static int _rsnd_dvc_pcm_new_e(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_dvc_cfg_s *private, + const char * const *texts, + u32 max) +{ + private->cfg.max = max; + private->cfg.size = 1; + private->cfg.val = &private->val; + private->cfg.texts = texts; + return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); +} + static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd) -- cgit v1.1 From b07597367001c2c4f36a97863530f71b84060d3d Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 7 Nov 2014 12:24:39 +0530 Subject: ASoC: Samsung: Add quirk for internal DMA Internal DMA is available only on some of Samsung platforms. So added a quirk for the same and made it optional. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s.c | 12 ++++++------ 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index a6591c6..5e0bc77 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -27,6 +27,7 @@ struct samsung_i2s { #define QUIRK_NO_MUXPSR (1 << 2) #define QUIRK_NEED_RSTCLR (1 << 3) #define QUIRK_SUPPORTS_TDM (1 << 4) +#define QUIRK_SUPPORTS_IDMA (1 << 5) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 9d51347..38b9a52 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -987,7 +987,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); - if (i2s->quirks & QUIRK_SEC_DAI) + if (i2s->quirks & QUIRK_SUPPORTS_IDMA) idma_reg_addr_init(i2s->addr, i2s->sec_dai->idma_playback.dma_addr); @@ -1199,10 +1199,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) quirks = i2s_dai_data->quirks; if (of_property_read_u32(np, "samsung,idma-addr", &idma_addr)) { - if (quirks & QUIRK_SEC_DAI) { - dev_err(&pdev->dev, "idma address is not"\ + if (quirks & QUIRK_SUPPORTS_IDMA) { + dev_info(&pdev->dev, "idma address is not"\ "specified"); - return -EINVAL; } } } @@ -1309,13 +1308,14 @@ static const struct samsung_i2s_dai_data i2sv3_dai_type = { static const struct samsung_i2s_dai_data i2sv5_dai_type = { .dai_type = TYPE_PRI, - .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | + QUIRK_SUPPORTS_IDMA, }; static const struct samsung_i2s_dai_data i2sv6_dai_type = { .dai_type = TYPE_PRI, .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | - QUIRK_SUPPORTS_TDM, + QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA, }; static const struct samsung_i2s_dai_data samsung_dai_type_pri = { -- cgit v1.1 From a5a56871f804edac93a53b5e871c0e9818fb9033 Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Fri, 7 Nov 2014 12:24:40 +0530 Subject: ASoC: samsung: add support for exynos7 I2S controller Exynos7 I2S controller has no internal dma, supports more no. of root clock sampling frequencies and has more no.of Rx fifos to support 7.1CH recording in TDM mode. Due to more no. of root clock frequency values some of the bit offsets got shifted up by one. Also I2S1 on previous Samsung platforms uses v3 dai type but on Exynos7 it is upgraded to v5 with slightly modified register offsets for supporting more no.of RFS values. Due to the above changes, the driver has to be modified to handle all versions of I2S controller. For this I introduced a new structure to hold modified bit offsets and masks which is passed as dai data. Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 218 +++++++++++++++------ 4 files changed, 174 insertions(+), 71 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index 7386d44..d188296 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -6,10 +6,17 @@ Required SoC Specific Properties: - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with - secondary fifo, s/w reset control, internal mux for root clk src and - TDM support. TDM (Time division multiplexing) is to allow transfer of - multiple channel audio data on single data line. + - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, sterio channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + - samsung,exynos7-i2s: with all the available features of exynos5 i2s, + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. - reg: physical base address of the controller and length of memory mapped region. diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 55a3869..e0e737f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG + depends on (PLAT_SAMSUNG || ARCH_EXYNOS) depends on S3C64XX_PL080 || !ARCH_S3C64XX depends on S3C24XX_DMAC || !ARCH_S3C24XX select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h index 821a502..9170c31 100644 --- a/sound/soc/samsung/i2s-regs.h +++ b/sound/soc/samsung/i2s-regs.h @@ -33,8 +33,9 @@ #define I2SLVL3ADDR 0x3c #define I2SSTR1 0x40 #define I2SVER 0x44 -#define I2SFIC2 0x48 +#define I2SFIC1 0x48 #define I2STDM 0x4c +#define I2SFSTA 0x50 #define CON_RSTCLR (1 << 31) #define CON_FRXOFSTATUS (1 << 26) @@ -93,8 +94,6 @@ #define MOD_BLC_24BIT (2 << 13) #define MOD_BLC_MASK (3 << 13) -#define MOD_IMS_SYSMUX (1 << 10) -#define MOD_SLAVE (1 << 11) #define MOD_TXONLY (0 << 8) #define MOD_RXONLY (1 << 8) #define MOD_TXRX (2 << 8) @@ -132,7 +131,10 @@ #define EXYNOS5420_MOD_BCLK_256FS 8 #define EXYNOS5420_MOD_BCLK_MASK 0xf -#define MOD_CDCLKCON (1 << 12) +#define EXYNOS7_MOD_RCLK_64FS 4 +#define EXYNOS7_MOD_RCLK_128FS 5 +#define EXYNOS7_MOD_RCLK_96FS 6 +#define EXYNOS7_MOD_RCLK_192FS 7 #define PSR_PSREN (1 << 15) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 38b9a52..947352d 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -36,9 +36,24 @@ enum samsung_dai_type { TYPE_SEC, }; +struct samsung_i2s_variant_regs { + unsigned int bfs_off; + unsigned int rfs_off; + unsigned int sdf_off; + unsigned int txr_off; + unsigned int rclksrc_off; + unsigned int mss_off; + unsigned int cdclkcon_off; + unsigned int lrp_off; + unsigned int bfs_mask; + unsigned int rfs_mask; + unsigned int ftx0cnt_off; +}; + struct samsung_i2s_dai_data { int dai_type; u32 quirks; + const struct samsung_i2s_variant_regs *i2s_variant_regs; }; struct i2s_dai { @@ -81,6 +96,7 @@ struct i2s_dai { u32 suspend_i2scon; u32 suspend_i2spsr; unsigned long gpios[7]; /* i2s gpio line numbers */ + const struct samsung_i2s_variant_regs *variant_regs; }; /* Lock for cross i/f checks */ @@ -95,7 +111,8 @@ static inline bool is_secondary(struct i2s_dai *i2s) /* If operating in SoC-Slave mode */ static inline bool is_slave(struct i2s_dai *i2s) { - return (readl(i2s->addr + I2SMOD) & MOD_SLAVE) ? true : false; + u32 mod = readl(i2s->addr + I2SMOD); + return (mod & (1 << i2s->variant_regs->mss_off)) ? true : false; } /* If this interface of the controller is transmitting data */ @@ -200,14 +217,14 @@ static inline bool is_manager(struct i2s_dai *i2s) static inline unsigned get_rfs(struct i2s_dai *i2s) { u32 rfs; - - if (i2s->quirks & QUIRK_SUPPORTS_TDM) - rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT; - else - rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT); - rfs &= MOD_RCLK_MASK; + rfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->rfs_off; + rfs &= i2s->variant_regs->rfs_mask; switch (rfs) { + case 7: return 192; + case 6: return 96; + case 5: return 128; + case 4: return 64; case 3: return 768; case 2: return 384; case 1: return 512; @@ -219,15 +236,23 @@ static inline unsigned get_rfs(struct i2s_dai *i2s) static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) { u32 mod = readl(i2s->addr + I2SMOD); - int rfs_shift; + int rfs_shift = i2s->variant_regs->rfs_off; - if (i2s->quirks & QUIRK_SUPPORTS_TDM) - rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT; - else - rfs_shift = MOD_RCLK_SHIFT; - mod &= ~(MOD_RCLK_MASK << rfs_shift); + mod &= ~(i2s->variant_regs->rfs_mask << rfs_shift); switch (rfs) { + case 192: + mod |= (EXYNOS7_MOD_RCLK_192FS << rfs_shift); + break; + case 96: + mod |= (EXYNOS7_MOD_RCLK_96FS << rfs_shift); + break; + case 128: + mod |= (EXYNOS7_MOD_RCLK_128FS << rfs_shift); + break; + case 64: + mod |= (EXYNOS7_MOD_RCLK_64FS << rfs_shift); + break; case 768: mod |= (MOD_RCLK_768FS << rfs_shift); break; @@ -249,14 +274,8 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs) static inline unsigned get_bfs(struct i2s_dai *i2s) { u32 bfs; - - if (i2s->quirks & QUIRK_SUPPORTS_TDM) { - bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT; - bfs &= EXYNOS5420_MOD_BCLK_MASK; - } else { - bfs = readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT; - bfs &= MOD_BCLK_MASK; - } + bfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->bfs_off; + bfs &= i2s->variant_regs->bfs_mask; switch (bfs) { case 8: return 256; @@ -275,16 +294,8 @@ static inline unsigned get_bfs(struct i2s_dai *i2s) static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) { u32 mod = readl(i2s->addr + I2SMOD); - int bfs_shift; int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM; - - if (i2s->quirks & QUIRK_SUPPORTS_TDM) { - bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT; - mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift); - } else { - bfs_shift = MOD_BCLK_SHIFT; - mod &= ~(MOD_BCLK_MASK << bfs_shift); - } + int bfs_shift = i2s->variant_regs->bfs_off; /* Non-TDM I2S controllers do not support BCLK > 48 * FS */ if (!tdm && bfs > 48) { @@ -292,6 +303,8 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs) return; } + mod &= ~(i2s->variant_regs->bfs_mask << bfs_shift); + switch (bfs) { case 48: mod |= (MOD_BCLK_48FS << bfs_shift); @@ -346,8 +359,9 @@ static inline int get_blc(struct i2s_dai *i2s) static void i2s_txctrl(struct i2s_dai *i2s, int on) { void __iomem *addr = i2s->addr; + int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); - u32 mod = readl(addr + I2SMOD) & ~MOD_MASK; + u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); if (on) { con |= CON_ACTIVE; @@ -362,9 +376,9 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) } if (any_rx_active(i2s)) - mod |= MOD_TXRX; + mod |= 2 << txr_off; else - mod |= MOD_TXONLY; + mod |= 0 << txr_off; } else { if (is_secondary(i2s)) { con |= CON_TXSDMA_PAUSE; @@ -382,7 +396,7 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) con |= CON_TXCH_PAUSE; if (any_rx_active(i2s)) - mod |= MOD_RXONLY; + mod |= 1 << txr_off; else con &= ~CON_ACTIVE; } @@ -395,23 +409,24 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on) static void i2s_rxctrl(struct i2s_dai *i2s, int on) { void __iomem *addr = i2s->addr; + int txr_off = i2s->variant_regs->txr_off; u32 con = readl(addr + I2SCON); - u32 mod = readl(addr + I2SMOD) & ~MOD_MASK; + u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off); if (on) { con |= CON_RXDMA_ACTIVE | CON_ACTIVE; con &= ~(CON_RXDMA_PAUSE | CON_RXCH_PAUSE); if (any_tx_active(i2s)) - mod |= MOD_TXRX; + mod |= 2 << txr_off; else - mod |= MOD_RXONLY; + mod |= 1 << txr_off; } else { con |= CON_RXDMA_PAUSE | CON_RXCH_PAUSE; con &= ~CON_RXDMA_ACTIVE; if (any_tx_active(i2s)) - mod |= MOD_TXONLY; + mod |= 0 << txr_off; else con &= ~CON_ACTIVE; } @@ -451,6 +466,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; u32 mod = readl(i2s->addr + I2SMOD); + const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; + unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off; + unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off; switch (clk_id) { case SAMSUNG_I2S_OPCLK: @@ -465,18 +483,18 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) - && !(mod & MOD_CDCLKCON)) || + && !(mod & cdcon_mask)) || ((dir == SND_SOC_CLOCK_OUT) - && (mod & MOD_CDCLKCON))))) { + && (mod & cdcon_mask))))) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } if (dir == SND_SOC_CLOCK_IN) - mod |= MOD_CDCLKCON; + mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= ~MOD_CDCLKCON; + mod &= 0 << i2s_regs->cdclkcon_off; i2s->rfs = rfs; break; @@ -491,8 +509,8 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (!any_active(i2s)) { if (i2s->op_clk && !IS_ERR(i2s->op_clk)) { - if ((clk_id && !(mod & MOD_IMS_SYSMUX)) || - (!clk_id && (mod & MOD_IMS_SYSMUX))) { + if ((clk_id && !(mod & rsrc_mask)) || + (!clk_id && (mod & rsrc_mask))) { clk_disable_unprepare(i2s->op_clk); clk_put(i2s->op_clk); } else { @@ -520,8 +538,8 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, other->op_clk = i2s->op_clk; other->rclk_srcrate = i2s->rclk_srcrate; } - } else if ((!clk_id && (mod & MOD_IMS_SYSMUX)) - || (clk_id && !(mod & MOD_IMS_SYSMUX))) { + } else if ((!clk_id && (mod & rsrc_mask)) + || (clk_id && !(mod & rsrc_mask))) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; @@ -533,10 +551,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= ~MOD_IMS_SYSMUX; + mod &= 0 << i2s_regs->rclksrc_off; else - mod |= MOD_IMS_SYSMUX; - break; + mod |= 1 << i2s_regs->rclksrc_off; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); @@ -553,16 +570,12 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, { struct i2s_dai *i2s = to_info(dai); u32 mod = readl(i2s->addr + I2SMOD); - int lrp_shift, sdf_shift, sdf_mask, lrp_rlow; + int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave; u32 tmp = 0; - if (i2s->quirks & QUIRK_SUPPORTS_TDM) { - lrp_shift = EXYNOS5420_MOD_LRP_SHIFT; - sdf_shift = EXYNOS5420_MOD_SDF_SHIFT; - } else { - lrp_shift = MOD_LRP_SHIFT; - sdf_shift = MOD_SDF_SHIFT; - } + lrp_shift = i2s->variant_regs->lrp_off; + sdf_shift = i2s->variant_regs->sdf_off; + mod_slave = 1 << i2s->variant_regs->mss_off; sdf_mask = MOD_SDF_MASK << sdf_shift; lrp_rlow = MOD_LR_RLOW << lrp_shift; @@ -605,7 +618,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - tmp |= MOD_SLAVE; + tmp |= mod_slave; break; case SND_SOC_DAIFMT_CBS_CFS: /* Set default source clock in Master mode */ @@ -623,13 +636,13 @@ static int i2s_set_fmt(struct snd_soc_dai *dai, * channel. */ if (any_active(i2s) && - ((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) { + ((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) { dev_err(&i2s->pdev->dev, "%s:%d Other DAI busy\n", __func__, __LINE__); return -EAGAIN; } - mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE); + mod &= ~(sdf_mask | lrp_rlow | mod_slave); mod |= tmp; writel(mod, i2s->addr + I2SMOD); @@ -751,6 +764,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; unsigned long flags; + const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; spin_lock_irqsave(&lock, flags); @@ -761,7 +775,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, other->mode |= DAI_MANAGER; } else { u32 mod = readl(i2s->addr + I2SMOD); - i2s->cdclk_out = !(mod & MOD_CDCLKCON); + i2s->cdclk_out = !(mod & (1 << i2s_regs->cdclkcon_off)); if (other) other->cdclk_out = i2s->cdclk_out; } @@ -914,13 +928,14 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) struct i2s_dai *i2s = to_info(dai); u32 reg = readl(i2s->addr + I2SFIC); snd_pcm_sframes_t delay; + const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) delay = FIC_RXCOUNT(reg); else if (is_secondary(i2s)) delay = FICS_TXCOUNT(readl(i2s->addr + I2SFICS)); else - delay = FIC_TXCOUNT(reg); + delay = (reg >> i2s_regs->ftx0cnt_off) & 0x7f; return delay; } @@ -1227,6 +1242,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.dma_size = 4; pri_dai->base = regs_base; pri_dai->quirks = quirks; + pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; if (quirks & QUIRK_PRI_6CHAN) pri_dai->i2s_dai_drv.playback.channels_max = 6; @@ -1301,21 +1317,93 @@ static int samsung_i2s_remove(struct platform_device *pdev) return 0; } +static const struct samsung_i2s_variant_regs i2sv3_regs = { + .bfs_off = 1, + .rfs_off = 3, + .sdf_off = 5, + .txr_off = 8, + .rclksrc_off = 10, + .mss_off = 11, + .cdclkcon_off = 12, + .lrp_off = 7, + .bfs_mask = 0x3, + .rfs_mask = 0x3, + .ftx0cnt_off = 8, +}; + +static const struct samsung_i2s_variant_regs i2sv6_regs = { + .bfs_off = 0, + .rfs_off = 4, + .sdf_off = 6, + .txr_off = 8, + .rclksrc_off = 10, + .mss_off = 11, + .cdclkcon_off = 12, + .lrp_off = 15, + .bfs_mask = 0xf, + .rfs_mask = 0x3, + .ftx0cnt_off = 8, +}; + +static const struct samsung_i2s_variant_regs i2sv7_regs = { + .bfs_off = 0, + .rfs_off = 4, + .sdf_off = 7, + .txr_off = 9, + .rclksrc_off = 11, + .mss_off = 12, + .cdclkcon_off = 22, + .lrp_off = 15, + .bfs_mask = 0xf, + .rfs_mask = 0x7, + .ftx0cnt_off = 0, +}; + +static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = { + .bfs_off = 0, + .rfs_off = 3, + .sdf_off = 6, + .txr_off = 8, + .rclksrc_off = 10, + .mss_off = 11, + .cdclkcon_off = 12, + .lrp_off = 15, + .bfs_mask = 0x7, + .rfs_mask = 0x7, + .ftx0cnt_off = 8, +}; + static const struct samsung_i2s_dai_data i2sv3_dai_type = { .dai_type = TYPE_PRI, .quirks = QUIRK_NO_MUXPSR, + .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv5_dai_type = { .dai_type = TYPE_PRI, .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_IDMA, + .i2s_variant_regs = &i2sv3_regs, }; static const struct samsung_i2s_dai_data i2sv6_dai_type = { .dai_type = TYPE_PRI, .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA, + .i2s_variant_regs = &i2sv6_regs, +}; + +static const struct samsung_i2s_dai_data i2sv7_dai_type = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR | + QUIRK_SUPPORTS_TDM, + .i2s_variant_regs = &i2sv7_regs, +}; + +static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = { + .dai_type = TYPE_PRI, + .quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR, + .i2s_variant_regs = &i2sv5_i2s1_regs, }; static const struct samsung_i2s_dai_data samsung_dai_type_pri = { @@ -1349,6 +1437,12 @@ static const struct of_device_id exynos_i2s_match[] = { }, { .compatible = "samsung,exynos5420-i2s", .data = &i2sv6_dai_type, + }, { + .compatible = "samsung,exynos7-i2s", + .data = &i2sv7_dai_type, + }, { + .compatible = "samsung,exynos7-i2s1", + .data = &i2sv5_dai_type_i2s1, }, {}, }; -- cgit v1.1 From 19ba484d7b15c8650b30377aad6e65b34d3cf3d5 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 5 Nov 2014 13:42:53 +0800 Subject: ASoC: rt5677: Use specific r/w function for DSP mode In DSP mode, the register r/w should use the specific function to access that is invoked by address mapping of the DSP. The MX-65[1] is for switching DSP or codec mode. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 167 +++++++++++++++++++++++++++------------------- sound/soc/codecs/rt5677.h | 3 +- 2 files changed, 102 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 0d24dc4..4b6f7d5 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -541,49 +541,51 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) /** * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. - * @codec: SoC audio codec device. + * @rt5677: Private Data. * @addr: Address index. * @value: Address data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write_addr(struct rt5677_priv *rt5677, unsigned int addr, unsigned int value, unsigned int opcode) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_codec *codec = rt5677->codec; int ret; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_MSB, + addr >> 16); if (ret < 0) { dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_MSB, value >> 16); if (ret < 0) { dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_LSB, value & 0xffff); if (ret < 0) { dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_OP_CODE, + opcode); if (ret < 0) { dev_err(codec->dev, "Failed to set op code value: %d\n", ret); goto err; @@ -597,42 +599,45 @@ err: /** * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @addr: Address index. * @value: Address data. * + * * Returns 0 for success or negative error code. */ static int rt5677_dsp_mode_i2c_read_addr( - struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) + struct rt5677_priv *rt5677, unsigned int addr, unsigned int *value) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_codec *codec = rt5677->codec; int ret; unsigned int msb, lsb; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_MSB, + addr >> 16); if (ret < 0) { dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + ret = regmap_write(rt5677->regmap_physical, RT5677_DSP_I2C_OP_CODE, + 0x0002); if (ret < 0) { dev_err(codec->dev, "Failed to set op code value: %d\n", ret); goto err; } - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + regmap_read(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_MSB, &msb); + regmap_read(rt5677->regmap_physical, RT5677_DSP_I2C_DATA_LSB, &lsb); *value = (msb << 16) | lsb; err: @@ -643,17 +648,17 @@ err: /** * rt5677_dsp_mode_i2c_write - Write register on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @reg: Register index. * @value: Register data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write(struct rt5677_priv *rt5677, unsigned int reg, unsigned int value) { - return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + return rt5677_dsp_mode_i2c_write_addr(rt5677, 0x18020000 + reg * 2, value, 0x0001); } @@ -661,57 +666,33 @@ static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, * rt5677_dsp_mode_i2c_read - Read register on DSP mode. * @codec: SoC audio codec device. * @reg: Register index. + * @value: Register data. * * - * Returns Register value. + * Returns 0 for success or negative error code. */ -static unsigned int rt5677_dsp_mode_i2c_read( - struct snd_soc_codec *codec, unsigned int reg) +static int rt5677_dsp_mode_i2c_read( + struct rt5677_priv *rt5677, unsigned int reg, unsigned int *value) { - unsigned int value = 0; + int ret = rt5677_dsp_mode_i2c_read_addr(rt5677, 0x18020000 + reg * 2, + value); - rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + *value &= 0xffff; - return value; + return ret; } -/** - * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. - * @codec: audio codec - * @reg: register index. - * @mask: register mask - * @value: new value - * - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, unsigned int value) +static void rt5677_set_dsp_mode(struct snd_soc_codec *codec, bool on) { - unsigned int old, new; - int change, ret; - - ret = rt5677_dsp_mode_i2c_read(codec, reg); - if (ret < 0) { - dev_err(codec->dev, "Failed to read reg: %d\n", ret); - goto err; - } + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = rt5677_dsp_mode_i2c_write(codec, reg, new); - if (ret < 0) { - dev_err(codec->dev, - "Failed to write reg: %d\n", ret); - goto err; - } + if (on) { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2); + rt5677->is_dsp_mode = true; + } else { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0); + rt5677->is_dsp_mode = false; } - return change; - -err: - return ret; } static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) @@ -733,9 +714,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); + rt5677_set_dsp_mode(codec, true); ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); @@ -751,8 +737,7 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) release_firmware(rt5677->fw2); } - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0); regcache_cache_bypass(rt5677->regmap, false); regcache_cache_only(rt5677->regmap, true); @@ -762,9 +747,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) regcache_cache_only(rt5677->regmap, false); regcache_cache_bypass(rt5677->regmap, true); - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x1); - rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1); + rt5677_set_dsp_mode(codec, false); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); @@ -4019,6 +4004,32 @@ static int rt5677_resume(struct snd_soc_codec *codec) #define rt5677_resume NULL #endif +static int rt5677_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + else + regmap_read(rt5677->regmap_physical, reg, val); + + return 0; +} + +static int rt5677_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + else + regmap_write(rt5677->regmap_physical, reg, val); + + return 0; +} + #define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) @@ -4144,6 +4155,17 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5677 = { .num_dapm_routes = ARRAY_SIZE(rt5677_dapm_routes), }; +static const struct regmap_config rt5677_regmap_physical = { + .name = "physical", + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5677_VENDOR_ID2 + 1, + .readable_reg = rt5677_readable_register, + + .cache_type = REGCACHE_NONE, +}; + static const struct regmap_config rt5677_regmap = { .reg_bits = 8, .val_bits = 16, @@ -4153,6 +4175,8 @@ static const struct regmap_config rt5677_regmap = { .volatile_reg = rt5677_volatile_register, .readable_reg = rt5677_readable_register, + .reg_read = rt5677_read, + .reg_write = rt5677_write, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -4309,7 +4333,16 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, msleep(10); } - rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); + rt5677->regmap_physical = devm_regmap_init_i2c(i2c, + &rt5677_regmap_physical); + if (IS_ERR(rt5677->regmap_physical)) { + ret = PTR_ERR(rt5677->regmap_physical); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt5677->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 2f5b8c6..9d473b2 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1628,7 +1628,7 @@ enum { struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; - struct regmap *regmap; + struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; struct mutex dsp_cmd_lock; @@ -1646,6 +1646,7 @@ struct rt5677_priv { #endif bool dsp_vad_en; struct regmap_irq_chip_data *irq_data; + bool is_dsp_mode; }; #endif /* __RT5677_H__ */ -- cgit v1.1 From 9e2683530d6f78b30bcf4cabb97d1b7d6b925b85 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 31 Oct 2014 15:37:55 +0800 Subject: ASoC: rt5645: Add ASRC support This patch add ASRC support for rt5645 codec. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 144 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 144 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 286438d..1dbbebc 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -441,6 +441,65 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -552,6 +611,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5645_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5645_ASRC_3; + shift = 4; + break; + case 3: + reg = RT5645_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5645_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5645_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5645_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1244,6 +1350,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1, + 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1, + 0, 0, NULL, 0), + /* Input Side */ /* micbias */ SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, @@ -1502,6 +1632,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + { "IN1P", NULL, "LDO2" }, { "IN2P", NULL, "LDO2" }, @@ -1548,12 +1689,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" }, { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" }, { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" }, { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, -- cgit v1.1 From 31584ed18c073176a7ad96ddbfd09765e21e813d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 7 Nov 2014 14:12:34 +0100 Subject: ALSA: snd_ctl_activate_id(): Fix index look-up We want to know the offset for the id that was passed to the function, not the offset of the first id of the control (which is always 0). Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 99aa3aa..bb96a46 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -572,7 +572,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, ret = -ENOENT; goto unlock; } - index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + index_offset = snd_ctl_get_ioff(kctl, id); vd = &kctl->vd[index_offset]; ret = 0; if (active) { -- cgit v1.1 From cf1f2ebe8d6176de80ef9d9c979f998ec38fb265 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 3 Nov 2014 19:33:03 +0100 Subject: ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks The wm9712/wm9713 has separate mixers for the left and the right channel, but the inputs to the mixers are enabled/disabled by the same control. Currently this is implemented by the driver by registering two virtual controls for each physical control, one for the left mixer and one for the right mixer. Using virtual registers will no longer work when the driver has been converted to regmap. This patch converts the driver to use controls with custom put/get callbacks instead which implement the logic making sure that the physical control is unmuted when either the left or the right control is unmuted. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 163 +++++++++++++++++++++++++++++----------------- sound/soc/codecs/wm9713.c | 153 +++++++++++++++++++++++++------------------ 2 files changed, 194 insertions(+), 122 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f3aab6e..3fad37e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -23,6 +23,11 @@ #include #include "wm9712.h" +struct wm9712_priv { + unsigned int hp_mixer[2]; + struct mutex lock; +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, @@ -48,12 +53,10 @@ static const u16 wm9712_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ - 0x0000, 0x0000 /* virtual hp mixers */ }; -/* virtual HP mixers regs */ -#define HPL_MIXER 0x80 -#define HPR_MIXER 0x82 +#define HPL_MIXER 0x0 +#define HPR_MIXER 0x1 static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"}; static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"}; @@ -157,75 +160,108 @@ SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv), }; +static const unsigned int wm9712_mixer_mute_regs[] = { + AC97_VIDEO, + AC97_PCM, + AC97_LINE, + AC97_PHONE, + AC97_CD, + AC97_PC_BEEP, +}; + /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. */ -static int mixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) +static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - u16 l, r, beep, line, phone, mic, pcm, aux; - - l = ac97_read(w->codec, HPL_MIXER); - r = ac97_read(w->codec, HPR_MIXER); - beep = ac97_read(w->codec, AC97_PC_BEEP); - mic = ac97_read(w->codec, AC97_VIDEO); - phone = ac97_read(w->codec, AC97_PHONE); - line = ac97_read(w->codec, AC97_LINE); - pcm = ac97_read(w->codec, AC97_PCM); - aux = ac97_read(w->codec, AC97_CD); - - if (l & 0x1 || r & 0x1) - ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + unsigned int val = ucontrol->value.enumerated.item[0]; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mixer, mask, shift, old; + struct snd_soc_dapm_update update; + bool change; + + mixer = mc->shift >> 8; + shift = mc->shift & 0xff; + mask = 1 << shift; + + mutex_lock(&wm9712->lock); + old = wm9712->hp_mixer[mixer]; + if (ucontrol->value.enumerated.item[0]) + wm9712->hp_mixer[mixer] |= mask; else - ac97_write(w->codec, AC97_VIDEO, mic | 0x8000); + wm9712->hp_mixer[mixer] &= ~mask; + + change = old != wm9712->hp_mixer[mixer]; + if (change) { + update.kcontrol = kcontrol; + update.reg = wm9712_mixer_mute_regs[shift]; + update.mask = 0x8000; + if ((wm9712->hp_mixer[0] & mask) || + (wm9712->hp_mixer[1] & mask)) + update.val = 0x0; + else + update.val = 0x8000; + + snd_soc_dapm_mixer_update_power(dapm, kcontrol, val, + &update); + } - if (l & 0x2 || r & 0x2) - ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); - else - ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + mutex_unlock(&wm9712->lock); - if (l & 0x4 || r & 0x4) - ac97_write(w->codec, AC97_LINE, line & 0x7fff); - else - ac97_write(w->codec, AC97_LINE, line | 0x8000); + return change; +} - if (l & 0x8 || r & 0x8) - ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); - else - ac97_write(w->codec, AC97_PHONE, phone | 0x8000); +static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int shift, mixer; - if (l & 0x10 || r & 0x10) - ac97_write(w->codec, AC97_CD, aux & 0x7fff); - else - ac97_write(w->codec, AC97_CD, aux | 0x8000); + mixer = mc->shift >> 8; + shift = mc->shift & 0xff; - if (l & 0x20 || r & 0x20) - ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); - else - ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + ucontrol->value.enumerated.item[0] = + (wm9712->hp_mixer[mixer] >> shift) & 1; return 0; } +#define WM9712_HP_MIXER_CTRL(xname, xmixer, xshift) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = wm9712_hp_mixer_get, .put = wm9712_hp_mixer_put, \ + .private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, \ + (xmixer << 8) | xshift, 1, 0, 0) \ +} + /* Left Headphone Mixers */ static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = { - SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0), - SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0), - SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0), - SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0), - SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0), - SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0), + WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPL_MIXER, 5), + WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 4), + WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPL_MIXER, 3), + WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPL_MIXER, 2), + WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 1), + WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPL_MIXER, 0), }; /* Right Headphone Mixers */ static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = { - SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0), - SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0), - SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0), - SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0), - SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0), - SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0), + WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPR_MIXER, 5), + WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 4), + WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPR_MIXER, 3), + WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPR_MIXER, 2), + WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 1), + WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPR_MIXER, 0), }; /* Speaker Mixer */ @@ -299,12 +335,10 @@ SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1, - &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls), - mixer_event, SND_SOC_DAPM_POST_REG), -SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1, - &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls), - mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_INT_PAGING, 9, 1, + &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_INT_PAGING, 8, 1, + &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls)), SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1, &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)), SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1, @@ -471,8 +505,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - if (reg < 0x7c) - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -684,6 +717,16 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { static int wm9712_probe(struct platform_device *pdev) { + struct wm9712_priv *wm9712; + + wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL); + if (wm9712 == NULL) + return -ENOMEM; + + mutex_init(&wm9712->lock); + + platform_set_drvdata(pdev, wm9712); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai)); } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ac13fc8..998e4c7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -31,6 +31,8 @@ struct wm9713_priv { u32 pll_in; /* PLL input frequency */ + unsigned int hp_mixer[2]; + struct mutex lock; }; static unsigned int ac97_read(struct snd_soc_codec *codec, @@ -59,12 +61,10 @@ static const u16 wm9713_reg[] = { 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0006, 0x0001, 0x0000, 0x574d, 0x4c13, - 0x0000, 0x0000 }; -/* virtual HP mixers regs */ -#define HPL_MIXER 0x80 -#define HPR_MIXER 0x82 +#define HPL_MIXER 0 +#define HPR_MIXER 1 static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; @@ -233,6 +233,14 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, return 0; } +static const unsigned int wm9713_mixer_mute_regs[] = { + AC97_PC_BEEP, + AC97_MASTER_TONE, + AC97_PHONE, + AC97_REC_SEL, + AC97_PCM, + AC97_AUX, +}; /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. @@ -240,73 +248,95 @@ static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, * register map, thus we add a new (virtual) register to help determine the * audio route within the device. */ -static int mixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int wm9713_hp_mixer_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - u16 l, r, beep, tone, phone, rec, pcm, aux; - - l = ac97_read(w->codec, HPL_MIXER); - r = ac97_read(w->codec, HPR_MIXER); - beep = ac97_read(w->codec, AC97_PC_BEEP); - tone = ac97_read(w->codec, AC97_MASTER_TONE); - phone = ac97_read(w->codec, AC97_PHONE); - rec = ac97_read(w->codec, AC97_REC_SEL); - pcm = ac97_read(w->codec, AC97_PCM); - aux = ac97_read(w->codec, AC97_AUX); - - if (event & SND_SOC_DAPM_PRE_REG) - return 0; - if ((l & 0x1) || (r & 0x1)) - ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + unsigned int val = ucontrol->value.enumerated.item[0]; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mixer, mask, shift, old; + struct snd_soc_dapm_update update; + bool change; + + mixer = mc->shift >> 8; + shift = mc->shift & 0xff; + mask = (1 << shift); + + mutex_lock(&wm9713->lock); + old = wm9713->hp_mixer[mixer]; + if (ucontrol->value.enumerated.item[0]) + wm9713->hp_mixer[mixer] |= mask; else - ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + wm9713->hp_mixer[mixer] &= ~mask; + + change = old != wm9713->hp_mixer[mixer]; + if (change) { + update.kcontrol = kcontrol; + update.reg = wm9713_mixer_mute_regs[shift]; + update.mask = 0x8000; + if ((wm9713->hp_mixer[0] & mask) || + (wm9713->hp_mixer[1] & mask)) + update.val = 0x0; + else + update.val = 0x8000; + + snd_soc_dapm_mixer_update_power(dapm, kcontrol, val, + &update); + } - if ((l & 0x2) || (r & 0x2)) - ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); - else - ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); + mutex_unlock(&wm9713->lock); - if ((l & 0x4) || (r & 0x4)) - ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); - else - ac97_write(w->codec, AC97_PHONE, phone | 0x8000); + return change; +} - if ((l & 0x8) || (r & 0x8)) - ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); - else - ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); +static int wm9713_hp_mixer_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mixer, shift; - if ((l & 0x10) || (r & 0x10)) - ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); - else - ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + mixer = mc->shift >> 8; + shift = mc->shift & 0xff; - if ((l & 0x20) || (r & 0x20)) - ac97_write(w->codec, AC97_AUX, aux & 0x7fff); - else - ac97_write(w->codec, AC97_AUX, aux | 0x8000); + ucontrol->value.enumerated.item[0] = + (wm9713->hp_mixer[mixer] >> shift) & 1; return 0; } +#define WM9713_HP_MIXER_CTRL(xname, xmixer, xshift) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = wm9713_hp_mixer_get, .put = wm9713_hp_mixer_put, \ + .private_value = SOC_DOUBLE_VALUE(SND_SOC_NOPM, \ + xshift, xmixer, 1, 0, 0) \ +} + /* Left Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { -SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0), -SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), -SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), -SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), -SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), +WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPL_MIXER, 5), +WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPL_MIXER, 4), +WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 3), +WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 2), +WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPL_MIXER, 1), +WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPL_MIXER, 0), }; /* Right Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { -SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0), -SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), -SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), -SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), -SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), +WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPR_MIXER, 5), +WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPR_MIXER, 4), +WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 3), +WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 2), +WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPR_MIXER, 1), +WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPR_MIXER, 0), }; /* headphone capture mux */ @@ -428,12 +458,10 @@ SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, &wm9713_mic_sel_mux_controls), SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, &wm9713_micb_sel_mux_controls), -SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, - &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), - mixer_event, SND_SOC_DAPM_POST_REG), -SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, - &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), - mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, + &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, + &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, @@ -666,8 +694,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { u16 *cache = codec->reg_cache; - if (reg < 0x7c) - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1251,6 +1278,8 @@ static int wm9713_probe(struct platform_device *pdev) if (wm9713 == NULL) return -ENOMEM; + mutex_init(&wm9713->lock); + platform_set_drvdata(pdev, wm9713); return snd_soc_register_codec(&pdev->dev, -- cgit v1.1 From 6cc79294efefde2593eaf72effebc8b1cc71d5ac Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 8 Nov 2014 16:38:06 +0100 Subject: ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync() For convenience for drivers that do not want to keep their own pointer to regmap struct around forward calls to snd_soc_cache_sync() to regcache_sync() if the driver is using regmap. This is similar to what we do for snd_soc_read()/snd_soc_write(). This patch also fixes drivers which already have been converted to regmap, but still use snd_soc_cache_sync() for trying to the sync the cache. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a9f82b5..6dab817 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -187,6 +187,9 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) const char *name = "flat"; int ret; + if (codec->component.regmap) + return regcache_sync(codec->component.regmap); + if (!codec->cache_sync) return 0; -- cgit v1.1 From 427d204c86e095bb91eb8af381bd90a48376a860 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 8 Nov 2014 16:38:07 +0100 Subject: ASoC: Remove snd_soc_cache_sync() implementation This function has no more non regmap user, which means we can remove the implementation of the function and associated functions and structure fields. For convenience we keep a static inline version of the function that forwards calls to regcache_sync() unconditionally. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 19 ++++-- include/trace/events/asoc.h | 25 -------- sound/soc/soc-cache.c | 152 -------------------------------------------- sound/soc/soc-core.c | 4 -- 4 files changed, 12 insertions(+), 188 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..fadcb35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -409,13 +409,9 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); -int snd_soc_cache_sync(struct snd_soc_codec *codec); int snd_soc_cache_init(struct snd_soc_codec *codec); int snd_soc_cache_exit(struct snd_soc_codec *codec); -int snd_soc_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value); -int snd_soc_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value); + int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -791,13 +787,11 @@ struct snd_soc_codec { unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int cache_init:1; /* codec cache has been initialized */ - u32 cache_sync; /* Cache needs to be synced to hardware */ /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ hw_write_t hw_write; void *reg_cache; - struct mutex cache_rw_mutex; /* component */ struct snd_soc_component component; @@ -1264,6 +1258,17 @@ unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); +/** + * snd_soc_cache_sync() - Sync the register cache with the hardware + * @codec: CODEC to sync + * + * Note: This function will call regcache_sync() + */ +static inline int snd_soc_cache_sync(struct snd_soc_codec *codec) +{ + return regcache_sync(codec->component.regmap); +} + /* component IO */ int snd_soc_component_read(struct snd_soc_component *component, unsigned int reg, unsigned int *val); diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index b04ee7e..88cf39d 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -288,31 +288,6 @@ TRACE_EVENT(snd_soc_jack_notify, TP_printk("jack=%s %x", __get_str(name), (int)__entry->val) ); -TRACE_EVENT(snd_soc_cache_sync, - - TP_PROTO(struct snd_soc_codec *codec, const char *type, - const char *status), - - TP_ARGS(codec, type, status), - - TP_STRUCT__entry( - __string( name, codec->component.name) - __string( status, status ) - __string( type, type ) - __field( int, id ) - ), - - TP_fast_assign( - __assign_str(name, codec->component.name); - __assign_str(status, status); - __assign_str(type, type); - __entry->id = codec->component.id; - ), - - TP_printk("codec=%s.%d type=%s status=%s", __get_str(name), - (int)__entry->id, __get_str(type), __get_str(status)) -); - #endif /* _TRACE_ASOC_H */ /* This part must be outside protection */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6dab817..07f4335 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -15,56 +15,6 @@ #include #include -#include - -static bool snd_soc_set_cache_val(void *base, unsigned int idx, - unsigned int val, unsigned int word_size) -{ - switch (word_size) { - case 1: { - u8 *cache = base; - if (cache[idx] == val) - return true; - cache[idx] = val; - break; - } - case 2: { - u16 *cache = base; - if (cache[idx] == val) - return true; - cache[idx] = val; - break; - } - default: - WARN(1, "Invalid word_size %d\n", word_size); - break; - } - return false; -} - -static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, - unsigned int word_size) -{ - if (!base) - return -1; - - switch (word_size) { - case 1: { - const u8 *cache = base; - return cache[idx]; - } - case 2: { - const u16 *cache = base; - return cache[idx]; - } - default: - WARN(1, "Invalid word_size %d\n", word_size); - break; - } - /* unreachable */ - return -1; -} - int snd_soc_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv = codec->driver; @@ -75,8 +25,6 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) if (!reg_size) return 0; - mutex_init(&codec->cache_rw_mutex); - dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", codec->component.name); @@ -103,103 +51,3 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) codec->reg_cache = NULL; return 0; } - -/** - * snd_soc_cache_read: Fetch the value of a given register from the cache. - * - * @codec: CODEC to configure. - * @reg: The register index. - * @value: The value to be returned. - */ -int snd_soc_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - if (!value) - return -EINVAL; - - mutex_lock(&codec->cache_rw_mutex); - if (!ZERO_OR_NULL_PTR(codec->reg_cache)) - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); - mutex_unlock(&codec->cache_rw_mutex); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_cache_read); - -/** - * snd_soc_cache_write: Set the value of a given register in the cache. - * - * @codec: CODEC to configure. - * @reg: The register index. - * @value: The new register value. - */ -int snd_soc_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - mutex_lock(&codec->cache_rw_mutex); - if (!ZERO_OR_NULL_PTR(codec->reg_cache)) - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); - mutex_unlock(&codec->cache_rw_mutex); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_cache_write); - -static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) -{ - int i; - int ret; - const struct snd_soc_codec_driver *codec_drv; - unsigned int val; - - codec_drv = codec->driver; - for (i = 0; i < codec_drv->reg_cache_size; ++i) { - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - if (codec_drv->reg_cache_default) - if (snd_soc_get_cache_val(codec_drv->reg_cache_default, - i, codec_drv->reg_word_size) == val) - continue; - - ret = snd_soc_write(codec, i, val); - if (ret) - return ret; - dev_dbg(codec->dev, "ASoC: Synced register %#x, value = %#x\n", - i, val); - } - return 0; -} - -/** - * snd_soc_cache_sync: Sync the register cache with the hardware. - * - * @codec: CODEC to configure. - * - * Any registers that should not be synced should be marked as - * volatile. In general drivers can choose not to use the provided - * syncing functionality if they so require. - */ -int snd_soc_cache_sync(struct snd_soc_codec *codec) -{ - const char *name = "flat"; - int ret; - - if (codec->component.regmap) - return regcache_sync(codec->component.regmap); - - if (!codec->cache_sync) - return 0; - - dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n", - codec->component.name); - trace_snd_soc_cache_sync(codec, name, "start"); - ret = snd_soc_flat_cache_sync(codec); - if (!ret) - codec->cache_sync = 0; - trace_snd_soc_cache_sync(codec, name, "end"); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_cache_sync); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..4e0e32b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -309,9 +309,6 @@ static void soc_init_codec_debugfs(struct snd_soc_component *component) { struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, - &codec->cache_sync); - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->component.debugfs_root, codec, &codec_reg_fops); @@ -656,7 +653,6 @@ int snd_soc_suspend(struct device *dev) if (codec->driver->suspend) codec->driver->suspend(codec); codec->suspended = 1; - codec->cache_sync = 1; if (codec->component.regmap) regcache_mark_dirty(codec->component.regmap); /* deactivate pins to sleep state */ -- cgit v1.1 From fbace43e8817113475ebda00e28593baa436a131 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Sat, 8 Nov 2014 14:40:17 +0100 Subject: ASoC: tfa9879: New driver for NXP Semiconductors TFA9879 amplifier. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- MAINTAINERS | 6 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tfa9879.c | 328 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tfa9879.h | 202 ++++++++++++++++++++++++++++ 5 files changed, 543 insertions(+) create mode 100644 sound/soc/codecs/tfa9879.c create mode 100644 sound/soc/codecs/tfa9879.h diff --git a/MAINTAINERS b/MAINTAINERS index a20df9b..28a3329 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6565,6 +6565,12 @@ S: Supported F: drivers/gpu/drm/i2c/tda998x_drv.c F: include/drm/i2c/tda998x.h +NXP TFA9879 DRIVER +M: Peter Rosin +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +S: Maintained +F: sound/soc/codecs/tfa9879* + OMAP SUPPORT M: Tony Lindgren L: linux-omap@vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..99b34a9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -101,6 +101,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C + select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -577,6 +578,10 @@ config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C +config SND_SOC_TFA9879 + tristate "NXP Semiconductors TFA9879 amplifier" + depends on I2C + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..ccfa2ab 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -101,6 +101,7 @@ snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o +snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -274,6 +275,7 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o +obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c new file mode 100644 index 0000000..16f1b71 --- /dev/null +++ b/sound/soc/codecs/tfa9879.c @@ -0,0 +1,328 @@ +/* + * tfa9879.c -- driver for NXP Semiconductors TFA9879 + * + * Copyright (C) 2014 Axentia Technologies AB + * Author: Peter Rosin + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "tfa9879.h" + +struct tfa9879_priv { + struct regmap *regmap; + int lsb_justified; +}; + +static int tfa9879_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec); + int fs; + int i2s_set = 0; + + switch (params_rate(params)) { + case 8000: + fs = TFA9879_I2S_FS_8000; + break; + case 11025: + fs = TFA9879_I2S_FS_11025; + break; + case 12000: + fs = TFA9879_I2S_FS_12000; + break; + case 16000: + fs = TFA9879_I2S_FS_16000; + break; + case 22050: + fs = TFA9879_I2S_FS_22050; + break; + case 24000: + fs = TFA9879_I2S_FS_24000; + break; + case 32000: + fs = TFA9879_I2S_FS_32000; + break; + case 44100: + fs = TFA9879_I2S_FS_44100; + break; + case 48000: + fs = TFA9879_I2S_FS_48000; + break; + case 64000: + fs = TFA9879_I2S_FS_64000; + break; + case 88200: + fs = TFA9879_I2S_FS_88200; + break; + case 96000: + fs = TFA9879_I2S_FS_96000; + break; + default: + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + i2s_set = TFA9879_I2S_SET_LSB_J_16; + break; + case 24: + i2s_set = TFA9879_I2S_SET_LSB_J_24; + break; + default: + return -EINVAL; + } + + if (tfa9879->lsb_justified) + snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + TFA9879_I2S_SET_MASK, + i2s_set << TFA9879_I2S_SET_SHIFT); + + snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + TFA9879_I2S_FS_MASK, + fs << TFA9879_I2S_FS_SHIFT); + return 0; +} + +static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + snd_soc_update_bits(codec, TFA9879_MISC_CONTROL, + TFA9879_S_MUTE_MASK, + !!mute << TFA9879_S_MUTE_SHIFT); + + return 0; +} + +static int tfa9879_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct tfa9879_priv *tfa9879 = snd_soc_codec_get_drvdata(codec); + int i2s_set; + int sck_pol; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sck_pol = TFA9879_SCK_POL_NORMAL; + break; + case SND_SOC_DAIFMT_IB_NF: + sck_pol = TFA9879_SCK_POL_INVERSE; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + tfa9879->lsb_justified = 0; + i2s_set = TFA9879_I2S_SET_I2S_24; + break; + case SND_SOC_DAIFMT_LEFT_J: + tfa9879->lsb_justified = 0; + i2s_set = TFA9879_I2S_SET_MSB_J_24; + break; + case SND_SOC_DAIFMT_RIGHT_J: + tfa9879->lsb_justified = 1; + i2s_set = TFA9879_I2S_SET_LSB_J_24; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + TFA9879_SCK_POL_MASK, + sck_pol << TFA9879_SCK_POL_SHIFT); + snd_soc_update_bits(codec, TFA9879_SERIAL_INTERFACE_1, + TFA9879_I2S_SET_MASK, + i2s_set << TFA9879_I2S_SET_SHIFT); + return 0; +} + +static struct reg_default tfa9879_regs[] = { + { TFA9879_DEVICE_CONTROL, 0x0000 }, /* 0x00 */ + { TFA9879_SERIAL_INTERFACE_1, 0x0a18 }, /* 0x01 */ + { TFA9879_PCM_IOM2_FORMAT_1, 0x0007 }, /* 0x02 */ + { TFA9879_SERIAL_INTERFACE_2, 0x0a18 }, /* 0x03 */ + { TFA9879_PCM_IOM2_FORMAT_2, 0x0007 }, /* 0x04 */ + { TFA9879_EQUALIZER_A1, 0x59dd }, /* 0x05 */ + { TFA9879_EQUALIZER_A2, 0xc63e }, /* 0x06 */ + { TFA9879_EQUALIZER_B1, 0x651a }, /* 0x07 */ + { TFA9879_EQUALIZER_B2, 0xe53e }, /* 0x08 */ + { TFA9879_EQUALIZER_C1, 0x4616 }, /* 0x09 */ + { TFA9879_EQUALIZER_C2, 0xd33e }, /* 0x0a */ + { TFA9879_EQUALIZER_D1, 0x4df3 }, /* 0x0b */ + { TFA9879_EQUALIZER_D2, 0xea3e }, /* 0x0c */ + { TFA9879_EQUALIZER_E1, 0x5ee0 }, /* 0x0d */ + { TFA9879_EQUALIZER_E2, 0xf93e }, /* 0x0e */ + { TFA9879_BYPASS_CONTROL, 0x0093 }, /* 0x0f */ + { TFA9879_DYNAMIC_RANGE_COMPR, 0x92ba }, /* 0x10 */ + { TFA9879_BASS_TREBLE, 0x12a5 }, /* 0x11 */ + { TFA9879_HIGH_PASS_FILTER, 0x0004 }, /* 0x12 */ + { TFA9879_VOLUME_CONTROL, 0x10bd }, /* 0x13 */ + { TFA9879_MISC_CONTROL, 0x0000 }, /* 0x14 */ +}; + +static bool tfa9879_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == TFA9879_MISC_STATUS; +} + +static const DECLARE_TLV_DB_SCALE(volume_tlv, -7050, 50, 1); +static const DECLARE_TLV_DB_SCALE(tb_gain_tlv, -1800, 200, 0); +static const char * const tb_freq_text[] = { + "Low", "Mid", "High" +}; +static const struct soc_enum treble_freq_enum = + SOC_ENUM_SINGLE(TFA9879_BASS_TREBLE, TFA9879_F_TRBLE_SHIFT, + ARRAY_SIZE(tb_freq_text), tb_freq_text); +static const struct soc_enum bass_freq_enum = + SOC_ENUM_SINGLE(TFA9879_BASS_TREBLE, TFA9879_F_BASS_SHIFT, + ARRAY_SIZE(tb_freq_text), tb_freq_text); + +static const struct snd_kcontrol_new tfa9879_controls[] = { + SOC_SINGLE_TLV("PCM Playback Volume", TFA9879_VOLUME_CONTROL, + TFA9879_VOL_SHIFT, 0xbd, 1, volume_tlv), + SOC_SINGLE_TLV("Treble Volume", TFA9879_BASS_TREBLE, + TFA9879_G_TRBLE_SHIFT, 18, 0, tb_gain_tlv), + SOC_SINGLE_TLV("Bass Volume", TFA9879_BASS_TREBLE, + TFA9879_G_BASS_SHIFT, 18, 0, tb_gain_tlv), + SOC_ENUM("Treble Corner Freq", treble_freq_enum), + SOC_ENUM("Bass Corner Freq", bass_freq_enum), +}; + +static const struct snd_soc_dapm_widget tfa9879_dapm_widgets[] = { +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DAC", NULL, TFA9879_DEVICE_CONTROL, TFA9879_OPMODE_SHIFT, 0), +SND_SOC_DAPM_OUTPUT("LINEOUT"), +SND_SOC_DAPM_SUPPLY("POWER", TFA9879_DEVICE_CONTROL, TFA9879_POWERUP_SHIFT, 0, + NULL, 0), +}; + +static const struct snd_soc_dapm_route tfa9879_dapm_routes[] = { + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + + { "LINEOUT", NULL, "DAC" }, + + { "DAC", NULL, "POWER" }, +}; + +static const struct snd_soc_codec_driver tfa9879_codec = { + .controls = tfa9879_controls, + .num_controls = ARRAY_SIZE(tfa9879_controls), + + .dapm_widgets = tfa9879_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tfa9879_dapm_widgets), + .dapm_routes = tfa9879_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tfa9879_dapm_routes), +}; + +static const struct regmap_config tfa9879_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .volatile_reg = tfa9879_volatile_reg, + .max_register = TFA9879_MISC_STATUS, + .reg_defaults = tfa9879_regs, + .num_reg_defaults = ARRAY_SIZE(tfa9879_regs), + .cache_type = REGCACHE_RBTREE, +}; + +static const struct snd_soc_dai_ops tfa9879_dai_ops = { + .hw_params = tfa9879_hw_params, + .digital_mute = tfa9879_digital_mute, + .set_fmt = tfa9879_set_fmt, +}; + +#define TFA9879_RATES SNDRV_PCM_RATE_8000_96000 + +#define TFA9879_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver tfa9879_dai = { + .name = "tfa9879-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TFA9879_RATES, + .formats = TFA9879_FORMATS, }, + .ops = &tfa9879_dai_ops, +}; + +static int tfa9879_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tfa9879_priv *tfa9879; + int i; + + tfa9879 = devm_kzalloc(&i2c->dev, sizeof(*tfa9879), GFP_KERNEL); + if (IS_ERR(tfa9879)) + return PTR_ERR(tfa9879); + + i2c_set_clientdata(i2c, tfa9879); + + tfa9879->regmap = devm_regmap_init_i2c(i2c, &tfa9879_regmap); + if (IS_ERR(tfa9879->regmap)) + return PTR_ERR(tfa9879->regmap); + + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(tfa9879_regs); i++) + regmap_write(tfa9879->regmap, + tfa9879_regs[i].reg, tfa9879_regs[i].def); + + return snd_soc_register_codec(&i2c->dev, &tfa9879_codec, + &tfa9879_dai, 1); +} + +static int tfa9879_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id tfa9879_i2c_id[] = { + { "tfa9879", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); + +static struct i2c_driver tfa9879_i2c_driver = { + .driver = { + .name = "tfa9879", + .owner = THIS_MODULE, + }, + .probe = tfa9879_i2c_probe, + .remove = tfa9879_i2c_remove, + .id_table = tfa9879_i2c_id, +}; + +module_i2c_driver(tfa9879_i2c_driver); + +MODULE_DESCRIPTION("ASoC NXP Semiconductors TFA9879 driver"); +MODULE_AUTHOR("Peter Rosin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tfa9879.h b/sound/soc/codecs/tfa9879.h new file mode 100644 index 0000000..3408c90 --- /dev/null +++ b/sound/soc/codecs/tfa9879.h @@ -0,0 +1,202 @@ +/* + * tfa9879.h -- driver for NXP Semiconductors TFA9879 + * + * Copyright (C) 2014 Axentia Technologies AB + * Author: Peter Rosin + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _TFA9879_H +#define _TFA9879_H + +#define TFA9879_DEVICE_CONTROL 0x00 +#define TFA9879_SERIAL_INTERFACE_1 0x01 +#define TFA9879_PCM_IOM2_FORMAT_1 0x02 +#define TFA9879_SERIAL_INTERFACE_2 0x03 +#define TFA9879_PCM_IOM2_FORMAT_2 0x04 +#define TFA9879_EQUALIZER_A1 0x05 +#define TFA9879_EQUALIZER_A2 0x06 +#define TFA9879_EQUALIZER_B1 0x07 +#define TFA9879_EQUALIZER_B2 0x08 +#define TFA9879_EQUALIZER_C1 0x09 +#define TFA9879_EQUALIZER_C2 0x0a +#define TFA9879_EQUALIZER_D1 0x0b +#define TFA9879_EQUALIZER_D2 0x0c +#define TFA9879_EQUALIZER_E1 0x0d +#define TFA9879_EQUALIZER_E2 0x0e +#define TFA9879_BYPASS_CONTROL 0x0f +#define TFA9879_DYNAMIC_RANGE_COMPR 0x10 +#define TFA9879_BASS_TREBLE 0x11 +#define TFA9879_HIGH_PASS_FILTER 0x12 +#define TFA9879_VOLUME_CONTROL 0x13 +#define TFA9879_MISC_CONTROL 0x14 +#define TFA9879_MISC_STATUS 0x15 + +/* TFA9879_DEVICE_CONTROL */ +#define TFA9879_INPUT_SEL_MASK 0x0010 +#define TFA9879_INPUT_SEL_SHIFT 4 +#define TFA9879_OPMODE_MASK 0x0008 +#define TFA9879_OPMODE_SHIFT 3 +#define TFA9879_RESET_MASK 0x0002 +#define TFA9879_RESET_SHIFT 1 +#define TFA9879_POWERUP_MASK 0x0001 +#define TFA9879_POWERUP_SHIFT 0 + +/* TFA9879_SERIAL_INTERFACE */ +#define TFA9879_MONO_SEL_MASK 0x0c00 +#define TFA9879_MONO_SEL_SHIFT 10 +#define TFA9879_MONO_SEL_LEFT 0 +#define TFA9879_MONO_SEL_RIGHT 1 +#define TFA9879_MONO_SEL_BOTH 2 +#define TFA9879_I2S_FS_MASK 0x03c0 +#define TFA9879_I2S_FS_SHIFT 6 +#define TFA9879_I2S_FS_8000 0 +#define TFA9879_I2S_FS_11025 1 +#define TFA9879_I2S_FS_12000 2 +#define TFA9879_I2S_FS_16000 3 +#define TFA9879_I2S_FS_22050 4 +#define TFA9879_I2S_FS_24000 5 +#define TFA9879_I2S_FS_32000 6 +#define TFA9879_I2S_FS_44100 7 +#define TFA9879_I2S_FS_48000 8 +#define TFA9879_I2S_FS_64000 9 +#define TFA9879_I2S_FS_88200 10 +#define TFA9879_I2S_FS_96000 11 +#define TFA9879_I2S_SET_MASK 0x0038 +#define TFA9879_I2S_SET_SHIFT 3 +#define TFA9879_I2S_SET_MSB_J_24 2 +#define TFA9879_I2S_SET_I2S_24 3 +#define TFA9879_I2S_SET_LSB_J_16 4 +#define TFA9879_I2S_SET_LSB_J_18 5 +#define TFA9879_I2S_SET_LSB_J_20 6 +#define TFA9879_I2S_SET_LSB_J_24 7 +#define TFA9879_SCK_POL_MASK 0x0004 +#define TFA9879_SCK_POL_SHIFT 2 +#define TFA9879_SCK_POL_NORMAL 0 +#define TFA9879_SCK_POL_INVERSE 1 +#define TFA9879_I_MODE_MASK 0x0003 +#define TFA9879_I_MODE_SHIFT 0 +#define TFA9879_I_MODE_I2S 0 +#define TFA9879_I_MODE_PCM_IOM2_SHORT 1 +#define TFA9879_I_MODE_PCM_IOM2_LONG 2 + +/* TFA9879_PCM_IOM2_FORMAT */ +#define TFA9879_PCM_FS_MASK 0x0800 +#define TFA9879_PCM_FS_SHIFT 11 +#define TFA9879_A_LAW_MASK 0x0400 +#define TFA9879_A_LAW_SHIFT 10 +#define TFA9879_PCM_COMP_MASK 0x0200 +#define TFA9879_PCM_COMP_SHIFT 9 +#define TFA9879_PCM_DL_MASK 0x0100 +#define TFA9879_PCM_DL_SHIFT 8 +#define TFA9879_D1_SLOT_MASK 0x00f0 +#define TFA9879_D1_SLOT_SHIFT 4 +#define TFA9879_D2_SLOT_MASK 0x000f +#define TFA9879_D2_SLOT_SHIFT 0 + +/* TFA9879_EQUALIZER_X1 */ +#define TFA9879_T1_MASK 0x8000 +#define TFA9879_T1_SHIFT 15 +#define TFA9879_K1M_MASK 0x7ff0 +#define TFA9879_K1M_SHIFT 4 +#define TFA9879_K1E_MASK 0x000f +#define TFA9879_K1E_SHIFT 0 + +/* TFA9879_EQUALIZER_X2 */ +#define TFA9879_T2_MASK 0x8000 +#define TFA9879_T2_SHIFT 15 +#define TFA9879_K2M_MASK 0x7800 +#define TFA9879_K2M_SHIFT 11 +#define TFA9879_K2E_MASK 0x0700 +#define TFA9879_K2E_SHIFT 8 +#define TFA9879_K0_MASK 0x00fe +#define TFA9879_K0_SHIFT 1 +#define TFA9879_S_MASK 0x0001 +#define TFA9879_S_SHIFT 0 + +/* TFA9879_BYPASS_CONTROL */ +#define TFA9879_L_OCP_MASK 0x00c0 +#define TFA9879_L_OCP_SHIFT 6 +#define TFA9879_L_OTP_MASK 0x0030 +#define TFA9879_L_OTP_SHIFT 4 +#define TFA9879_CLIPCTRL_MASK 0x0008 +#define TFA9879_CLIPCTRL_SHIFT 3 +#define TFA9879_HPF_BP_MASK 0x0004 +#define TFA9879_HPF_BP_SHIFT 2 +#define TFA9879_DRC_BP_MASK 0x0002 +#define TFA9879_DRC_BP_SHIFT 1 +#define TFA9879_EQ_BP_MASK 0x0001 +#define TFA9879_EQ_BP_SHIFT 0 + +/* TFA9879_DYNAMIC_RANGE_COMPR */ +#define TFA9879_AT_LVL_MASK 0xf000 +#define TFA9879_AT_LVL_SHIFT 12 +#define TFA9879_AT_RATE_MASK 0x0f00 +#define TFA9879_AT_RATE_SHIFT 8 +#define TFA9879_RL_LVL_MASK 0x00f0 +#define TFA9879_RL_LVL_SHIFT 4 +#define TFA9879_RL_RATE_MASK 0x000f +#define TFA9879_RL_RATE_SHIFT 0 + +/* TFA9879_BASS_TREBLE */ +#define TFA9879_G_TRBLE_MASK 0x3e00 +#define TFA9879_G_TRBLE_SHIFT 9 +#define TFA9879_F_TRBLE_MASK 0x0180 +#define TFA9879_F_TRBLE_SHIFT 7 +#define TFA9879_G_BASS_MASK 0x007c +#define TFA9879_G_BASS_SHIFT 2 +#define TFA9879_F_BASS_MASK 0x0003 +#define TFA9879_F_BASS_SHIFT 0 + +/* TFA9879_HIGH_PASS_FILTER */ +#define TFA9879_HP_CTRL_MASK 0x00ff +#define TFA9879_HP_CTRL_SHIFT 0 + +/* TFA9879_VOLUME_CONTROL */ +#define TFA9879_ZR_CRSS_MASK 0x1000 +#define TFA9879_ZR_CRSS_SHIFT 12 +#define TFA9879_VOL_MASK 0x00ff +#define TFA9879_VOL_SHIFT 0 + +/* TFA9879_MISC_CONTROL */ +#define TFA9879_DE_PHAS_MASK 0x0c00 +#define TFA9879_DE_PHAS_SHIFT 10 +#define TFA9879_H_MUTE_MASK 0x0200 +#define TFA9879_H_MUTE_SHIFT 9 +#define TFA9879_S_MUTE_MASK 0x0100 +#define TFA9879_S_MUTE_SHIFT 8 +#define TFA9879_P_LIM_MASK 0x00ff +#define TFA9879_P_LIM_SHIFT 0 + +/* TFA9879_MISC_STATUS */ +#define TFA9879_PS_MASK 0x4000 +#define TFA9879_PS_SHIFT 14 +#define TFA9879_PORA_MASK 0x2000 +#define TFA9879_PORA_SHIFT 13 +#define TFA9879_AMP_MASK 0x0600 +#define TFA9879_AMP_SHIFT 9 +#define TFA9879_IBP_2_MASK 0x0100 +#define TFA9879_IBP_2_SHIFT 8 +#define TFA9879_OFP_2_MASK 0x0080 +#define TFA9879_OFP_2_SHIFT 7 +#define TFA9879_UFP_2_MASK 0x0040 +#define TFA9879_UFP_2_SHIFT 6 +#define TFA9879_IBP_1_MASK 0x0020 +#define TFA9879_IBP_1_SHIFT 5 +#define TFA9879_OFP_1_MASK 0x0010 +#define TFA9879_OFP_1_SHIFT 4 +#define TFA9879_UFP_1_MASK 0x0008 +#define TFA9879_UFP_1_SHIFT 3 +#define TFA9879_OCPOKA_MASK 0x0004 +#define TFA9879_OCPOKA_SHIFT 2 +#define TFA9879_OCPOKB_MASK 0x0002 +#define TFA9879_OCPOKB_SHIFT 1 +#define TFA9879_OTPOK_MASK 0x0001 +#define TFA9879_OTPOK_SHIFT 0 + +#endif -- cgit v1.1 From 1fb8510cdb5b7befe8a59f533c7fc12ef0dac73e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 7 Nov 2014 17:08:28 +0100 Subject: ALSA: pcm: Add snd_pcm_stop_xrun() helper Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai --- drivers/media/pci/saa7134/saa7134-alsa.c | 4 +--- include/sound/pcm.h | 1 + sound/arm/pxa2xx-pcm-lib.c | 4 +--- sound/core/pcm_native.c | 23 +++++++++++++++++++++++ sound/firewire/amdtp.c | 8 ++------ sound/firewire/isight.c | 10 ++-------- sound/pci/asihpi/asihpi.c | 5 +---- sound/pci/atiixp.c | 4 +--- sound/pci/atiixp_modem.c | 4 +--- sound/soc/atmel/atmel-pcm-dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 9 +-------- sound/usb/6fire/pcm.c | 17 ++++------------- sound/usb/endpoint.c | 4 +--- sound/usb/misc/ua101.c | 18 ++++-------------- sound/usb/usx2y/usbusx2yaudio.c | 9 ++------- 15 files changed, 46 insertions(+), 78 deletions(-) diff --git a/drivers/media/pci/saa7134/saa7134-alsa.c b/drivers/media/pci/saa7134/saa7134-alsa.c index 4056989..ac3cd74 100644 --- a/drivers/media/pci/saa7134/saa7134-alsa.c +++ b/drivers/media/pci/saa7134/saa7134-alsa.c @@ -173,9 +173,7 @@ static void saa7134_irq_alsa_done(struct saa7134_dev *dev, dprintk("irq: overrun [full=%d/%d] - Blocks in %d\n",dev->dmasound.read_count, dev->dmasound.bufsize, dev->dmasound.blocks); spin_unlock(&dev->slock); - snd_pcm_stream_lock(dev->dmasound.substream); - snd_pcm_stop(dev->dmasound.substream,SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dev->dmasound.substream); + snd_pcm_stop_xrun(dev->dmasound.substream); return; } diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0b8daee..40289ec 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -506,6 +506,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index a61d7a9..01f8fdc 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -200,9 +200,7 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", dma_ch, dcsr); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dfb5031..a3d1221 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1098,6 +1098,29 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_SETUP); } +/** + * snd_pcm_stop_xrun - stop the running streams as XRUN + * @substream: the PCM substream instance + * @state: PCM state after stopping the stream + * + * This stops the given running substream (and all linked substreams) as XRUN. + * Unlike snd_pcm_stop(), this function takes the substream lock by itself. + * + * Return: Zero if successful, or a negative error code. + */ +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) +{ + unsigned long flags; + int ret = 0; + + snd_pcm_stream_lock_irqsave(substream, flags); + if (snd_pcm_running(substream)) + ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + return ret; +} +EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); + /* * pause callbacks */ diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 95fc2eaf..3badc70 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -1006,11 +1006,7 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s) struct snd_pcm_substream *pcm; pcm = ACCESS_ONCE(s->pcm); - if (pcm) { - snd_pcm_stream_lock_irq(pcm); - if (snd_pcm_running(pcm)) - snd_pcm_stop(pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(pcm); - } + if (pcm) + snd_pcm_stop_xrun(pcm); } EXPORT_SYMBOL(amdtp_stream_pcm_abort); diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 7ac9443..48d6dca 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -131,14 +131,8 @@ static void isight_samples(struct isight *isight, static void isight_pcm_abort(struct isight *isight) { - unsigned long flags; - - if (ACCESS_ONCE(isight->pcm_active)) { - snd_pcm_stream_lock_irqsave(isight->pcm, flags); - if (snd_pcm_running(isight->pcm)) - snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); - } + if (ACCESS_ONCE(isight->pcm_active)) + snd_pcm_stop_xrun(isight->pcm); } static void isight_dropped_samples(struct isight *isight, unsigned int total) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index ac66b32..ff9f9f1c 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,10 +769,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { - unsigned long flags; - snd_pcm_stream_lock_irqsave(s, flags); - snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(s, flags); + snd_pcm_stop_xrun(s); continue; } } else { diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7895c5d..9c1c445 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -688,9 +688,7 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 3c32413..b2f63e0 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,9 +638,7 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b79a2a8..33fb3bb 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -80,9 +80,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a609aaf..b2b1088 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -151,14 +151,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { */ static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); - - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } /** diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index ba40489..36f4115 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -679,25 +679,16 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; - unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) { - snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); - snd_pcm_stop(rt->playback.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); - } + if (rt->playback.instance) + snd_pcm_stop_xrun(rt->playback.instance); - if (rt->capture.instance) { - snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); - snd_pcm_stop(rt->capture.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); - } + if (rt->capture.instance) + snd_pcm_stop_xrun(rt->capture.instance); for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a467991..03b0744 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -391,9 +391,7 @@ static void snd_complete_urb(struct urb *urb) usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); if (ep->data_subs && ep->data_subs->pcm_substream) { substream = ep->data_subs->pcm_substream; - snd_pcm_stream_lock_irqsave(substream, flags); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } exit_clear: diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index a1bab14..9581089 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,24 +613,14 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); - snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); - } + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->capture.substream); } static void abort_alsa_playback(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); - snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); - } + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->playback.substream); } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index a63330d..61d5dc2 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -272,13 +272,8 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) for (s = 0; s < 4; s++) { struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { - if (atomic_read(&subs->state) >= state_PRERUNNING) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); - snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); - } + if (atomic_read(&subs->state) >= state_PRERUNNING) + snd_pcm_stop_xrun(subs->pcm_substream); for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; if (NULL != urb) -- cgit v1.1 From d4b8fc66f770e9b79830cfe6c342846293b99fda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 9 Nov 2014 18:21:23 +0100 Subject: ALSA: usb-audio: Allow multiple entries for the same iface in composite quirk Currently the composite quirk doesn't work when multiple entries are assigned to the same interface because it marks the interface as claimed then checks whether the interface has been already claimed for the secondary entry. But, if you look at the code, you'll notice that multiple entries are allowed if the entry is the current interface; i.e. the current behavior is anyway inconsistent, and this is an unintended shortcoming. This patch fixes the problem by marking the relevant interfaces as claimed after applying the all composite entries. This fix will be needed for the upcoming enhancements for Digidesign Mbox 1 quirks. Reviewed-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d2aa45a..e9ff3a6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -58,9 +58,17 @@ static int create_composite_quirk(struct snd_usb_audio *chip, err = snd_usb_create_quirk(chip, iface, driver, quirk); if (err < 0) return err; - if (quirk->ifnum != probed_ifnum) + } + + for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) { + iface = usb_ifnum_to_if(chip->dev, quirk->ifnum); + if (!iface) + continue; + if (quirk->ifnum != probed_ifnum && + !usb_interface_claimed(iface)) usb_driver_claim_interface(driver, iface, (void *)-1L); } + return 0; } -- cgit v1.1 From 48b217aa43abc8c3545bb9b4d7a5b525b71d6ac2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 07:38:22 +0100 Subject: ALSA: pcm: Fix document for snd_pcm_stop_xrun() Fix a copy & paste error: Warning(sound/core/pcm_native.c:1112): Excess function parameter 'state' description in 'snd_pcm_stop_xrun' The state argument was dropped from snd_pcm_stop_xrun(). Reported-by: kbuild test robot Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a3d1221..095d957 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1101,7 +1101,6 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) /** * snd_pcm_stop_xrun - stop the running streams as XRUN * @substream: the PCM substream instance - * @state: PCM state after stopping the stream * * This stops the given running substream (and all linked substreams) as XRUN. * Unlike snd_pcm_stop(), this function takes the substream lock by itself. -- cgit v1.1 From 85a8181329a919d58b7ef99211251f47d5e1049e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 07:41:59 +0100 Subject: ALSA: usb-audio: Fix Oops by composite quirk enhancement The quirk argument itself was used as iterator, so it cannot be taken back to the original value, obviously. Fixes: d4b8fc66f770 ('ALSA: usb-audio: Allow multiple entries for the same iface in composite quirk') Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e9ff3a6..809d7fa 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -43,12 +43,13 @@ static int create_composite_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, struct usb_driver *driver, - const struct snd_usb_audio_quirk *quirk) + const struct snd_usb_audio_quirk *quirk_comp) { int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber; + const struct snd_usb_audio_quirk *quirk; int err; - for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) { + for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) { iface = usb_ifnum_to_if(chip->dev, quirk->ifnum); if (!iface) continue; @@ -60,7 +61,7 @@ static int create_composite_quirk(struct snd_usb_audio *chip, return err; } - for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) { + for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) { iface = usb_ifnum_to_if(chip->dev, quirk->ifnum); if (!iface) continue; -- cgit v1.1 From 368494093354ac613a80c2e1d77602aa12473cf0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:27:33 +0200 Subject: ASoC: tlv320aic3x: Add TDM support TDM support is achieved using DSP transfer mode and setting a programmable offset which specifies where data begins with respect to the frame sync. It requires 256-clock mode if CODEC is master (not currently supported in the driver). No additional dependency if CODEC is slave. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 62 ++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 60 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f7c2a57..8770e28 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -78,6 +78,8 @@ struct aic3x_priv { struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; struct aic3x_setup_data *setup; unsigned int sysclk; + unsigned int dai_fmt; + unsigned int tdm_delay; struct list_head list; int master; int gpio_reset; @@ -1009,6 +1011,25 @@ found: return 0; } +static int aic3x_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + int delay = 0; + + /* TDM slot selection only valid in DSP_A/_B mode */ + if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A) + delay += (aic3x->tdm_delay + 1); + else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B) + delay += aic3x->tdm_delay; + + /* Configure data delay */ + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + + return 0; +} + static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -1048,7 +1069,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); u8 iface_areg, iface_breg; - int delay = 0; iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -1076,7 +1096,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): - delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -1090,10 +1109,45 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + aic3x->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* set iface */ snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); + + return 0; +} + +static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + unsigned int lsb; + + if (tx_mask != rx_mask) { + dev_err(codec->dev, "tx and rx masks must be symmetric\n"); + return -EINVAL; + } + + if (unlikely(!tx_mask)) { + dev_err(codec->dev, "tx and rx masks need to be non 0\n"); + return -EINVAL; + } + + /* TDM based on DSP mode requires slots to be adjacent */ + lsb = __ffs(tx_mask); + if ((lsb + 1) != __fls(tx_mask)) { + dev_err(codec->dev, "Invalid mask, slots must be adjacent\n"); + return -EINVAL; + } + + aic3x->tdm_delay = lsb * slot_width; + + /* DOUT in high-impedance on inactive bit clocks */ + snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA, + DOUT_TRISTATE, DOUT_TRISTATE); return 0; } @@ -1212,9 +1266,11 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops aic3x_dai_ops = { .hw_params = aic3x_hw_params, + .prepare = aic3x_prepare, .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, + .set_tdm_slot = aic3x_set_dai_tdm_slot, }; static struct snd_soc_dai_driver aic3x_dai = { diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e521ac3..89fa692 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -169,6 +169,7 @@ /* Audio serial data interface control register A bits */ #define BIT_CLK_MASTER 0x80 #define WORD_CLK_MASTER 0x40 +#define DOUT_TRISTATE 0x20 /* Codec Datapath setup register 7 */ #define FSREF_44100 (1 << 7) -- cgit v1.1 From ff7c532c3ae570981a46663d5810dda913d468d9 Mon Sep 17 00:00:00 2001 From: Pavel Machek Date: Sun, 9 Nov 2014 20:41:51 +0100 Subject: ASoC: omap: enable sound support on n900 on devicetree-based boot With device tree, it is possible (and encouraged) to build N900 kernels without CONFIG_MACH_NOKIA_RX51. Update config file to enable the driver build in this case. This makes sound work on my n900 under 3.18-rc1. Signed-off-by: Pavel Machek Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d44463a..2738b19 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -25,15 +25,15 @@ config SND_OMAP_SOC_N810 Say Y if you want to add support for SoC audio on Nokia N810. config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C + tristate "SoC Audio support for Nokia N900 (RX-51)" + depends on SND_OMAP_SOC && ARM && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 depends on GPIOLIB help - Say Y if you want to add support for SoC audio on Nokia RX-51 - hardware. This is also known as Nokia N900 product. + Say Y if you want to add support for SoC audio on Nokia N900 + cellphone. config SND_OMAP_SOC_AMS_DELTA tristate "SoC Audio support for Amstrad E3 (Delta) videophone" -- cgit v1.1 From 7771ef3286711a121b763c3620c4619f51b2acfd Mon Sep 17 00:00:00 2001 From: Anil Kumar Date: Sun, 9 Nov 2014 18:15:14 +0530 Subject: ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure If probe fails then we need to call pm_runtime_disable() to balance out the previous pm_runtime_enable() call. Signed-off-by: Anil Kumar Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 002351f..57f606e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1225,6 +1225,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = pm_runtime_get_sync(&pdev->dev); if (IS_ERR_VALUE(ret)) { dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); + pm_runtime_disable(&pdev->dev); return ret; } -- cgit v1.1 From 30cc4faf703955cd5cd07da489bd817ae43e3fec Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 9 Nov 2014 20:00:30 -0800 Subject: ASoC: rsnd: tidyup debug message format and timing Current Renesas R-Car sound driver debug message is using random format (ex "ssi0: xxx" / "SSI0 xxx" / "ssi[0]: xxx") and confusable timing ("xxx probe failed" and "xxx probed" are shown in same time) This patch fixes these Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/dvc.c | 3 ++- sound/soc/sh/rcar/gen.c | 6 +++--- sound/soc/sh/rcar/src.c | 11 +++++++---- sound/soc/sh/rcar/ssi.c | 25 ++++++++++++++++--------- 5 files changed, 29 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1922ec5..5205618 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -409,7 +409,7 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod) ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ - dev_dbg(dev, "%s [%d] %s\n", \ + dev_dbg(dev, "%s[%d] %s\n", \ rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \ (mod)->ops->func(mod, rdai); \ }) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 8504f6b..956b84e 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -85,7 +85,8 @@ static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 61dee68..a0fed66 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -78,7 +78,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - dev_dbg(dev, "r %s(%d) - %4d : %08x\n", + dev_dbg(dev, "r %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); @@ -96,7 +96,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "w %s(%d) - %4d : %08x\n", + dev_dbg(dev, "w %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); @@ -111,7 +111,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "b %s(%d) - %4d : %08x/%08x\n", + dev_dbg(dev, "b %s[%d] - %4d : %08x/%08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data, mask); regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 9183e01..4679501 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -438,7 +438,8 @@ static int rsnd_src_probe_gen1(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen1) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen1) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } @@ -578,9 +579,11 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, rsnd_info_is_playback(priv, src), src->info->dma_id); if (ret < 0) - dev_err(dev, "SRC DMA failed\n"); - - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (Gen2) failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 34e8400..cae08b7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -159,7 +159,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(j); - dev_dbg(dev, "ssi%d outputs %u Hz\n", + dev_dbg(dev, "%s[%d] outputs %u Hz\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod), rate); return 0; @@ -206,7 +207,8 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, ssi->usrcnt++; - dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw started\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, @@ -249,7 +251,8 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, clk_disable_unprepare(ssi->clk); } - dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw stopped\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } /* @@ -385,9 +388,11 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, IRQF_SHARED, dev_name(dev), ssi); if (ret) - dev_err(dev, "SSI request interrupt failed\n"); - - dev_dbg(dev, "%s (PIO) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (PIO) request interrupt failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (PIO) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -448,9 +453,11 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, dma_id); if (ret < 0) - dev_err(dev, "SSI DMA failed\n"); - - dev_dbg(dev, "%s (DMA) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (DMA) is failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (DMA) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } -- cgit v1.1 From d3a768233243b5892a9c74b85896b9e8c017b259 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 9 Nov 2014 20:00:58 -0800 Subject: ASoC: rsnd: fallback to PIO mode if DMA mode was failed Current Renesas R-Car sound driver probe will be failed if it try to use DMA mode and it couldn't use for some reasons. But PIO mode works even though in such case. This patch try to fallback to PIO mode if DMA mode probing was failed. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 74 +++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 15 ++++++++++ 2 files changed, 85 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5205618..110b99d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -349,7 +349,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_name); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); - return -EIO; + goto rsnd_dma_channel_err; } ret = dmaengine_slave_config(dma->chan, &cfg); @@ -363,8 +363,15 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, rsnd_dma_init_err: rsnd_dma_quit(priv, dma); +rsnd_dma_channel_err: - return ret; + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_dma_remove() + * rsnd_rdai_continuance_probe() + */ + return -EAGAIN; } void rsnd_dma_quit(struct rsnd_priv *priv, @@ -456,6 +463,13 @@ static int rsnd_dai_connect(struct rsnd_mod *mod, return 0; } +static void rsnd_dai_disconnect(struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + mod->io = NULL; + io->mod[mod->type] = NULL; +} + int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) { int id = rdai - priv->rdai; @@ -686,6 +700,20 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { ret; \ }) +#define rsnd_path_break(priv, io, type) \ +{ \ + struct rsnd_mod *mod; \ + int id = -1; \ + \ + if (rsnd_is_enable_path(io, type)) { \ + id = rsnd_info_id(priv, io, type); \ + if (id >= 0) { \ + mod = rsnd_##type##_mod_get(priv, id); \ + rsnd_dai_disconnect(mod, io); \ + } \ + } \ +} + static int rsnd_path_init(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -977,6 +1005,44 @@ static const struct snd_soc_component_driver rsnd_soc_component = { .name = "rsnd", }; +static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + int is_play) +{ + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + int ret; + + ret = rsnd_dai_call(probe, io, rdai); + if (ret == -EAGAIN) { + /* + * Fallback to PIO mode + */ + + /* + * call "remove" for SSI/SRC/DVC + * SSI will be switch to PIO mode if it was DMA mode + * see + * rsnd_dma_init() + * rsnd_ssi_dma_remove() + */ + rsnd_dai_call(remove, io, rdai); + + /* + * remove SRC/DVC from DAI, + */ + rsnd_path_break(priv, io, src); + rsnd_path_break(priv, io, dvc); + + /* + * retry to "probe". + * DAI has SSI which is PIO mode only now. + */ + ret = rsnd_dai_call(probe, io, rdai); + } + + return ret; +} + /* * rsnd probe */ @@ -1038,11 +1104,11 @@ static int rsnd_probe(struct platform_device *pdev) } for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(probe, &rdai->playback, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 1); if (ret) goto exit_snd_probe; - ret = rsnd_dai_call(probe, &rdai->capture, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 0); if (ret) goto exit_snd_probe; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index cae08b7..346d3dc 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -465,8 +465,23 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + /* + * fallback to PIO + * + * SSI .probe might be called again. + * see + * rsnd_rdai_continuance_probe() + */ + mod->ops = &rsnd_ssi_pio_ops; + + dev_info(dev, "%s[%d] fallback to PIO mode\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + return 0; } -- cgit v1.1 From d75249f54577d489d1642a246d3702416daa68f9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:32:18 +0200 Subject: ASoC: davinci-mcasp: Symmetric sample bits for IIS mode In IIS mode the tx and rx configuration is symmetric, the BCLK and FSYNC is shared. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 57f606e..80c54ed 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -961,6 +961,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }, .ops = &davinci_mcasp_dai_ops, + .symmetric_samplebits = 1, }, { .name = "davinci-mcasp.1", -- cgit v1.1 From d742b925244ce91f16d380befdca473e4536359b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:32:19 +0200 Subject: ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus When the bus is configured to have more BCLK then the data type demands we need to use the rotation to move the data to correct place. Reported-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 80c54ed..ea3ad74 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -490,8 +490,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, * both left and right channels), so it has to be divided by number of * tdm-slots (for I2S - divided by 2). */ - if (mcasp->bclk_lrclk_ratio) - word_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; + if (mcasp->bclk_lrclk_ratio) { + u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; + + /* + * When we have more bclk then it is needed for the data, we + * need to use the rotation to move the received samples to have + * correct alignment. + */ + rx_rotate = (slot_length - word_length) / 4; + word_length = slot_length; + } /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; -- cgit v1.1 From 1a5923da4e2e6997f03545a2b88f2cd2dea1a5c2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:32:15 +0200 Subject: ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time Instead of validating the tdm_slots parameter every time at hw_params we can do it once during probe. If the parameter is not valid (<2 or >32) print an error and fix it up. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6b1bfd9..10c2647 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -632,13 +632,7 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) u32 mask = 0; u32 busel = 0; - if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) { - dev_err(mcasp->dev, "tdm slot %d not supported\n", - mcasp->tdm_slots); - return -EINVAL; - } - - active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; + active_slots = mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -650,12 +644,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + FSXMOD(active_slots), FSXMOD(0x1FF)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + FSRMOD(active_slots), FSRMOD(0x1FF)); return 0; } @@ -1237,7 +1231,21 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } mcasp->op_mode = pdata->op_mode; - mcasp->tdm_slots = pdata->tdm_slots; + /* sanity check for tdm slots parameter */ + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) { + if (pdata->tdm_slots < 2) { + dev_err(&pdev->dev, "invalid tdm slots: %d\n", + pdata->tdm_slots); + mcasp->tdm_slots = 2; + } else if (pdata->tdm_slots > 32) { + dev_err(&pdev->dev, "invalid tdm slots: %d\n", + pdata->tdm_slots); + mcasp->tdm_slots = 32; + } else { + mcasp->tdm_slots = pdata->tdm_slots; + } + } + mcasp->num_serializer = pdata->num_serializer; #ifdef CONFIG_PM_SLEEP mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev, -- cgit v1.1 From 11277833ce8018ddd18925d2f85037bf02dcba63 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Nov 2014 12:32:16 +0200 Subject: ASoC: davinci-mcasp: Place constraint on number of channels In IIS (I2S, TDM, etc) mode the maximum number of allowed channels for either direction can be: number of serializers for the direction * tdm_slots. This constraint applicable for the first stream, while consequent stream should not have more channels then the first stream. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 60 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 60 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 10c2647..10ae75e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -90,6 +90,9 @@ struct davinci_mcasp { bool dat_port; + /* Used for comstraint setting on the second stream */ + u32 channels; + #ifdef CONFIG_PM_SLEEP struct davinci_mcasp_context context; #endif @@ -811,6 +814,9 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, davinci_config_channel_size(mcasp, word_length); + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) + mcasp->channels = channels; + return 0; } @@ -839,7 +845,61 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + u32 max_channels = 0; + int i, dir; + + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return 0; + + /* + * Limit the maximum allowed channels for the first stream: + * number of serializers for the direction * tdm slots per serializer + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = TX_MODE; + else + dir = RX_MODE; + + for (i = 0; i < mcasp->num_serializer; i++) { + if (mcasp->serial_dir[i] == dir) + max_channels++; + } + max_channels *= mcasp->tdm_slots; + /* + * If the already active stream has less channels than the calculated + * limnit based on the seirializers * tdm_slots, we need to use that as + * a constraint for the second stream. + * Otherwise (first stream or less allowed channels) we use the + * calculated constraint. + */ + if (mcasp->channels && mcasp->channels < max_channels) + max_channels = mcasp->channels; + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + 2, max_channels); + return 0; +} + +static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return; + + if (!cpu_dai->active) + mcasp->channels = 0; +} + static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, + .shutdown = davinci_mcasp_shutdown, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, -- cgit v1.1 From 18a4f55756ac945bd89d1fe63dafe36462ecba86 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 10 Nov 2014 12:32:17 +0200 Subject: ASoC: davinci-mcasp: Active slots depend on active serializers Active slots count depends on the number of channels in the stream and the number of active serializers. Each serializer will handle at most the number of channels specified via 'tdm-slots' parameter in DT. There are two possible scenarios: - Single serializer: channel count fits in the max slots supported by McASP serializers, active slots is same as channel count - Multiple serializers: channel count is bigger than max slots supported by a serializer. Channel count determines how many serializers are needed at their max slot count configuration Signed-off-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 10ae75e..a9822c7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -629,13 +629,29 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) +static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, + int channels) { int i, active_slots; + int total_slots; + int active_serializers; u32 mask = 0; u32 busel = 0; - active_slots = mcasp->tdm_slots; + total_slots = mcasp->tdm_slots; + + /* + * If more than one serializer is needed, then use them with + * their specified tdm_slots count. Otherwise, one serializer + * can cope with the transaction using as many slots as channels + * in the stream, requires channels symmetry + */ + active_serializers = (channels + total_slots - 1) / total_slots; + if (active_serializers == 1) + active_slots = channels; + else + active_slots = total_slots; + for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -647,12 +663,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(active_slots), FSXMOD(0x1FF)); + FSXMOD(total_slots), FSXMOD(0x1FF)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(active_slots), FSRMOD(0x1FF)); + FSRMOD(total_slots), FSRMOD(0x1FF)); return 0; } @@ -766,7 +782,8 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else - ret = mcasp_i2s_hw_param(mcasp, substream->stream); + ret = mcasp_i2s_hw_param(mcasp, substream->stream, + channels); if (ret) return ret; -- cgit v1.1 From 3539cacff2031f6d47881c5f3a4932b0ad5ec224 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 9 Nov 2014 19:52:06 -0800 Subject: ASoC: rsnd: Add Volume Ramp support This patch adds Volume Ramp to Renesas sound driver. amixer set "DVC Out" 100% amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" amixer set "DVC Out Ramp" on aplay xxx.wav & amixer set "DVC Out" 80% // Volume Down amixer set "DVC Out" 100% // Volume Up Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dvc.c | 87 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/sh/rcar/gen.c | 3 ++ sound/soc/sh/rcar/rsnd.h | 6 ++++ 3 files changed, 93 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 956b84e..e2c8473 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -38,6 +38,9 @@ struct rsnd_dvc { struct clk *clk; struct rsnd_dvc_cfg_m volume; struct rsnd_dvc_cfg_m mute; + struct rsnd_dvc_cfg_s ren; /* Ramp Enable */ + struct rsnd_dvc_cfg_s rup; /* Ramp Rate Up */ + struct rsnd_dvc_cfg_s rdown; /* Ramp Rate Down */ }; #define rsnd_mod_to_dvc(_mod) \ @@ -49,9 +52,37 @@ struct rsnd_dvc { ((pos) = (struct rsnd_dvc *)(priv)->dvc + i); \ i++) +static const char const *dvc_ramp_rate[] = { + "128 dB/1 step", /* 00000 */ + "64 dB/1 step", /* 00001 */ + "32 dB/1 step", /* 00010 */ + "16 dB/1 step", /* 00011 */ + "8 dB/1 step", /* 00100 */ + "4 dB/1 step", /* 00101 */ + "2 dB/1 step", /* 00110 */ + "1 dB/1 step", /* 00111 */ + "0.5 dB/1 step", /* 01000 */ + "0.25 dB/1 step", /* 01001 */ + "0.125 dB/1 step", /* 01010 */ + "0.125 dB/2 steps", /* 01011 */ + "0.125 dB/4 steps", /* 01100 */ + "0.125 dB/8 steps", /* 01101 */ + "0.125 dB/16 steps", /* 01110 */ + "0.125 dB/32 steps", /* 01111 */ + "0.125 dB/64 steps", /* 10000 */ + "0.125 dB/128 steps", /* 10001 */ + "0.125 dB/256 steps", /* 10010 */ + "0.125 dB/512 steps", /* 10011 */ + "0.125 dB/1024 steps", /* 10100 */ + "0.125 dB/2048 steps", /* 10101 */ + "0.125 dB/4096 steps", /* 10110 */ + "0.125 dB/8192 steps", /* 10111 */ +}; + static void rsnd_dvc_volume_update(struct rsnd_mod *mod) { struct rsnd_dvc *dvc = rsnd_mod_to_dvc(mod); + u32 val[RSND_DVC_CHANNELS]; u32 dvucr = 0; u32 mute = 0; int i; @@ -62,10 +93,35 @@ static void rsnd_dvc_volume_update(struct rsnd_mod *mod) /* Disable DVC Register access */ rsnd_mod_write(mod, DVC_DVUER, 0); + /* Enable Ramp */ + if (dvc->ren.val) { + dvucr |= 0x10; + + /* Digital Volume Max */ + for (i = 0; i < RSND_DVC_CHANNELS; i++) + val[i] = dvc->volume.cfg.max; + + rsnd_mod_write(mod, DVC_VRCTR, 0xff); + rsnd_mod_write(mod, DVC_VRPDR, dvc->rup.val << 8 | + dvc->rdown.val); + /* + * FIXME !! + * use scale-downed Digital Volume + * as Volume Ramp + * 7F FFFF -> 3FF + */ + rsnd_mod_write(mod, DVC_VRDBR, + 0x3ff - (dvc->volume.val[0] >> 13)); + + } else { + for (i = 0; i < RSND_DVC_CHANNELS; i++) + val[i] = dvc->volume.val[i]; + } + /* Enable Digital Volume */ - dvucr = 0x100; - rsnd_mod_write(mod, DVC_VOL0R, dvc->volume.val[0]); - rsnd_mod_write(mod, DVC_VOL1R, dvc->volume.val[1]); + dvucr |= 0x100; + rsnd_mod_write(mod, DVC_VOL0R, val[0]); + rsnd_mod_write(mod, DVC_VOL1R, val[1]); /* Enable Mute */ if (mute) { @@ -324,6 +380,31 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, if (ret < 0) return ret; + /* Ramp */ + ret = _rsnd_dvc_pcm_new_s(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Ramp Switch" : "DVC In Ramp Switch", + &dvc->ren, 1); + if (ret < 0) + return ret; + + ret = _rsnd_dvc_pcm_new_e(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", + &dvc->rup, + dvc_ramp_rate, ARRAY_SIZE(dvc_ramp_rate)); + if (ret < 0) + return ret; + + ret = _rsnd_dvc_pcm_new_e(mod, rdai, rtd, + rsnd_dai_is_play(rdai, io) ? + "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", + &dvc->rdown, + dvc_ramp_rate, ARRAY_SIZE(dvc_ramp_rate)); + + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index a0fed66..87a6f2d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -324,6 +324,9 @@ static int rsnd_gen2_probe(struct platform_device *pdev, RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100), RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100), RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100), + RSND_GEN_M_REG(DVC_VRCTR, 0xe18, 0x100), + RSND_GEN_M_REG(DVC_VRPDR, 0xe1c, 0x100), + RSND_GEN_M_REG(DVC_VRDBR, 0xe20, 0x100), RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100), RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index d119adf..ed44ca8 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -91,6 +91,9 @@ enum rsnd_reg { RSND_REG_SHARE20, RSND_REG_SHARE21, RSND_REG_SHARE22, + RSND_REG_SHARE23, + RSND_REG_SHARE24, + RSND_REG_SHARE25, RSND_REG_MAX, }; @@ -129,6 +132,9 @@ enum rsnd_reg { #define RSND_REG_CMD_CTRL RSND_REG_SHARE20 #define RSND_REG_CMDOUT_TIMSEL RSND_REG_SHARE21 #define RSND_REG_BUSIF_DALIGN RSND_REG_SHARE22 +#define RSND_REG_DVC_VRCTR RSND_REG_SHARE23 +#define RSND_REG_DVC_VRPDR RSND_REG_SHARE24 +#define RSND_REG_DVC_VRDBR RSND_REG_SHARE25 struct rsnd_of_data; struct rsnd_priv; -- cgit v1.1 From 0099c762855eeee8d3eacc11fcc1e0819e77b2ed Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Sun, 9 Nov 2014 12:38:56 +0100 Subject: ASoC: simple-card: Remove useless casts There is no need to cast the cpu_of_node or codec_of_node of the dai_links when calling of_put_node. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 7 ++----- sound/soc/samsung/odroidx2_max98090.c | 4 ++-- sound/soc/ux500/mop500.c | 6 ++---- 3 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index cd49d50..3e3fec4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -457,16 +457,13 @@ static int asoc_simple_card_unref(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct snd_soc_dai_link *dai_link; - struct device_node *np; int num_links; for (num_links = 0, dai_link = card->dai_link; num_links < card->num_links; num_links++, dai_link++) { - np = (struct device_node *) dai_link->cpu_of_node; - of_node_put(np); - np = (struct device_node *) dai_link->codec_of_node; - of_node_put(np); + of_node_put(dai_link->cpu_of_node); + of_node_put(dai_link->codec_of_node); } return 0; } diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 3c8f604..d7640e7 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -153,8 +153,8 @@ static int odroidx2_audio_remove(struct platform_device *pdev) snd_soc_unregister_card(card); - of_node_put((struct device_node *)odroidx2_dai[0].cpu_of_node); - of_node_put((struct device_node *)odroidx2_dai[0].codec_of_node); + of_node_put(odroidx2_dai[0].cpu_of_node); + of_node_put(odroidx2_dai[0].codec_of_node); return 0; } diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index b3b66aa..ea9ba284 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -64,11 +64,9 @@ static void mop500_of_node_put(void) for (i = 0; i < 2; i++) { if (mop500_dai_links[i].cpu_of_node) - of_node_put((struct device_node *) - mop500_dai_links[i].cpu_of_node); + of_node_put(mop500_dai_links[i].cpu_of_node); if (mop500_dai_links[i].codec_of_node) - of_node_put((struct device_node *) - mop500_dai_links[i].codec_of_node); + of_node_put(mop500_dai_links[i].codec_of_node); } } -- cgit v1.1 From 52ef6284a840bdef50b6ed505bdda014dd769cab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:00:57 +0100 Subject: ASoC: ab8500-codec: Move control lock to the driver level The ab8500 driver uses a driver specific lock to protect concurrent access to some of the control put/get handlers and uses the snd_soc_codec mutex for some others. This patch updates the driver to consistently use the driver specific lock for all controls. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index fd43827..7dfbc99 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -126,13 +126,13 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { struct regmap *regmap; + struct mutex ctrl_lock; /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; /* ANC */ - struct mutex anc_lock; long *anc_fir_values; long *anc_iir_values; enum anc_state anc_status; @@ -1129,9 +1129,9 @@ static int sid_status_control_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); - mutex_lock(&codec->mutex); + mutex_lock(&drvdata->ctrl_lock); ucontrol->value.integer.value[0] = drvdata->sid_status; - mutex_unlock(&codec->mutex); + mutex_unlock(&drvdata->ctrl_lock); return 0; } @@ -1154,7 +1154,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol, return -EIO; } - mutex_lock(&codec->mutex); + mutex_lock(&drvdata->ctrl_lock); sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF); if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) { @@ -1185,7 +1185,7 @@ static int sid_status_control_put(struct snd_kcontrol *kcontrol, drvdata->sid_status = SID_FIR_CONFIGURED; out: - mutex_unlock(&codec->mutex); + mutex_unlock(&drvdata->ctrl_lock); dev_dbg(codec->dev, "%s: Exit\n", __func__); @@ -1198,9 +1198,9 @@ static int anc_status_control_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); - mutex_lock(&codec->mutex); + mutex_lock(&drvdata->ctrl_lock); ucontrol->value.integer.value[0] = drvdata->anc_status; - mutex_unlock(&codec->mutex); + mutex_unlock(&drvdata->ctrl_lock); return 0; } @@ -1217,7 +1217,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, dev_dbg(dev, "%s: Enter.\n", __func__); - mutex_lock(&drvdata->anc_lock); + mutex_lock(&drvdata->ctrl_lock); req = ucontrol->value.integer.value[0]; if (req >= ARRAY_SIZE(enum_anc_state)) { @@ -1244,9 +1244,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, } snd_soc_dapm_sync(&codec->dapm); - mutex_lock(&codec->mutex); anc_configure(codec, apply_fir, apply_iir); - mutex_unlock(&codec->mutex); if (apply_fir) { if (drvdata->anc_status == ANC_IIR_CONFIGURED) @@ -1265,7 +1263,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(&codec->dapm); cleanup: - mutex_unlock(&drvdata->anc_lock); + mutex_unlock(&drvdata->ctrl_lock); if (status < 0) dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n", @@ -1294,14 +1292,15 @@ static int filter_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct ab8500_codec_drvdata *drvdata = snd_soc_codec_get_drvdata(codec); struct filter_control *fc = (struct filter_control *)kcontrol->private_value; unsigned int i; - mutex_lock(&codec->mutex); + mutex_lock(&drvdata->ctrl_lock); for (i = 0; i < fc->count; i++) ucontrol->value.integer.value[i] = fc->value[i]; - mutex_unlock(&codec->mutex); + mutex_unlock(&drvdata->ctrl_lock); return 0; } @@ -1310,14 +1309,15 @@ static int filter_control_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct ab8500_codec_drvdata *drvdata = snd_soc_codec_get_drvdata(codec); struct filter_control *fc = (struct filter_control *)kcontrol->private_value; unsigned int i; - mutex_lock(&codec->mutex); + mutex_lock(&drvdata->ctrl_lock); for (i = 0; i < fc->count; i++) fc->value[i] = ucontrol->value.integer.value[i]; - mutex_unlock(&codec->mutex); + mutex_unlock(&drvdata->ctrl_lock); return 0; } @@ -2545,7 +2545,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); - mutex_init(&drvdata->anc_lock); + mutex_init(&drvdata->ctrl_lock); return status; } -- cgit v1.1 From 210a5fae55c05174b8a5b571b6698626b3ae35d3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:00:58 +0100 Subject: ASoC: max98095: Move mutex to the driver level The max98095 uses the snd_soc_codec mutex to protect against concurrent access in some of its control put handlers. Move this mutex to the driver level so we can eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 0ee6797..01f3cc9 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -57,6 +58,7 @@ struct max98095_priv { unsigned int mic2pre; struct snd_soc_jack *headphone_jack; struct snd_soc_jack *mic_jack; + struct mutex lock; }; static const struct reg_default max98095_reg_def[] = { @@ -1803,7 +1805,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL); snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0); - mutex_lock(&codec->mutex); + mutex_lock(&max98095->lock); snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG); m98095_eq_band(codec, channel, 0, coef_set->band1); m98095_eq_band(codec, channel, 1, coef_set->band2); @@ -1811,7 +1813,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, m98095_eq_band(codec, channel, 3, coef_set->band4); m98095_eq_band(codec, channel, 4, coef_set->band5); snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0); - mutex_unlock(&codec->mutex); + mutex_unlock(&max98095->lock); /* Restore the original on/off state */ snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave); @@ -1957,12 +1959,12 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL); snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0); - mutex_lock(&codec->mutex); + mutex_lock(&max98095->lock); snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG); m98095_biquad_band(codec, channel, 0, coef_set->band1); m98095_biquad_band(codec, channel, 1, coef_set->band2); snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0); - mutex_unlock(&codec->mutex); + mutex_unlock(&max98095->lock); /* Restore the original on/off state */ snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave); @@ -2395,6 +2397,8 @@ static int max98095_i2c_probe(struct i2c_client *i2c, if (max98095 == NULL) return -ENOMEM; + mutex_init(&max98095->lock); + max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap); if (IS_ERR(max98095->regmap)) { ret = PTR_ERR(max98095->regmap); -- cgit v1.1 From d74bcaaeb66826192c9e361cbfe8fd1ffaccf74e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:00:59 +0100 Subject: ASoC: wm5102: Move ultrasonic response settings lock to the driver level The wm5102 driver currently uses the snd_soc_codec mutex to protect its ultrasonic response settings from concurrent access. This patch moves this lock to the driver level. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 4 ++-- sound/soc/codecs/wm5102.c | 16 ++++++++-------- 3 files changed, 11 insertions(+), 10 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index f34723f..910e3aa 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -141,6 +141,7 @@ struct arizona { uint16_t dac_comp_coeff; uint8_t dac_comp_enabled; + struct mutex dac_comp_lock; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c05e7a..730636c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1164,13 +1164,13 @@ static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, { 0x80, 0x0 }, }; - mutex_lock(&codec->mutex); + mutex_lock(&arizona->dac_comp_lock); dac_comp[1].def = arizona->dac_comp_coeff; if (rate >= 176400) dac_comp[2].def = arizona->dac_comp_enabled; - mutex_unlock(&codec->mutex); + mutex_unlock(&arizona->dac_comp_lock); regmap_multi_reg_write(arizona->regmap, dac_comp, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..1f75534 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -619,10 +619,10 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, struct arizona *arizona = dev_get_drvdata(codec->dev->parent); uint16_t data; - mutex_lock(&codec->mutex); + mutex_lock(&arizona->dac_comp_lock); data = cpu_to_be16(arizona->dac_comp_coeff); memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); - mutex_unlock(&codec->mutex); + mutex_unlock(&arizona->dac_comp_lock); return 0; } @@ -633,11 +633,11 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - mutex_lock(&codec->mutex); + mutex_lock(&arizona->dac_comp_lock); memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, sizeof(arizona->dac_comp_coeff)); arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); - mutex_unlock(&codec->mutex); + mutex_unlock(&arizona->dac_comp_lock); return 0; } @@ -648,9 +648,9 @@ static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - mutex_lock(&codec->mutex); + mutex_lock(&arizona->dac_comp_lock); ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; - mutex_unlock(&codec->mutex); + mutex_unlock(&arizona->dac_comp_lock); return 0; } @@ -661,9 +661,9 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - mutex_lock(&codec->mutex); + mutex_lock(&arizona->dac_comp_lock); arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; - mutex_unlock(&codec->mutex); + mutex_unlock(&arizona->dac_comp_lock); return 0; } -- cgit v1.1 From a51ff30f45473a80f78b2572666473887e010d91 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:01:00 +0100 Subject: ASoC: wm8731: Move the deemph lock to the driver level The wm8731 uses the snd_soc_codec mutex to protect its deemph settings from concurrent access. This patch moves this lock to the driver level. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index eebb328..5dae9a6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -50,6 +51,8 @@ struct wm8731_priv { int sysclk_type; int playback_fs; bool deemph; + + struct mutex lock; }; @@ -138,7 +141,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; - mutex_lock(&codec->mutex); + mutex_lock(&wm8731->lock); if (wm8731->deemph != deemph) { wm8731->deemph = deemph; @@ -146,7 +149,7 @@ static int wm8731_put_deemph(struct snd_kcontrol *kcontrol, ret = 1; } - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8731->lock); return ret; } @@ -685,6 +688,8 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- cgit v1.1 From 78660af7ba30e9d2cc9614465c8b65b3c85f49a9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:01:01 +0100 Subject: ASoC: wm8903: Move the deemph lock to the driver level The wm8903 uses the snd_soc_codec mutex to protect its deemph settings from concurrent access. This patch moves this lock to the driver level. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c038b3e..ffbe6df 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -123,6 +124,7 @@ struct wm8903_priv { int sysclk; int irq; + struct mutex lock; int fs; int deemph; @@ -457,7 +459,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; - mutex_lock(&codec->mutex); + mutex_lock(&wm8903->lock); if (wm8903->deemph != deemph) { wm8903->deemph = deemph; @@ -465,7 +467,7 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, ret = 1; } - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8903->lock); return ret; } @@ -2023,6 +2025,8 @@ static int wm8903_i2c_probe(struct i2c_client *i2c, GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; + + mutex_init(&wm8903->lock); wm8903->dev = &i2c->dev; wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap); -- cgit v1.1 From fabfad2f8b23529722c6ef5b3537c457e63d2c82 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:01:02 +0100 Subject: ASoC: wm8958: Move DSP firmware lock to driver level The wm8958 driver uses the snd_soc_codec mutex to protect the various firmware pointers from concurrent assignment. This patch moves this lock to the driver level. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 12 ++++++------ sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm8994.h | 2 ++ 3 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 0dada7f..3cbc82b 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -867,9 +867,9 @@ static void wm8958_enh_eq_loaded(const struct firmware *fw, void *context) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); if (fw && (wm8958_dsp2_fw(codec, "ENH_EQ", fw, true) == 0)) { - mutex_lock(&codec->mutex); + mutex_lock(&wm8994->fw_lock); wm8994->enh_eq = fw; - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8994->fw_lock); } } @@ -879,9 +879,9 @@ static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); if (fw && (wm8958_dsp2_fw(codec, "MBC+VSS", fw, true) == 0)) { - mutex_lock(&codec->mutex); + mutex_lock(&wm8994->fw_lock); wm8994->mbc_vss = fw; - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8994->fw_lock); } } @@ -891,9 +891,9 @@ static void wm8958_mbc_loaded(const struct firmware *fw, void *context) struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); if (fw && (wm8958_dsp2_fw(codec, "MBC", fw, true) == 0)) { - mutex_lock(&codec->mutex); + mutex_lock(&wm8994->fw_lock); wm8994->mbc = fw; - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8994->fw_lock); } } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1fcb9f3..dbca6e0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4457,6 +4457,8 @@ static int wm8994_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm8994); + mutex_init(&wm8994->fw_lock); + wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 6536f8d..dd73387 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -13,6 +13,7 @@ #include #include #include +#include #include "wm_hubs.h" @@ -156,6 +157,7 @@ struct wm8994_priv { unsigned int aif1clk_disable:1; unsigned int aif2clk_disable:1; + struct mutex fw_lock; int dsp_active; const struct firmware *cur_fw; const struct firmware *mbc; -- cgit v1.1 From 3e4199ef0105fb718b24cbcc837ad527fd60c880 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:01:03 +0100 Subject: ASoC: wm8962: Move DSP enable lock to the driver level The wm8962 uses the snd_soc_codec mutex to protect the wm8962_dsp2_ena_put() function from concurrent execution. This patch moves that lock to the driver level. This will allow us to eventually remove the snd_soc_codec mutex. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 9077411..61ca4a7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -67,6 +68,7 @@ struct wm8962_priv { int fll_fref; int fll_fout; + struct mutex dsp2_ena_lock; u16 dsp2_ena; struct delayed_work mic_work; @@ -1570,7 +1572,7 @@ static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) & WM8962_DSP2_ENA; - mutex_lock(&codec->mutex); + mutex_lock(&wm8962->dsp2_ena_lock); if (ucontrol->value.integer.value[0]) wm8962->dsp2_ena |= 1 << shift; @@ -1590,7 +1592,7 @@ static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol, } out: - mutex_unlock(&codec->mutex); + mutex_unlock(&wm8962->dsp2_ena_lock); return ret; } @@ -3557,6 +3559,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, if (wm8962 == NULL) return -ENOMEM; + mutex_init(&wm8962->dsp2_ena_lock); + i2c_set_clientdata(i2c, wm8962); INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); -- cgit v1.1 From bd6b87c104bae49816808fde5f55a262093e85ed Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 9 Nov 2014 17:01:04 +0100 Subject: ASoC: Remove CODEC mutex The CODEC mutex is now unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - sound/soc/soc-core.c | 1 - 2 files changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..5c91b06 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -780,7 +780,6 @@ struct snd_soc_codec { struct device *dev; const struct snd_soc_codec_driver *driver; - struct mutex mutex; struct list_head list; struct list_head card_list; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..cc7bb7a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4362,7 +4362,6 @@ int snd_soc_register_codec(struct device *dev, codec->dev = dev; codec->driver = codec_drv; codec->component.val_bytes = codec_drv->reg_word_size; - mutex_init(&codec->mutex); #ifdef CONFIG_DEBUG_FS codec->component.init_debugfs = soc_init_codec_debugfs; -- cgit v1.1 From 8c2727f97b4825adaad43bf98632abc9940345a4 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Nov 2014 17:49:30 -0200 Subject: ASoC: mxs: mxs-saif: Register the irq with the device name Instead of registering the irq name with the driver name, it's better to pass the device name so that we have a more explicit indication as to what saif instance the irq is related: $ cat /proc/interrupts CPU0 ... 214: 4 - 59 80042000.saif 215: 0 - 58 80046000.saif Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 231d7e7..83b2fea 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -773,7 +773,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->dev = &pdev->dev; ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, - "mxs-saif", saif); + dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); return ret; -- cgit v1.1 From 2a9e8df00951092e825144a9968285398f8aa162 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 16:46:35 +0100 Subject: ALSA: vx: Fix missing kerneldoc parameter descriptions The file isn't processed, but it's not bad to fix beforehand. Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_core.c | 10 ++++++++++ sound/pci/vx222/vx222_ops.c | 5 +++++ sound/pcmcia/vx/vxpocket.c | 1 + 3 files changed, 16 insertions(+) diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index e8cc169..fc05a37 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -416,6 +416,7 @@ int vx_send_rih(struct vx_core *chip, int cmd) /** * snd_vx_boot_xilinx - boot up the xilinx interface + * @chip: VX core instance * @boot: the boot record to load */ int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot) @@ -538,6 +539,8 @@ EXPORT_SYMBOL(snd_vx_threaded_irq_handler); /** * snd_vx_irq_handler - interrupt handler + * @irq: irq number + * @dev: VX core instance */ irqreturn_t snd_vx_irq_handler(int irq, void *dev) { @@ -649,6 +652,8 @@ static void vx_proc_init(struct vx_core *chip) /** * snd_vx_dsp_boot - load the DSP boot + * @chip: VX core instance + * @boot: firmware data */ int snd_vx_dsp_boot(struct vx_core *chip, const struct firmware *boot) { @@ -669,6 +674,8 @@ EXPORT_SYMBOL(snd_vx_dsp_boot); /** * snd_vx_dsp_load - load the DSP image + * @chip: VX core instance + * @dsp: firmware data */ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) { @@ -768,7 +775,10 @@ EXPORT_SYMBOL(snd_vx_resume); /** * snd_vx_create - constructor for struct vx_core + * @card: card instance * @hw: hardware specific record + * @ops: VX ops pointer + * @extra_size: extra byte size to allocate appending to chip * * this function allocates the instance and prepare for the hardware * initialization. diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 2d15702..52c1a8d 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -92,6 +92,7 @@ static inline unsigned long vx2_reg_addr(struct vx_core *_chip, int reg) /** * snd_vx_inb - read a byte from the register + * @chip: VX core instance * @offset: register enum */ static unsigned char vx2_inb(struct vx_core *chip, int offset) @@ -101,6 +102,7 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset) /** * snd_vx_outb - write a byte on the register + * @chip: VX core instance * @offset: the register offset * @val: the value to write */ @@ -114,6 +116,7 @@ static void vx2_outb(struct vx_core *chip, int offset, unsigned char val) /** * snd_vx_inl - read a 32bit word from the register + * @chip: VX core instance * @offset: register enum */ static unsigned int vx2_inl(struct vx_core *chip, int offset) @@ -123,6 +126,7 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset) /** * snd_vx_outl - write a 32bit word on the register + * @chip: VX core instance * @offset: the register enum * @val: the value to write */ @@ -223,6 +227,7 @@ static int vx2_test_xilinx(struct vx_core *_chip) /** * vx_setup_pseudo_dma - set up the pseudo dma read/write mode. + * @chip: VX core instance * @do_write: 0 = read, 1 = set up for DMA write */ static void vx2_setup_pseudo_dma(struct vx_core *chip, int do_write) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 92ec114..b16f42d 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -174,6 +174,7 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl, /** * snd_vxpocket_assign_resources - initialize the hardware and card instance. + * @chip: VX core instance * @port: i/o port for the card * @irq: irq number for the card * -- cgit v1.1 From e60b2c7fcdef03256cde864d678df240877a5e80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 16:47:26 +0100 Subject: ALSA: hda - Fix kerneldoc errors in patch_ca0132.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4f7ffa8..e0383ee 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2417,7 +2417,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, * @reloc: Relocation address for loading single-segment overlays, or 0 for * no relocation * @sample_rate: sampling rate of the stream used for DSP download - * @number_channels: channels of the stream used for DSP download + * @channels: channels of the stream used for DSP download * @ovly: TRUE if overlay format is required * * Returns zero or a negative error code. @@ -2556,10 +2556,7 @@ static void dspload_post_setup(struct hda_codec *codec) } /** - * Download DSP from a DSP Image Fast Load structure. This structure is a - * linear, non-constant sized element array of structures, each of which - * contain the count of the data to be loaded, the data itself, and the - * corresponding starting chip address of the starting data location. + * dspload_image - Download DSP from a DSP Image Fast Load structure. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -2570,6 +2567,10 @@ static void dspload_post_setup(struct hda_codec *codec) * @router_chans: number of audio router channels to be allocated (0 means use * internal defaults; max is 32) * + * Download DSP from a DSP Image Fast Load structure. This structure is a + * linear, non-constant sized element array of structures, each of which + * contain the count of the data to be loaded, the data itself, and the + * corresponding starting chip address of the starting data location. * Returns zero or a negative error code. */ static int dspload_image(struct hda_codec *codec, -- cgit v1.1 From 3f60c87d129acbb232afbc7269c726d009a01869 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 17:09:22 +0100 Subject: ALSA: mixart: Fix kerneldoc comments Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart_hwdep.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 581e1e7..9996a4d 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -37,10 +37,11 @@ /** * wait for a value on a peudo register, exit with a timeout * - * @param mgr pointer to miXart manager structure - * @param offset unsigned pseudo_register base + offset of value - * @param value value - * @param timeout timeout in centisenconds + * @mgr: pointer to miXart manager structure + * @offset: unsigned pseudo_register base + offset of value + * @is_egal: wait for the equal value + * @value: value + * @timeout: timeout in centisenconds */ static int mixart_wait_nice_for_register_value(struct mixart_mgr *mgr, u32 offset, int is_egal, -- cgit v1.1 From ddcecf6b6ae7b91c8735e52f50cd403ee9cbe298 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Nov 2014 17:24:26 +0100 Subject: ALSA: Fix invalid kerneldoc markers They are no real kerneldoc comments, so drop such markers. Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 12 +++++------ sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/lx6464es/lx_defs.h | 2 +- sound/pci/rme9652/hdspm.c | 48 ++++++++++++++++++++++---------------------- 4 files changed, 32 insertions(+), 32 deletions(-) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index d956c35..b357f1a5 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -74,14 +74,14 @@ struct hdspm_config { #define SNDRV_HDSPM_IOCTL_GET_CONFIG \ _IOR('H', 0x41, struct hdspm_config) -/** +/* * If there's a TCO (TimeCode Option) board installed, * there are further options and status data available. * The hdspm_ltc structure contains the current SMPTE * timecode and some status information and can be * obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the * hdspm_status struct. - **/ + */ enum hdspm_ltc_format { format_invalid, @@ -113,11 +113,11 @@ struct hdspm_ltc { #define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc) -/** +/* * The status data reflects the device's current state * as determined by the card's configuration and * connection status. - **/ + */ enum hdspm_sync { hdspm_sync_no_lock = 0, @@ -171,9 +171,9 @@ struct hdspm_status { #define SNDRV_HDSPM_IOCTL_GET_STATUS \ _IOR('H', 0x47, struct hdspm_status) -/** +/* * Get information about the card and its add-ons. - **/ + */ #define HDSPM_ADDON_TCO 1 diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index e223de1..15933f9 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -180,7 +180,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard."); /* From 0x50 - 0x5f, last samples captured */ -/** +/* * The hardware has 3 channels for playback and 1 for capture. * - channel 0 is the front channel * - channel 1 is the rear channel diff --git a/sound/pci/lx6464es/lx_defs.h b/sound/pci/lx6464es/lx_defs.h index 49d36bd..469bcc6 100644 --- a/sound/pci/lx6464es/lx_defs.h +++ b/sound/pci/lx6464es/lx_defs.h @@ -175,7 +175,7 @@ enum buffer_flags { BF_ZERO = 0x00, /* no flags (init).*/ }; -/** +/* * Stream Flags definitions */ enum stream_flags { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e09348c1..3342705 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2201,10 +2201,10 @@ static inline int hdspm_get_pll_freq(struct hdspm *hdspm) return rate; } -/** +/* * Calculate the real sample rate from the * current DDS value. - **/ + */ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) { unsigned int rate; @@ -2270,9 +2270,9 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, } -/** +/* * Returns the WordClock sample rate class for the given card. - **/ + */ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) { int status; @@ -2295,9 +2295,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) } -/** +/* * Returns the TCO sample rate class for the given card. - **/ + */ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) { int status; @@ -2321,9 +2321,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) } -/** +/* * Returns the SYNC_IN sample rate class for the given card. - **/ + */ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) { int status; @@ -2343,9 +2343,9 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) return 0; } -/** +/* * Returns the AES sample rate class for the given card. - **/ + */ static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) { int timecode; @@ -2361,10 +2361,10 @@ static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) return 0; } -/** +/* * Returns the sample rate class for input source for * 'new style' cards like the AIO and RayDAT. - **/ + */ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) { int status = hdspm_read(hdspm, HDSPM_RD_STATUS_2); @@ -2512,10 +2512,10 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, } -/** +/* * Returns the system clock mode for the given card. * @returns 0 - master, 1 - slave - **/ + */ static int hdspm_system_clock_mode(struct hdspm *hdspm) { switch (hdspm->io_type) { @@ -2534,10 +2534,10 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm) } -/** +/* * Sets the system clock mode. * @param mode 0 - master, 1 - slave - **/ + */ static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode) { hdspm_set_toggle_setting(hdspm, @@ -2692,11 +2692,11 @@ static int snd_hdspm_put_clock_source(struct snd_kcontrol *kcontrol, } -/** +/* * Returns the current preferred sync reference setting. * The semantics of the return value are depending on the * card, please see the comments for clarification. - **/ + */ static int hdspm_pref_sync_ref(struct hdspm * hdspm) { switch (hdspm->io_type) { @@ -2795,11 +2795,11 @@ static int hdspm_pref_sync_ref(struct hdspm * hdspm) } -/** +/* * Set the preferred sync reference to . The semantics * of are depending on the card type, see the comments * for clarification. - **/ + */ static int hdspm_set_pref_sync_ref(struct hdspm * hdspm, int pref) { int p = 0; @@ -4101,9 +4101,9 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, -/** +/* * TCO controls - **/ + */ static void hdspm_tco_write(struct hdspm *hdspm) { unsigned int tc[4] = { 0, 0, 0, 0}; @@ -5403,7 +5403,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) HDSPM_midi2IRQPending | HDSPM_midi3IRQPending); /* now = get_cycles(); */ - /** + /* * LAT_2..LAT_0 period counter (win) counter (mac) * 6 4096 ~256053425 ~514672358 * 5 2048 ~128024983 ~257373821 @@ -5412,7 +5412,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) * 2 256 ~16003039 ~32260176 * 1 128 ~7998738 ~16194507 * 0 64 ~3998231 ~8191558 - **/ + */ /* dev_info(hdspm->card->dev, "snd_hdspm_interrupt %llu @ %llx\n", now-hdspm->last_interrupt, status & 0xFFC0); -- cgit v1.1 From 0cf1863219b474e82af9cb1f6715a0bbfa3fdf1a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 11 Nov 2014 17:59:50 +0800 Subject: ASoC: rt5670: add rt5672 codec support rt5672 is very similar to rt5670. Therefore we use one codec driver to support both codecs. The difference between rt5670 and rt5672 is there is some difference in their dapm routing table. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 71 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/rt5670.h | 6 ++++ 2 files changed, 65 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index ba9d9b4..066b583 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1595,29 +1595,40 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { /* PDM */ SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, RT5670_PWR_PDM1_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, - RT5670_PWR_PDM2_BIT, 0, NULL, 0), SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), - SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), - SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), /* Output Lines */ SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), SND_SOC_DAPM_OUTPUT("PDM1L"), SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("PDM2L"), SND_SOC_DAPM_OUTPUT("PDM2R"), }; +static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { + SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), +}; + static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1970,12 +1981,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, { "PDM1 R Mux", NULL, "PDM1 Power" }, - { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, - { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, - { "PDM2 L Mux", NULL, "PDM2 Power" }, - { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, - { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, - { "PDM2 R Mux", NULL, "PDM2 Power" }, { "HP Amp", NULL, "HPO MIX" }, { "HP Amp", NULL, "Mic Det Power" }, @@ -1993,13 +1998,30 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "LOUTR", NULL, "LOUT R Playback" }, { "LOUTL", NULL, "Improve HP Amp Drv" }, { "LOUTR", NULL, "Improve HP Amp Drv" }, +}; +static const struct snd_soc_dapm_route rt5670_specific_dapm_routes[] = { + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, { "PDM1L", NULL, "PDM1 L Mux" }, { "PDM1R", NULL, "PDM1 R Mux" }, { "PDM2L", NULL, "PDM2 L Mux" }, { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5672_specific_dapm_routes[] = { + { "SPO Amp", NULL, "PDM1 L Mux" }, + { "SPO Amp", NULL, "PDM1 R Mux" }, + { "SPOLP", NULL, "SPO Amp" }, + { "SPOLN", NULL, "SPO Amp" }, + { "SPORP", NULL, "SPO Amp" }, + { "SPORN", NULL, "SPO Amp" }, +}; + static int rt5670_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2331,6 +2353,29 @@ static int rt5670_probe(struct snd_soc_codec *codec) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) { + case RT5670_ID_5670: + case RT5670_ID_5671: + snd_soc_dapm_new_controls(&codec->dapm, + rt5670_specific_dapm_widgets, + ARRAY_SIZE(rt5670_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5670_specific_dapm_routes, + ARRAY_SIZE(rt5670_specific_dapm_routes)); + break; + case RT5670_ID_5672: + snd_soc_dapm_new_controls(&codec->dapm, + rt5672_specific_dapm_widgets, + ARRAY_SIZE(rt5672_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5672_specific_dapm_routes, + ARRAY_SIZE(rt5672_specific_dapm_routes)); + break; + default: + dev_err(codec->dev, + "The driver is for RT5670 RT5671 or RT5672 only\n"); + return -ENODEV; + } rt5670->codec = codec; return 0; @@ -2452,6 +2497,8 @@ static const struct regmap_config rt5670_regmap = { static const struct i2c_device_id rt5670_i2c_id[] = { { "rt5670", 0 }, + { "rt5671", 0 }, + { "rt5672", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a0b5c85..d11b9c2 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -228,6 +228,12 @@ #define RT5670_R_VOL_MASK (0x3f) #define RT5670_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5670_ID_MASK (0x3 << 1) +#define RT5670_ID_5670 (0x0 << 1) +#define RT5670_ID_5672 (0x1 << 1) +#define RT5670_ID_5671 (0x2 << 1) + /* Combo Jack Control 1 (0x0a) */ #define RT5670_CBJ_BST1_MASK (0xf << 12) #define RT5670_CBJ_BST1_SFT (12) -- cgit v1.1 From 5563502cb68d9520e13fe2350922ca88c4531c63 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 11 Nov 2014 11:31:28 +0800 Subject: ASoC: rt5645: remove unused rt5645_clk_sel_put Remove rt5645_clk_sel_put function since it is never used. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 59 ----------------------------------------------- 1 file changed, 59 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1dbbebc..665f8b6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -441,65 +441,6 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); -static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - unsigned int u_bit = 0, p_bit = 0; - struct soc_enum *em = - (struct soc_enum *)kcontrol->private_value; - - switch (em->reg) { - case RT5645_ASRC_2: - switch (em->shift_l) { - case 0: - u_bit = 0x8; - p_bit = RT5645_PWR_ADC_S1F; - break; - case 4: - u_bit = 0x100; - p_bit = RT5645_PWR_DAC_MF_R; - break; - case 8: - u_bit = 0x200; - p_bit = RT5645_PWR_DAC_MF_L; - break; - case 12: - u_bit = 0x400; - p_bit = RT5645_PWR_DAC_S1F; - break; - } - break; - case RT5645_ASRC_3: - switch (em->shift_l) { - case 0: - u_bit = 0x1; - p_bit = RT5645_PWR_ADC_MF_R; - break; - case 4: - u_bit = 0x2; - p_bit = RT5645_PWR_ADC_MF_L; - break; - } - break; - } - - if (u_bit || p_bit) { - switch (ucontrol->value.integer.value[0]) { - case 1 ... 4: /*enable*/ - if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) - snd_soc_update_bits(codec, - RT5645_ASRC_1, u_bit, u_bit); - break; - default: /*disable*/ - snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); - break; - } - } - - return snd_soc_put_enum_double(kcontrol, ucontrol); -} - static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, -- cgit v1.1 From a60e654be733a69879148cb4c56d0f58b749e3c4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 11 Nov 2014 10:59:00 +0200 Subject: ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual It is easier to find the relevant enums in the code. Use the SOC_ENUM_*_DECL macro for the individual items. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 151 +++++++++++++++++++++-------------------- 1 file changed, 79 insertions(+), 72 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8770e28..9901400 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -216,61 +216,68 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; -static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; -static const char *aic3x_left_hpcom_mux[] = - { "differential of HPLOUT", "constant VCM", "single-ended" }; -static const char *aic3x_right_hpcom_mux[] = - { "differential of HPROUT", "constant VCM", "single-ended", - "differential of HPLCOM", "external feedback" }; -static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; -static const char *aic3x_adc_hpf[] = - { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; - -#define LDAC_ENUM 0 -#define RDAC_ENUM 1 -#define LHPCOM_ENUM 2 -#define RHPCOM_ENUM 3 -#define LINE1L_2_L_ENUM 4 -#define LINE1L_2_R_ENUM 5 -#define LINE1R_2_L_ENUM 6 -#define LINE1R_2_R_ENUM 7 -#define LINE2L_ENUM 8 -#define LINE2R_ENUM 9 -#define ADC_HPF_ENUM 10 - -static const struct soc_enum aic3x_enum[] = { - SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), - SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux), - SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux), - SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux), - SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), - SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), -}; - -static const char *aic3x_agc_level[] = - { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; -static const struct soc_enum aic3x_agc_level_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), -}; - -static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; -static const struct soc_enum aic3x_agc_attack_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), -}; - -static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; -static const struct soc_enum aic3x_agc_decay_enum[] = { - SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), - SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), -}; +static const char * const aic3x_left_dac_mux[] = { + "DAC_L1", "DAC_L3", "DAC_L2" }; +static SOC_ENUM_SINGLE_DECL(aic3x_left_dac_enum, DAC_LINE_MUX, 6, + aic3x_left_dac_mux); + +static const char * const aic3x_right_dac_mux[] = { + "DAC_R1", "DAC_R3", "DAC_R2" }; +static SOC_ENUM_SINGLE_DECL(aic3x_right_dac_enum, DAC_LINE_MUX, 4, + aic3x_right_dac_mux); + +static const char * const aic3x_left_hpcom_mux[] = { + "differential of HPLOUT", "constant VCM", "single-ended" }; +static SOC_ENUM_SINGLE_DECL(aic3x_left_hpcom_enum, HPLCOM_CFG, 4, + aic3x_left_hpcom_mux); + +static const char * const aic3x_right_hpcom_mux[] = { + "differential of HPROUT", "constant VCM", "single-ended", + "differential of HPLCOM", "external feedback" }; +static SOC_ENUM_SINGLE_DECL(aic3x_right_hpcom_enum, HPRCOM_CFG, 3, + aic3x_right_hpcom_mux); + +static const char * const aic3x_linein_mode_mux[] = { + "single-ended", "differential" }; +static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_l_enum, LINE1L_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_r_enum, LINE1L_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_l_enum, LINE1R_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_r_enum, LINE1R_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line2l_2_ldac_enum, LINE2L_2_LADC_CTRL, 7, + aic3x_linein_mode_mux); +static SOC_ENUM_SINGLE_DECL(aic3x_line2r_2_rdac_enum, LINE2R_2_RADC_CTRL, 7, + aic3x_linein_mode_mux); + +static const char * const aic3x_adc_hpf[] = { + "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static SOC_ENUM_DOUBLE_DECL(aic3x_adc_hpf_enum, AIC3X_CODEC_DFILT_CTRL, 6, 4, + aic3x_adc_hpf); + +static const char * const aic3x_agc_level[] = { + "-5.5dB", "-8dB", "-10dB", "-12dB", + "-14dB", "-17dB", "-20dB", "-24dB" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_level_enum, LAGC_CTRL_A, 4, + aic3x_agc_level); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_level_enum, RAGC_CTRL_A, 4, + aic3x_agc_level); + +static const char * const aic3x_agc_attack[] = { + "8ms", "11ms", "16ms", "20ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_attack_enum, LAGC_CTRL_A, 2, + aic3x_agc_attack); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_attack_enum, RAGC_CTRL_A, 2, + aic3x_agc_attack); + +static const char * const aic3x_agc_decay[] = { + "100ms", "200ms", "400ms", "500ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0, + aic3x_agc_decay); +static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0, + aic3x_agc_decay); /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps @@ -385,12 +392,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), - SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), - SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), - SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), - SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), - SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), - SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), + SOC_ENUM("Left AGC Target level", aic3x_lagc_level_enum), + SOC_ENUM("Right AGC Target level", aic3x_ragc_level_enum), + SOC_ENUM("Left AGC Attack time", aic3x_lagc_attack_enum), + SOC_ENUM("Right AGC Attack time", aic3x_ragc_attack_enum), + SOC_ENUM("Left AGC Decay time", aic3x_lagc_decay_enum), + SOC_ENUM("Right AGC Decay time", aic3x_ragc_decay_enum), /* De-emphasis */ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), @@ -400,7 +407,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { 0, 119, 0, adc_tlv), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), - SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { @@ -427,19 +434,19 @@ static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_left_dac_enum); /* Right DAC Mux */ static const struct snd_kcontrol_new aic3x_right_dac_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_right_dac_enum); /* Left HPCOM Mux */ static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_left_hpcom_enum); /* Right HPCOM Mux */ static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { @@ -531,23 +538,23 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1l_2_r_enum); /* Right Line1 Mux */ static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1r_2_r_enum); static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line1r_2_l_enum); /* Left Line2 Mux */ static const struct snd_kcontrol_new aic3x_left_line2_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line2l_2_ldac_enum); /* Right Line2 Mux */ static const struct snd_kcontrol_new aic3x_right_line2_mux_controls = -SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); +SOC_DAPM_ENUM("Route", aic3x_line2r_2_rdac_enum); static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Left DAC to Left Outputs */ -- cgit v1.1 From 68d6626925c3529790a2055d41578415fa98495e Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Tue, 11 Nov 2014 10:59:01 +0200 Subject: ASoC: tlv320aic3x: Add output driver pop reduction controls Output driver has two parameters that can be configured to reduce pop noise: power-on delay and ramp-up step time. Two new kcontrols have been added to set these parameters. Signed-off-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 9901400..f0a8281 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -279,6 +279,16 @@ static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0, static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0, aic3x_agc_decay); +static const char * const aic3x_poweron_time[] = { + "0us", "10us", "100us", "1ms", "10ms", "50ms", + "100ms", "200ms", "400ms", "800ms", "2s", "4s" }; +static SOC_ENUM_SINGLE_DECL(aic3x_poweron_time_enum, HPOUT_POP_REDUCTION, 4, + aic3x_poweron_time); + +static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" }; +static SOC_ENUM_SINGLE_DECL(aic3x_rampup_step_enum, HPOUT_POP_REDUCTION, 2, + aic3x_rampup_step); + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -408,6 +418,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_adc_hpf_enum), + + /* Pop reduction */ + SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), + SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { -- cgit v1.1 From d497a82fb18ed4b73c08f8b5a0935f937e2ea1fb Mon Sep 17 00:00:00 2001 From: Damien Zammit Date: Wed, 12 Nov 2014 01:09:54 +1100 Subject: ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source This patch provides the infrastructure for the Digidesign Mbox 1 to have a mixer control for selecting the clock source. Valid options are Internal and S/PDIF external sync. A non-documented command is sent to the device to enable this feature found by reverse engineering and bus snooping. Signed-off-by: Damien Zammit Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 138 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index d1d72ff..1994d41 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -179,6 +179,11 @@ static struct usbmix_name_map audigy2nx_map[] = { { 0 } /* terminator */ }; +static struct usbmix_name_map mbox1_map[] = { + { 1, "Clock" }, + { 0 } /* terminator */ +}; + static struct usbmix_selector_map c400_selectors[] = { { .id = 0x80, @@ -416,6 +421,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = aureon_51_2_map, }, { + .id = USB_ID(0x0dba, 0x1000), + .map = mbox1_map, + }, + { .id = USB_ID(0x13e5, 0x0001), .map = scratch_live_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 3980bf5..4520316 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -565,6 +565,131 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* Digidesign Mbox 1 clock source switch (internal/spdif) */ + +static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kctl->private_value; + return 0; +} + +static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_usb_audio *chip; + struct usb_mixer_interface *mixer; + int err; + bool cur_val, new_val; + unsigned char buff[3]; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + + mixer = snd_kcontrol_chip(kctl); + if (snd_BUG_ON(!mixer)) + return -EINVAL; + + chip = mixer->chip; + if (snd_BUG_ON(!chip)) + return -EINVAL; + + if (cur_val == new_val) + return 0; + + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { + err = -ENODEV; + goto err; + } + + /* Prepare for magic command to toggle clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); + if (err < 0) + goto err; + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + goto err; + + /* 2 possibilities: Internal -> send sample rate + * S/PDIF sync -> send zeroes + * NB: Sample rate locked to 48kHz on purpose to + * prevent user from resetting the sample rate + * while S/PDIF sync is enabled and confusing + * this configuration. + */ + if (new_val == 0) { + buff[0] = 0x80; + buff[1] = 0xbb; + buff[2] = 0x00; + } else { + buff[0] = buff[1] = buff[2] = 0x00; + } + + /* Send the magic command to toggle the clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + goto err; + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + goto err; + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x2, buff, 3); + if (err < 0) + goto err; + kctl->private_value = new_val; + +err: + up_read(&chip->shutdown_rwsem); + return err < 0 ? err : 1; +} + +static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Internal", + "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static struct snd_kcontrol_new snd_mbox1_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Clock Source", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_mbox1_switch_info, + .get = snd_mbox1_switch_get, + .put = snd_mbox1_switch_put, + .private_value = 0 +}; + +static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) +{ + return snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_mbox1_switch, mixer)); +} + /* Native Instruments device quirks */ #define _MAKE_NI_CONTROL(bRequest,wIndex) ((bRequest) << 16 | (wIndex)) @@ -1632,6 +1757,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_microii_controls_create(mixer); break; + case USB_ID(0x0dba, 0x1000): /* Digidesign Mbox 1 */ + err = snd_mbox1_create_sync_switch(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, -- cgit v1.1 From c63fcb9b67777b906c4515a868afbd96bae4e799 Mon Sep 17 00:00:00 2001 From: Damien Zammit Date: Wed, 12 Nov 2014 01:09:55 +1100 Subject: ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer This patch provides duplex support for the Digidesign Mbox 1 sound card and has been a work in progress for about a year. Users have confirmed on my website that previous versions of this patch have worked on the hardware and I have been testing extensively. It also enables the mixer control for providing clock source selector based on the previous patch. The sample rate has been hardcoded to 48kHz because it works better with the S/PDIF sync mode when the sample rate is locked. This is the highest rate that the device supports and no loss of functionality is observed by restricting the sample rate apart from the inability to selec a lower rate. Signed-off-by: Damien Zammit Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 41 ++++++++++++++++++++++++++++++++--------- 1 file changed, 32 insertions(+), 9 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..13f44fd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2944,7 +2944,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]){ { .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE, + .type = QUIRK_AUDIO_STANDARD_MIXER, }, { .ifnum = 1, @@ -2955,16 +2955,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .attributes = 0x4, .endpoint = 0x02, - .ep_attr = 0x01, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000, - .rate_min = 44100, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_SYNC, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, .rate_max = 48000, - .nr_rates = 2, + .nr_rates = 1, .rate_table = (unsigned int[]) { - 44100, 48000 + 48000 + } + } + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x4, + .endpoint = 0x81, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 } } }, @@ -2972,7 +2996,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = -1 } } - } }, -- cgit v1.1 From 91159ecaf4401f7b4b0d48f59c877a0779209af5 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 11 Nov 2014 15:31:19 +0800 Subject: ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2 It is the slot selection in DAC side of IF1 and IF2. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 330 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 48 ++++++- 2 files changed, 358 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 4b6f7d5..5d317c68 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1906,6 +1906,126 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); +/* TDM IF1/2 DAC Data Selection */ /* MX-3E[14:12][10:8][6:4][2:0] + MX-3F[14:12][10:8][6:4][2:0] + MX-43[14:12][10:8][6:4][2:0] + MX-44[14:12][10:8][6:4][2:0] */ +static const char * const rt5677_if12_dac_tdm_sel_src[] = { + "Slot0", "Slot1", "Slot2", "Slot3", "Slot4", "Slot5", "Slot6", "Slot7" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac0_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC0 TDM Source", rt5677_if1_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac1_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC1 TDM Source", rt5677_if1_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac2_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC2 TDM Source", rt5677_if1_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac3_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC3 TDM Source", rt5677_if1_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac4_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC4 TDM Source", rt5677_if1_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac5_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC5 TDM Source", rt5677_if1_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac6_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC6 TDM Source", rt5677_if1_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac7_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC7 TDM Source", rt5677_if1_dac7_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac0_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC0 TDM Source", rt5677_if2_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac1_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC1 TDM Source", rt5677_if2_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac2_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC2 TDM Source", rt5677_if2_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac3_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC3 TDM Source", rt5677_if2_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac4_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC4 TDM Source", rt5677_if2_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac5_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC5 TDM Source", rt5677_if2_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac6_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC6 TDM Source", rt5677_if2_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac7_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC7 TDM Source", rt5677_if2_dac7_tdm_sel_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2389,6 +2509,40 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_MUX("SLB ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_slb_adc4_mux), + SND_SOC_DAPM_MUX("IF1 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac7_tdm_sel_mux), + + SND_SOC_DAPM_MUX("IF2 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac7_tdm_sel_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), @@ -3036,14 +3190,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 DAC6", NULL, "I2S1" }, { "IF1 DAC7", NULL, "I2S1" }, - { "IF1 DAC01", NULL, "IF1 DAC0" }, - { "IF1 DAC01", NULL, "IF1 DAC1" }, - { "IF1 DAC23", NULL, "IF1 DAC2" }, - { "IF1 DAC23", NULL, "IF1 DAC3" }, - { "IF1 DAC45", NULL, "IF1 DAC4" }, - { "IF1 DAC45", NULL, "IF1 DAC5" }, - { "IF1 DAC67", NULL, "IF1 DAC6" }, - { "IF1 DAC67", NULL, "IF1 DAC7" }, + { "IF1 DAC0 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC0 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC0 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC0 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC0 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC0 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC0 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC0 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC1 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC1 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC1 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC1 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC1 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC1 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC1 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC1 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC2 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC2 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC2 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC2 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC2 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC2 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC2 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC2 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC3 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC3 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC3 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC3 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC3 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC3 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC3 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC3 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC4 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC4 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC4 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC4 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC4 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC4 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC4 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC4 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC5 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC5 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC5 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC5 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC5 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC5 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC5 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC5 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC6 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC6 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC6 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC6 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC6 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC6 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC6 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC6 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC7 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC7 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC7 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC7 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC7 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC7 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC7 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC7 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC01", NULL, "IF1 DAC0 Mux" }, + { "IF1 DAC01", NULL, "IF1 DAC1 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC2 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC3 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC4 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC5 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC6 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC7 Mux" }, { "IF2 DAC0", NULL, "AIF2RX" }, { "IF2 DAC1", NULL, "AIF2RX" }, @@ -3062,14 +3288,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 DAC6", NULL, "I2S2" }, { "IF2 DAC7", NULL, "I2S2" }, - { "IF2 DAC01", NULL, "IF2 DAC0" }, - { "IF2 DAC01", NULL, "IF2 DAC1" }, - { "IF2 DAC23", NULL, "IF2 DAC2" }, - { "IF2 DAC23", NULL, "IF2 DAC3" }, - { "IF2 DAC45", NULL, "IF2 DAC4" }, - { "IF2 DAC45", NULL, "IF2 DAC5" }, - { "IF2 DAC67", NULL, "IF2 DAC6" }, - { "IF2 DAC67", NULL, "IF2 DAC7" }, + { "IF2 DAC0 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC0 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC0 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC0 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC0 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC0 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC0 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC0 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC1 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC1 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC1 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC1 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC1 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC1 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC1 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC1 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC2 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC2 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC2 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC2 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC2 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC2 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC2 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC2 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC3 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC3 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC3 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC3 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC3 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC3 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC3 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC3 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC4 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC4 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC4 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC4 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC4 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC4 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC4 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC4 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC5 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC5 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC5 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC5 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC5 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC5 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC5 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC5 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC6 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC6 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC6 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC6 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC6 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC6 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC6 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC6 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC7 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC7 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC7 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC7 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC7 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC7 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC7 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC7 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC01", NULL, "IF2 DAC0 Mux" }, + { "IF2 DAC01", NULL, "IF2 DAC1 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC2 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC3 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC4 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC5 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC6 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC7 Mux" }, { "IF3 DAC", NULL, "AIF3RX" }, { "IF3 DAC", NULL, "I2S3" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 9d473b2..2979d5a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,7 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3B TDM1 control 1 (0x3b) */ +/* TDM1 control 1 (0x3b) */ #define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF1_ADC_MODE_SFT 12 #define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) @@ -813,7 +813,7 @@ #define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF1_ADC4_SWAP_SFT 0 -/* MX3C TDM1 control 2 (0x3c) */ +/* TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -825,7 +825,27 @@ #define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF1_ADC_CTRL_SFT 0 -/* MX40 TDM2 control 1 (0x40) */ +/* TDM1 control 4 (0x3e) */ +#define RT5677_IF1_DAC0_MASK (0x7 << 12) +#define RT5677_IF1_DAC0_SFT 12 +#define RT5677_IF1_DAC1_MASK (0x7 << 8) +#define RT5677_IF1_DAC1_SFT 8 +#define RT5677_IF1_DAC2_MASK (0x7 << 4) +#define RT5677_IF1_DAC2_SFT 4 +#define RT5677_IF1_DAC3_MASK (0x7 << 0) +#define RT5677_IF1_DAC3_SFT 0 + +/* TDM1 control 5 (0x3f) */ +#define RT5677_IF1_DAC4_MASK (0x7 << 12) +#define RT5677_IF1_DAC4_SFT 12 +#define RT5677_IF1_DAC5_MASK (0x7 << 8) +#define RT5677_IF1_DAC5_SFT 8 +#define RT5677_IF1_DAC6_MASK (0x7 << 4) +#define RT5677_IF1_DAC6_SFT 4 +#define RT5677_IF1_DAC7_MASK (0x7 << 0) +#define RT5677_IF1_DAC7_SFT 0 + +/* TDM2 control 1 (0x40) */ #define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF2_ADC_MODE_SFT 12 #define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) @@ -839,7 +859,7 @@ #define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF2_ADC4_SWAP_SFT 0 -/* MX41 TDM2 control 2 (0x41) */ +/* TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -851,6 +871,26 @@ #define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF2_ADC_CTRL_SFT 0 +/* TDM2 control 4 (0x43) */ +#define RT5677_IF2_DAC0_MASK (0x7 << 12) +#define RT5677_IF2_DAC0_SFT 12 +#define RT5677_IF2_DAC1_MASK (0x7 << 8) +#define RT5677_IF2_DAC1_SFT 8 +#define RT5677_IF2_DAC2_MASK (0x7 << 4) +#define RT5677_IF2_DAC2_SFT 4 +#define RT5677_IF2_DAC3_MASK (0x7 << 0) +#define RT5677_IF2_DAC3_SFT 0 + +/* TDM2 control 5 (0x44) */ +#define RT5677_IF2_DAC4_MASK (0x7 << 12) +#define RT5677_IF2_DAC4_SFT 12 +#define RT5677_IF2_DAC5_MASK (0x7 << 8) +#define RT5677_IF2_DAC5_SFT 8 +#define RT5677_IF2_DAC6_MASK (0x7 << 4) +#define RT5677_IF2_DAC6_SFT 4 +#define RT5677_IF2_DAC7_MASK (0x7 << 0) +#define RT5677_IF2_DAC7_SFT 0 + /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) #define RT5677_DMIC_1_EN_SFT 15 -- cgit v1.1 From a7a3324a602cd7ebabfb7f5990006ec4f3d6449f Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Wed, 12 Nov 2014 16:38:05 +0200 Subject: ASoC: davinci-mcasp: Add overrun/underrun event handling An underrun (playback) event occurs when the serializer transfer data from the XRBUF buffer to the XRSR shift register, but the XRBUF hasn't been filled. Similarly, the overrun (capture) event occurs when data from the XRSR shift register is transferred to the XRBUF but it hasn't been read yet. These events are handled as XRUN events that cause the pcm to stop. The stream has to be explicitly restarted by the userspace which ensures that after stopping/starting McASP the data transfer is aligned with DMA. The other possibility was to internally stop and start McASP without DMA even knowing about it. Signed-off-by: Misael Lopez Cruz Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 124 +++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 11 ++ 3 files changed, 136 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 60ca079..46bc982 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -32,7 +32,7 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupts : Interrupt numbers for McASP - interrupt-names : Known interrupt names are "tx" and "rx" - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a9822c7..e460f97 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -70,6 +70,7 @@ struct davinci_mcasp { void __iomem *base; u32 fifo_base; struct device *dev; + struct snd_pcm_substream *substreams[2]; /* McASP specific data */ int tdm_slots; @@ -80,6 +81,7 @@ struct davinci_mcasp { u8 bclk_div; u16 bclk_lrclk_ratio; int streams; + u32 irq_request[2]; int sysclk_freq; bool bclk_master; @@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable receive IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable transmit IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); + /* * In synchronous mode stop the TX clocks if no other stream is * running @@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { u32 val = 0; + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); + /* * In synchronous mode keep TX clocks running if the capture stream is * still running. @@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) mcasp_stop_rx(mcasp); } +static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG); + if (stat & XUNDRN & irq_mask) { + dev_warn(mcasp->dev, "Transmit buffer underflow\n"); + handled_mask |= XUNDRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + +static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG); + if (stat & ROVRN & irq_mask) { + dev_warn(mcasp->dev, "Receive buffer overflow\n"); + handled_mask |= ROVRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -869,6 +957,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir; + mcasp->substreams[substream->stream] = substream; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; @@ -907,6 +997,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + mcasp->substreams[substream->stream] = NULL; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; @@ -1256,6 +1348,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; + char *irq_name; + int irq; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -1336,6 +1430,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "rx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_rx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "RX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + + irq = platform_get_irq_byname(pdev, "tx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_tx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "TX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + } + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (dat) mcasp->dat_port = true; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9737108..79dc511 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -256,6 +256,7 @@ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits */ +#define XRERR BIT(8) /* Transmit/Receive error */ #define XRDATA BIT(5) /* Transmit/Receive data ready */ /* @@ -285,6 +286,16 @@ #define TXDATADMADIS BIT(0) /* + * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits + */ +#define ROVRN BIT(0) + +/* + * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits + */ +#define XUNDRN BIT(0) + +/* * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits */ #define FIFO_ENABLE BIT(16) -- cgit v1.1 From ef326f4bb2675e9309ba318b19442d9823e58ee2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 12 Nov 2014 14:55:26 +0000 Subject: ASoC: arizona: Add support for 768kHz DMIC operation The new IPs supports a new lower frequency 768kHz DMIC operation add support for this into the OSR control. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c05e7a..786464f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -648,7 +648,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { - "1.536MHz", "3.072MHz", "6.144MHz", + "1.536MHz", "3.072MHz", "6.144MHz", "768kHz", }; const struct soc_enum arizona_in_dmic_osr[] = { -- cgit v1.1 From e9c7f34a7eba13e1a53212246c607d13574f9eff Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 12 Nov 2014 16:12:46 +0000 Subject: ASoC: arizona: Add DSP_B and LEFT_J mode support Signed-off-by: Richard Fitzgerald Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 786464f..19887bf 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -61,6 +61,11 @@ #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 +#define ARIZONA_FMT_DSP_MODE_A 0 +#define ARIZONA_FMT_DSP_MODE_B 1 +#define ARIZONA_FMT_I2S_MODE 2 +#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - mode = 0; + mode = ARIZONA_FMT_DSP_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "DSP_B not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_DSP_MODE_B; break; case SND_SOC_DAIFMT_I2S: - mode = 2; + mode = ARIZONA_FMT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "LEFT_J not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE; break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", @@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { + val &= ARIZONA_AIF1_FMT_MASK; + if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); bclk_target /= channels; bclk_target *= channels + 1; -- cgit v1.1 From caaeb6a96f35af14f615838b924f54c9d0f33ab3 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 10 Nov 2014 20:00:40 +0100 Subject: ASoC: sh: fsi: Document SoC-specific bindings The documentation only mentioned the generic fallback compatible property. Add the missing SoC-specific compatible properties, some of which are already in use. Also fix a small typo, while we're at it. Signed-off-by: Geert Uytterhoeven Acked-by: Simon Horman Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,fsi.txt | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt index c5be003..0d0ab51 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt @@ -1,11 +1,16 @@ Renesas FSI Required properties: -- compatible : "renesas,sh_fsi2" or "renesas,sh_fsi" +- compatible : "renesas,fsi2-", + "renesas,sh_fsi2" or "renesas,sh_fsi" as + fallback. + Examples with soctypes are: + - "renesas,fsi2-r8a7740" (R-Mobile A1) + - "renesas,fsi2-sh73a0" (SH-Mobile AG5) - reg : Should contain the register physical address and length - interrupts : Should contain FSI interrupt -- fsia,spdif-connection : FSI is connected by S/PDFI +- fsia,spdif-connection : FSI is connected by S/PDIF - fsia,stream-mode-support : FSI supports 16bit stream mode. - fsia,use-internal-clock : FSI uses internal clock when master mode. -- cgit v1.1 From 56ba98acc398883324c0e70dc8aee1dc53eb2331 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 10 Nov 2014 20:00:42 +0100 Subject: ASoC: rsnd: Document SoC-specific bindings The documentation only mentioned the generic fallback compatible property. Add the missing SoC-specific compatible properties, which are already in use. Also drop a bogus 0x unit-address prefix while we're at it. Signed-off-by: Geert Uytterhoeven Acked-by: Simon Horman Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index aa697ab..2dd690b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -1,8 +1,12 @@ Renesas R-Car sound Required properties: -- compatible : "renesas,rcar_sound-gen1" if generation1 +- compatible : "renesas,rcar_sound-", fallbacks + "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 + Examples with soctypes are: + - "renesas,rcar_sound-r8a7790" (R-Car H2) + - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 @@ -35,9 +39,9 @@ DAI subnode properties: Example: -rcar_sound: rcar_sound@0xffd90000 { +rcar_sound: rcar_sound@ec500000 { #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-gen2"; + compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; reg = <0 0xec500000 0 0x1000>, /* SCU */ <0 0xec5a0000 0 0x100>, /* ADG */ <0 0xec540000 0 0x1000>, /* SSIU */ -- cgit v1.1 From ef9566a3a1c0e46dadfa6c722e8a685ac0cea081 Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 12 Nov 2014 12:06:59 -0600 Subject: Revert "ALSA: usb-audio: Add quirk for Focusrite Scarlett This reverts commit 1762a59d8e8b5e99f6f4a0f292b40f3cacb108ba. This quirk is not needed because support for the Scarlett mixers will be added. Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 51 ------------------------------------------------ 1 file changed, 51 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 13f44fd..013cba8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2667,57 +2667,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, -{ - /* - * Focusrite Scarlett 18i6 - * - * Avoid mixer creation, which otherwise fails because some of - * the interface descriptor subtypes for interface 0 are - * unknown. That should be fixed or worked-around but this at - * least allows the device to be used successfully with a DAW - * and an external mixer. See comments below about other - * ignored interfaces. - */ - USB_DEVICE(0x1235, 0x8004), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Focusrite", - .product_name = "Scarlett 18i6", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Control Device) */ - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - /* InterfaceSubClass 1 (Device Firmware Update) */ - .ifnum = 5, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } - } - } -}, /* Access Music devices */ { -- cgit v1.1 From f41d6049d18694e8b3d938464432d0e51f671089 Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 12 Nov 2014 12:07:00 -0600 Subject: ALSA: usb-audio: Add private_data pointer to usb_mixer_elem_info Add a private_data pointer to usb_mixer_elem_info to allow other mixer implementations to extend the structure as necessary. Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/mixer.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 2c7b9c9..7423f99 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -53,6 +53,7 @@ struct usb_mixer_elem_info { int cached; int cache_val[MAX_CHANNELS]; u8 initialized; + void *private_data; }; int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, -- cgit v1.1 From eef90451605d79a5703756505087e0ef16da9077 Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 12 Nov 2014 12:07:01 -0600 Subject: ALSA: usb-audio: make set_*_mix_values functions public Make the functions set_cur_mix_value and get_cur_mix_value accessible by files that include mixer.h. In addition make usb_mixer_elem_free accessible. This allows reuse of these functions by mixers that may require quirks. The following summarizes the renamed functions: - set_cur_mix_value -> snd_usb_set_cur_mix_value - get_cur_mix_value -> snd_usb_get_cur_mix_value - usb_mixer_elem_free -> snd_usb_mixer_elem_free Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 34 +++++++++++++++++----------------- sound/usb/mixer.h | 8 ++++++++ sound/usb/mixer_quirks.c | 9 +-------- 3 files changed, 26 insertions(+), 25 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e4aaa21..14e1455 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -412,7 +412,7 @@ static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, value); } -static int get_cur_mix_value(struct usb_mixer_elem_info *cval, +int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int *value) { int err; @@ -497,7 +497,7 @@ static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, return snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, validx, value); } -static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, +int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; @@ -815,7 +815,7 @@ static struct usb_feature_control_info audio_feature_info[] = { }; /* private_free callback */ -static void usb_mixer_elem_free(struct snd_kcontrol *kctl) +void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) { kfree(kctl->private_data); kctl->private_data = NULL; @@ -998,7 +998,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, else test -= cval->res; if (test < cval->min || test > cval->max || - set_cur_mix_value(cval, minchn, 0, test) || + snd_usb_set_cur_mix_value(cval, minchn, 0, test) || get_cur_mix_raw(cval, minchn, &check)) { cval->res = last_valid_res; break; @@ -1007,7 +1007,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, break; cval->res *= 2; } - set_cur_mix_value(cval, minchn, 0, saved); + snd_usb_set_cur_mix_value(cval, minchn, 0, saved); } cval->initialized = 1; @@ -1086,7 +1086,7 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, for (c = 0; c < MAX_CHANNELS; c++) { if (!(cval->cmask & (1 << c))) continue; - err = get_cur_mix_value(cval, c + 1, cnt, &val); + err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &val); if (err < 0) return cval->mixer->ignore_ctl_error ? 0 : err; val = get_relative_value(cval, val); @@ -1096,7 +1096,7 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, return 0; } else { /* master channel */ - err = get_cur_mix_value(cval, 0, 0, &val); + err = snd_usb_get_cur_mix_value(cval, 0, 0, &val); if (err < 0) return cval->mixer->ignore_ctl_error ? 0 : err; val = get_relative_value(cval, val); @@ -1118,26 +1118,26 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, for (c = 0; c < MAX_CHANNELS; c++) { if (!(cval->cmask & (1 << c))) continue; - err = get_cur_mix_value(cval, c + 1, cnt, &oval); + err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &oval); if (err < 0) return cval->mixer->ignore_ctl_error ? 0 : err; val = ucontrol->value.integer.value[cnt]; val = get_abs_value(cval, val); if (oval != val) { - set_cur_mix_value(cval, c + 1, cnt, val); + snd_usb_set_cur_mix_value(cval, c + 1, cnt, val); changed = 1; } cnt++; } } else { /* master channel */ - err = get_cur_mix_value(cval, 0, 0, &oval); + err = snd_usb_get_cur_mix_value(cval, 0, 0, &oval); if (err < 0) return cval->mixer->ignore_ctl_error ? 0 : err; val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_mix_value(cval, 0, 0, val); + snd_usb_set_cur_mix_value(cval, 0, 0, val); changed = 1; } } @@ -1250,7 +1250,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, /* * If all channels in the mask are marked read-only, make the control - * read-only. set_cur_mix_value() will check the mask again and won't + * read-only. snd_usb_set_cur_mix_value() will check the mask again and won't * issue write commands to read-only channels. */ if (cval->channels == readonly_mask) @@ -1263,7 +1263,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, kfree(cval); return; } - kctl->private_free = usb_mixer_elem_free; + kctl->private_free = snd_usb_mixer_elem_free; len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; @@ -1546,7 +1546,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, kfree(cval); return; } - kctl->private_free = usb_mixer_elem_free; + kctl->private_free = snd_usb_mixer_elem_free; len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (!len) @@ -1846,7 +1846,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, kfree(cval); return -ENOMEM; } - kctl->private_free = usb_mixer_elem_free; + kctl->private_free = snd_usb_mixer_elem_free; if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name))) { /* nothing */ ; @@ -2526,7 +2526,7 @@ static int restore_mixer_value(struct usb_mixer_elem_info *cval) if (!(cval->cmask & (1 << c))) continue; if (cval->cached & (1 << c)) { - err = set_cur_mix_value(cval, c + 1, idx, + err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) return err; @@ -2536,7 +2536,7 @@ static int restore_mixer_value(struct usb_mixer_elem_info *cval) } else { /* master */ if (cval->cached) { - err = set_cur_mix_value(cval, 0, 0, *cval->cache_val); + err = snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); if (err < 0) return err; } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 7423f99..2478a84 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -76,4 +76,12 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); #endif +int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, + int index, int value); + +int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, + int channel, int index, int *value); + +extern void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl); + #endif /* __USBMIXER_H */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4520316..b8b1f48 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -52,13 +52,6 @@ struct std_mono_table { snd_kcontrol_tlv_rw_t *tlv_callback; }; -/* private_free callback */ -static void usb_mixer_elem_free(struct snd_kcontrol *kctl) -{ - kfree(kctl->private_data); - kctl->private_data = NULL; -} - /* This function allows for the creation of standard UAC controls. * See the quirks for M-Audio FTUs or Ebox-44. * If you don't want to set a TLV callback pass NULL. @@ -108,7 +101,7 @@ static int snd_create_std_mono_ctl_offset(struct usb_mixer_interface *mixer, /* Set name */ snprintf(kctl->id.name, sizeof(kctl->id.name), name); - kctl->private_free = usb_mixer_elem_free; + kctl->private_free = snd_usb_mixer_elem_free; /* set TLV */ if (tlv_callback) { -- cgit v1.1 From 76b188c4b370876018e3a778ec11a94a5316dbe4 Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 12 Nov 2014 12:07:02 -0600 Subject: ALSA: usb-audio: Scarlett mixer interface for 6i6, 18i6, 18i8 and 18i20 This code contains the Scarlett mixer interface code that was originally written by Tobias Hoffman and Robin Gareus. Because the device doesn't properly implement UAC2 this code adds a mixer quirk for the device. Changes from the original code include removing the metering code along with dead code and comments. Compiler warnings were fixed. The code to initialize the sampling rate was causing a crash this was fixed as discussed on the mailing list. Error, and info messages were convered to dev_err and dev_info interfaces. The custom scarlett_mixer_elem_info struct was replaced with the more generic usb_mixer_elem_info to be able to recycle more code from mixer.c. This patch also makes additional modifications based on upstream comments. Individual control creation functions are removed and a generic function is no used. Macros for function calls are removed to improve readability. Hardcoded control initialization is removed. Save to HW functionality has been removed. Strings for enums are created dynamically for the mixer. Strings used for controls are now SNDRV_CTL_ELEM_ID_NAME_MAXLEN length. Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/Makefile | 1 + sound/usb/mixer_quirks.c | 9 + sound/usb/mixer_scarlett.c | 967 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_scarlett.h | 6 + 4 files changed, 983 insertions(+) create mode 100644 sound/usb/mixer_scarlett.c create mode 100644 sound/usb/mixer_scarlett.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2b92f0d..bcee406 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -9,6 +9,7 @@ snd-usb-audio-objs := card.o \ helper.o \ mixer.o \ mixer_quirks.o \ + mixer_scarlett.o \ pcm.o \ proc.o \ quirks.o \ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index b8b1f48..7b9331a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -41,6 +41,7 @@ #include "usbaudio.h" #include "mixer.h" #include "mixer_quirks.h" +#include "mixer_scarlett.h" #include "helper.h" extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl; @@ -1770,6 +1771,14 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) /* detection is disabled in mixer_maps.c */ err = snd_create_std_mono_table(mixer, ebox44_table); break; + + case USB_ID(0x1235, 0x8012): /* Focusrite Scarlett 6i6 */ + case USB_ID(0x1235, 0x8002): /* Focusrite Scarlett 8i6 */ + case USB_ID(0x1235, 0x8004): /* Focusrite Scarlett 18i6 */ + case USB_ID(0x1235, 0x8014): /* Focusrite Scarlett 18i8 */ + case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ + err = snd_scarlett_controls_create(mixer); + break; } return err; diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c new file mode 100644 index 0000000..a0a8745 --- /dev/null +++ b/sound/usb/mixer_scarlett.c @@ -0,0 +1,967 @@ +/* + * Scarlett Driver for ALSA + * + * Copyright (c) 2013 by Tobias Hoffmann + * Copyright (c) 2013 by Robin Gareus + * Copyright (c) 2002 by Takashi Iwai + * Copyright (c) 2014 by Chris J Arges + * + * Many codes borrowed from audio.c by + * Alan Cox (alan at lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer at ife.ee.ethz.ch) + * + * Code cleanup: + * David Henningsson + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +/* + * Rewritten and extended to support more models, e.g. Scarlett 18i8. + * + * Auto-detection via UAC2 is not feasible to properly discover the vast + * majority of features. It's related to both Linux/ALSA's UAC2 as well as + * Focusrite's implementation of it. Eventually quirks may be sufficient but + * right now it's a major headache to work arount these things. + * + * NB. Neither the OSX nor the win driver provided by Focusrite performs + * discovery, they seem to operate the same as this driver. + */ + +/* Mixer Interface for the Focusrite Scarlett 18i6 audio interface. + * + * The protocol was reverse engineered by looking at communication between + * Scarlett MixControl (v 1.2.128.0) and the Focusrite(R) Scarlett 18i6 + * (firmware v305) using wireshark and usbmon in January 2013. + * Extended in July 2013. + * + * this mixer gives complete access to all features of the device: + * - change Impedance of inputs (Line-in, Mic / Instrument, Hi-Z) + * - select clock source + * - dynamic input to mixer-matrix assignment + * - 18 x 6 mixer-matrix gain stages + * - bus routing & volume control + * - automatic re-initialization on connect if device was power-cycled + * + * USB URB commands overview (bRequest = 0x01 = UAC2_CS_CUR) + * wIndex + * 0x01 Analog Input line/instrument impedance switch, wValue=0x0901 + + * channel, data=Line/Inst (2bytes) + * pad (-10dB) switch, wValue=0x0b01 + channel, data=Off/On (2bytes) + * ?? wValue=0x0803/04, ?? (2bytes) + * 0x0a Master Volume, wValue=0x0200+bus[0:all + only 1..4?] data(2bytes) + * Bus Mute/Unmute wValue=0x0100+bus[0:all + only 1..4?], data(2bytes) + * 0x28 Clock source, wValue=0x0100, data={1:int,2:spdif,3:adat} (1byte) + * 0x29 Set Sample-rate, wValue=0x0100, data=sample-rate(4bytes) + * 0x32 Mixer mux, wValue=0x0600 + mixer-channel, data=input-to-connect(2bytes) + * 0x33 Output mux, wValue=bus, data=input-to-connect(2bytes) + * 0x34 Capture mux, wValue=0...18, data=input-to-connect(2bytes) + * 0x3c Matrix Mixer gains, wValue=mixer-node data=gain(2bytes) + * ?? [sometimes](4bytes, e.g 0x000003be 0x000003bf ...03ff) + * + * USB reads: (i.e. actually issued by original software) + * 0x01 wValue=0x0901+channel (1byte!!), wValue=0x0b01+channed (1byte!!) + * 0x29 wValue=0x0100 sample-rate(4bytes) + * wValue=0x0200 ?? 1byte (only once) + * 0x2a wValue=0x0100 ?? 4bytes, sample-rate2 ?? + * + * USB reads with bRequest = 0x03 = UAC2_CS_MEM + * 0x3c wValue=0x0002 1byte: sync status (locked=1) + * wValue=0x0000 18*2byte: peak meter (inputs) + * wValue=0x0001 8(?)*2byte: peak meter (mix) + * wValue=0x0003 6*2byte: peak meter (pcm/daw) + * + * USB write with bRequest = 0x03 + * 0x3c Save settings to hardware: wValue=0x005a, data=0xa5 + * + * + * + * /--------------\ 18chn 6chn /--------------\ + * | Hardware in +--+-------\ /------+--+ ALSA PCM out | + * \--------------/ | | | | \--------------/ + * | | | | + * | v v | + * | +---------------+ | + * | \ Matrix Mux / | + * | +-----+-----+ | + * | | | + * | | 18chn | + * | v | + * | +-----------+ | + * | | Mixer | | + * | | Matrix | | + * | | | | + * | | 18x6 Gain | | + * | | stages | | + * | +-----+-----+ | + * | | | + * | | | + * | 18chn | 6chn | 6chn + * v v v + * ========================= + * +---------------+ +--—------------+ + * \ Output Mux / \ Capture Mux / + * +-----+-----+ +-----+-----+ + * | | + * | 6chn | + * v | + * +-------------+ | + * | Master Gain | | + * +------+------+ | + * | | + * | 6chn | 18chn + * | (3 stereo pairs) | + * /--------------\ | | /--------------\ + * | Hardware out |<--/ \-->| ALSA PCM in | + * \--------------/ \--------------/ + * + * + */ + +#include +#include +#include + +#include +#include +#include + +#include "usbaudio.h" +#include "mixer.h" +#include "helper.h" +#include "power.h" + +#include "mixer_scarlett.h" + +/* some gui mixers can't handle negative ctl values */ +#define SND_SCARLETT_LEVEL_BIAS 128 +#define SND_SCARLETT_MATRIX_IN_MAX 18 +#define SND_SCARLETT_CONTROLS_MAX 10 +#define SND_SCARLETT_OFFSETS_MAX 5 + +enum { + SCARLETT_OUTPUTS, + SCARLETT_SWITCH_IMPEDANCE, + SCARLETT_SWITCH_PAD, +}; + +enum { + SCARLETT_OFFSET_PCM = 0, + SCARLETT_OFFSET_ANALOG = 1, + SCARLETT_OFFSET_SPDIF = 2, + SCARLETT_OFFSET_ADAT = 3, + SCARLETT_OFFSET_MIX = 4, +}; + +struct scarlett_mixer_elem_enum_info { + int start; + int len; + int offsets[SND_SCARLETT_OFFSETS_MAX]; + char const * const *names; +}; + +struct scarlett_mixer_control { + unsigned char num; + unsigned char type; + const char *name; +}; + +struct scarlett_device_info { + int matrix_in; + int matrix_out; + int input_len; + int output_len; + + struct scarlett_mixer_elem_enum_info opt_master; + struct scarlett_mixer_elem_enum_info opt_matrix; + + /* initial values for matrix mux */ + int matrix_mux_init[SND_SCARLETT_MATRIX_IN_MAX]; + + int num_controls; /* number of items in controls */ + const struct scarlett_mixer_control controls[SND_SCARLETT_CONTROLS_MAX]; +}; + +/********************** Enum Strings *************************/ + +static const struct scarlett_mixer_elem_enum_info opt_pad = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "0dB", "-10dB" + } +}; + +static const struct scarlett_mixer_elem_enum_info opt_impedance = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Line", "Hi-Z" + } +}; + +static const struct scarlett_mixer_elem_enum_info opt_clock = { + .start = 1, + .len = 3, + .offsets = {}, + .names = (char const * const []){ + "Internal", "SPDIF", "ADAT" + } +}; + +static const struct scarlett_mixer_elem_enum_info opt_sync = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "No Lock", "Locked" + } +}; + +static int scarlett_ctl_switch_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int scarlett_ctl_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + int i, err, val; + + for (i = 0; i < elem->channels; i++) { + err = snd_usb_get_cur_mix_value(elem, i, i, &val); + if (err < 0) + return err; + + val = !val; /* invert mute logic for mixer */ + ucontrol->value.integer.value[i] = val; + } + + return 0; +} + +static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + int i, changed = 0; + int err, oval, val; + + for (i = 0; i < elem->channels; i++) { + err = snd_usb_get_cur_mix_value(elem, i, i, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[i]; + val = !val; + if (oval != val) { + err = snd_usb_set_cur_mix_value(elem, i, i, val); + if (err < 0) + return err; + + changed = 1; + } + } + + return changed; +} + +static int scarlett_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)kctl->private_value + + SND_SCARLETT_LEVEL_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + int i, err, val; + + for (i = 0; i < elem->channels; i++) { + err = snd_usb_get_cur_mix_value(elem, i, i, &val); + if (err < 0) + return err; + + val = clamp(val / 256, -128, (int)kctl->private_value) + + SND_SCARLETT_LEVEL_BIAS; + ucontrol->value.integer.value[i] = val; + } + + return 0; +} + +static int scarlett_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + int i, changed = 0; + int err, oval, val; + + for (i = 0; i < elem->channels; i++) { + err = snd_usb_get_cur_mix_value(elem, i, i, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[i] - + SND_SCARLETT_LEVEL_BIAS; + val = val * 256; + if (oval != val) { + err = snd_usb_set_cur_mix_value(elem, i, i, val); + if (err < 0) + return err; + + changed = 1; + } + } + + return changed; +} + +static void scarlett_generate_name(int i, char *dst, int offsets[]) +{ + if (i > offsets[SCARLETT_OFFSET_MIX]) + sprintf(dst, "Mix %c", + 'A'+(i - offsets[SCARLETT_OFFSET_MIX] - 1)); + else if (i > offsets[SCARLETT_OFFSET_ADAT]) + sprintf(dst, "ADAT %d", i - offsets[SCARLETT_OFFSET_ADAT]); + else if (i > offsets[SCARLETT_OFFSET_SPDIF]) + sprintf(dst, "SPDIF %d", i - offsets[SCARLETT_OFFSET_SPDIF]); + else if (i > offsets[SCARLETT_OFFSET_ANALOG]) + sprintf(dst, "Analog %d", i - offsets[SCARLETT_OFFSET_ANALOG]); + else if (i > offsets[SCARLETT_OFFSET_PCM]) + sprintf(dst, "PCM %d", i - offsets[SCARLETT_OFFSET_PCM]); + else + sprintf(dst, "Off"); +} + +static int scarlett_ctl_enum_dynamic_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett_mixer_elem_enum_info *opt = elem->private_data; + unsigned int items = opt->len; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = elem->channels; + uinfo->value.enumerated.items = items; + + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + + /* generate name dynamically based on item number and offset info */ + scarlett_generate_name(uinfo->value.enumerated.item, + uinfo->value.enumerated.name, + opt->offsets); + + return 0; +} + +static int scarlett_ctl_enum_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett_mixer_elem_enum_info *opt = elem->private_data; + + return snd_ctl_enum_info(uinfo, elem->channels, opt->len, + (const char * const *)opt->names); +} + +static int scarlett_ctl_enum_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett_mixer_elem_enum_info *opt = elem->private_data; + int err, val; + + err = snd_usb_get_cur_mix_value(elem, 0, 0, &val); + if (err < 0) + return err; + + val = clamp(val - opt->start, 0, opt->len-1); + + ucontrol->value.enumerated.item[0] = val; + + return 0; +} + +static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett_mixer_elem_enum_info *opt = elem->private_data; + int err, oval, val; + + err = snd_usb_get_cur_mix_value(elem, 0, 0, &oval); + if (err < 0) + return err; + + val = ucontrol->value.integer.value[0]; + val = val + opt->start; + if (val != oval) { + snd_usb_set_cur_mix_value(elem, 0, 0, val); + return 1; + } + return 0; +} + +static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct snd_usb_audio *chip = elem->mixer->chip; + unsigned char buf[2 * MAX_CHANNELS] = {0, }; + int wValue = (elem->control << 8) | elem->idx_off; + int idx = snd_usb_ctrl_intf(chip) | (elem->id << 8); + int err; + + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + UAC2_CS_MEM, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | + USB_DIR_IN, wValue, idx, buf, elem->channels); + if (err < 0) + return err; + + ucontrol->value.enumerated.item[0] = clamp((int)buf[0], 0, 1); + return 0; +} + +static struct snd_kcontrol_new usb_scarlett_ctl_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_switch_info, + .get = scarlett_ctl_switch_get, + .put = scarlett_ctl_switch_put, +}; + +static const DECLARE_TLV_DB_SCALE(db_scale_scarlett_gain, -12800, 100, 0); + +static struct snd_kcontrol_new usb_scarlett_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett_ctl_info, + .get = scarlett_ctl_get, + .put = scarlett_ctl_put, + .private_value = 6, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_master = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett_ctl_info, + .get = scarlett_ctl_get, + .put = scarlett_ctl_put, + .private_value = 6, /* max value */ + .tlv = { .p = db_scale_scarlett_gain } +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_enum_info, + .get = scarlett_ctl_enum_get, + .put = scarlett_ctl_enum_put, +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_dynamic_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett_ctl_enum_dynamic_info, + .get = scarlett_ctl_enum_get, + .put = scarlett_ctl_enum_put, +}; + +static struct snd_kcontrol_new usb_scarlett_ctl_sync = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .name = "", + .info = scarlett_ctl_enum_info, + .get = scarlett_ctl_meter_get, +}; + +static int add_new_ctl(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *ncontrol, + int index, int offset, int num, + int val_type, int channels, const char *name, + const struct scarlett_mixer_elem_enum_info *opt, + struct usb_mixer_elem_info **elem_ret +) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + int err; + + elem = kzalloc(sizeof(*elem), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->mixer = mixer; + elem->control = offset; + elem->idx_off = num; + elem->id = index; + elem->val_type = val_type; + + elem->channels = channels; + + /* add scarlett_mixer_elem_enum_info struct */ + elem->private_data = (void *)opt; + + kctl = snd_ctl_new1(ncontrol, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); + + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) + return err; + + if (elem_ret) + *elem_ret = elem; + + return 0; +} + +static int add_output_ctls(struct usb_mixer_interface *mixer, + int index, const char *name, + const struct scarlett_device_info *info) +{ + int err; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct usb_mixer_elem_info *elem; + + /* Add mute switch */ + snprintf(mx, sizeof(mx), "Master %d (%s) Playback Switch", + index + 1, name); + err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, 0x0a, 0x01, + 2*index+1, USB_MIXER_S16, 2, mx, NULL, &elem); + if (err < 0) + return err; + + /* Add volume control and initialize to 0 */ + snprintf(mx, sizeof(mx), "Master %d (%s) Playback Volume", + index + 1, name); + err = add_new_ctl(mixer, &usb_scarlett_ctl_master, 0x0a, 0x02, + 2*index+1, USB_MIXER_S16, 2, mx, NULL, &elem); + if (err < 0) + return err; + + /* Add L channel source playback enumeration */ + snprintf(mx, sizeof(mx), "Master %dL (%s) Source Playback Enum", + index + 1, name); + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x33, 0x00, + 2*index, USB_MIXER_S16, 1, mx, &info->opt_master, + &elem); + if (err < 0) + return err; + + /* Add R channel source playback enumeration */ + snprintf(mx, sizeof(mx), "Master %dR (%s) Source Playback Enum", + index + 1, name); + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x33, 0x00, + 2*index+1, USB_MIXER_S16, 1, mx, &info->opt_master, + &elem); + if (err < 0) + return err; + + return 0; +} + +/********************** device-specific config *************************/ + +/* untested... */ +static struct scarlett_device_info s6i6_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 6, + .output_len = 6, + + .opt_master = { + .start = -1, + .len = 27, + .offsets = {0, 12, 16, 18, 18}, + .names = NULL + }, + + .opt_matrix = { + .start = -1, + .len = 19, + .offsets = {0, 12, 16, 18, 18}, + .names = NULL + }, + + .num_controls = 0, + .controls = { + { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, + { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, + { .num = 2, .type = SCARLETT_OUTPUTS, .name = "SPDIF" }, + { .num = 1, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + }, + + .matrix_mux_init = { + 12, 13, 14, 15, /* Analog -> 1..4 */ + 16, 17, /* SPDIF -> 5,6 */ + 0, 1, 2, 3, 4, 5, 6, 7, /* PCM[1..12] -> 7..18 */ + 8, 9, 10, 11 + } +}; + +/* untested... */ +static struct scarlett_device_info s8i6_info = { + .matrix_in = 18, + .matrix_out = 6, + .input_len = 8, + .output_len = 6, + + .opt_master = { + .start = -1, + .len = 25, + .offsets = {0, 12, 16, 18, 18}, + .names = NULL + }, + + .opt_matrix = { + .start = -1, + .len = 19, + .offsets = {0, 12, 16, 18, 18}, + .names = NULL + }, + + .num_controls = 7, + .controls = { + { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, + { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, + { .num = 2, .type = SCARLETT_OUTPUTS, .name = "SPDIF" }, + { .num = 1, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + }, + + .matrix_mux_init = { + 12, 13, 14, 15, /* Analog -> 1..4 */ + 16, 17, /* SPDIF -> 5,6 */ + 0, 1, 2, 3, 4, 5, 6, 7, /* PCM[1..12] -> 7..18 */ + 8, 9, 10, 11 + } +}; + +static struct scarlett_device_info s18i6_info = { + .matrix_in = 18, + .matrix_out = 6, + .input_len = 18, + .output_len = 6, + + .opt_master = { + .start = -1, + .len = 31, + .offsets = {0, 6, 14, 16, 24}, + .names = NULL, + }, + + .opt_matrix = { + .start = -1, + .len = 25, + .offsets = {0, 6, 14, 16, 24}, + .names = NULL, + }, + + .num_controls = 5, + .controls = { + { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, + { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, + { .num = 2, .type = SCARLETT_OUTPUTS, .name = "SPDIF" }, + { .num = 1, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + }, + + .matrix_mux_init = { + 6, 7, 8, 9, 10, 11, 12, 13, /* Analog -> 1..8 */ + 16, 17, 18, 19, 20, 21, /* ADAT[1..6] -> 9..14 */ + 14, 15, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + +static struct scarlett_device_info s18i8_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 18, + .output_len = 8, + + .opt_master = { + .start = -1, + .len = 35, + .offsets = {0, 8, 16, 18, 26}, + .names = NULL + }, + + .opt_matrix = { + .start = -1, + .len = 27, + .offsets = {0, 8, 16, 18, 26}, + .names = NULL + }, + + .num_controls = 10, + .controls = { + { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, + { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone 1" }, + { .num = 2, .type = SCARLETT_OUTPUTS, .name = "Headphone 2" }, + { .num = 3, .type = SCARLETT_OUTPUTS, .name = "SPDIF" }, + { .num = 1, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + }, + + .matrix_mux_init = { + 8, 9, 10, 11, 12, 13, 14, 15, /* Analog -> 1..8 */ + 18, 19, 20, 21, 22, 23, /* ADAT[1..6] -> 9..14 */ + 16, 17, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + +static struct scarlett_device_info s18i20_info = { + .matrix_in = 18, + .matrix_out = 8, + .input_len = 18, + .output_len = 20, + + .opt_master = { + .start = -1, + .len = 47, + .offsets = {0, 20, 28, 30, 38}, + .names = NULL + }, + + .opt_matrix = { + .start = -1, + .len = 39, + .offsets = {0, 20, 28, 30, 38}, + .names = NULL + }, + + .num_controls = 10, + .controls = { + { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, + { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Line 3/4" }, + { .num = 2, .type = SCARLETT_OUTPUTS, .name = "Line 5/6" }, + { .num = 3, .type = SCARLETT_OUTPUTS, .name = "Line 7/8" }, + { .num = 4, .type = SCARLETT_OUTPUTS, .name = "Line 9/10" }, + { .num = 5, .type = SCARLETT_OUTPUTS, .name = "SPDIF" }, + { .num = 6, .type = SCARLETT_OUTPUTS, .name = "ADAT 1/2" }, + { .num = 7, .type = SCARLETT_OUTPUTS, .name = "ADAT 3/4" }, + { .num = 8, .type = SCARLETT_OUTPUTS, .name = "ADAT 5/6" }, + { .num = 9, .type = SCARLETT_OUTPUTS, .name = "ADAT 7/8" }, + /*{ .num = 1, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, + { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL},*/ + }, + + .matrix_mux_init = { + 20, 21, 22, 23, 24, 25, 26, 27, /* Analog -> 1..8 */ + 30, 31, 32, 33, 34, 35, /* ADAT[1..6] -> 9..14 */ + 28, 29, /* SPDIF -> 15,16 */ + 0, 1 /* PCM[1,2] -> 17,18 */ + } +}; + + +static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, + struct scarlett_device_info *info) +{ + int i, err; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const struct scarlett_mixer_control *ctl; + struct usb_mixer_elem_info *elem; + + /* create master switch and playback volume */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, 0x0a, 0x01, 0, + USB_MIXER_S16, 1, "Master Playback Switch", NULL, + &elem); + if (err < 0) + return err; + + err = add_new_ctl(mixer, &usb_scarlett_ctl_master, 0x0a, 0x02, 0, + USB_MIXER_S16, 1, "Master Playback Volume", NULL, + &elem); + if (err < 0) + return err; + + /* iterate through controls in info struct and create each one */ + for (i = 0; i < info->num_controls; i++) { + ctl = &info->controls[i]; + + switch (ctl->type) { + case SCARLETT_OUTPUTS: + err = add_output_ctls(mixer, ctl->num, ctl->name, info); + if (err < 0) + return err; + break; + case SCARLETT_SWITCH_IMPEDANCE: + sprintf(mx, "Input %d Impedance Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x01, + 0x09, ctl->num, USB_MIXER_S16, 1, mx, + &opt_impedance, &elem); + if (err < 0) + return err; + break; + case SCARLETT_SWITCH_PAD: + sprintf(mx, "Input %d Pad Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x01, + 0x0b, ctl->num, USB_MIXER_S16, 1, mx, + &opt_pad, &elem); + if (err < 0) + return err; + break; + } + } + + return 0; +} + +/* + * Create and initialize a mixer for the Focusrite(R) Scarlett + */ +int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + struct scarlett_device_info *info; + struct usb_mixer_elem_info *elem; + static char sample_rate_buffer[4] = { '\x80', '\xbb', '\x00', '\x00' }; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (mixer->chip->usb_id) { + case USB_ID(0x1235, 0x8012): + info = &s6i6_info; + break; + case USB_ID(0x1235, 0x8002): + info = &s8i6_info; + break; + case USB_ID(0x1235, 0x8004): + info = &s18i6_info; + break; + case USB_ID(0x1235, 0x8014): + info = &s18i8_info; + break; + case USB_ID(0x1235, 0x800c): + info = &s18i20_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + /* generic function to create controls */ + err = scarlett_controls_create_generic(mixer, info); + if (err < 0) + return err; + + /* setup matrix controls */ + for (i = 0; i < info->matrix_in; i++) { + snprintf(mx, sizeof(mx), "Matrix %02d Input Playback Route", + i+1); + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x32, + 0x06, i, USB_MIXER_S16, 1, mx, + &info->opt_matrix, &elem); + if (err < 0) + return err; + + for (o = 0; o < info->matrix_out; o++) { + sprintf(mx, "Matrix %02d Mix %c Playback Volume", i+1, + o+'A'); + err = add_new_ctl(mixer, &usb_scarlett_ctl, 0x3c, 0x00, + (i << 3) + (o & 0x07), USB_MIXER_S16, + 1, mx, NULL, &elem); + if (err < 0) + return err; + + } + } + + for (i = 0; i < info->input_len; i++) { + snprintf(mx, sizeof(mx), "Input Source %02d Capture Route", + i+1); + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x34, + 0x00, i, USB_MIXER_S16, 1, mx, + &info->opt_master, &elem); + if (err < 0) + return err; + } + + /* val_len == 1 needed here */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x28, 0x01, 0, + USB_MIXER_U8, 1, "Sample Clock Source", + &opt_clock, &elem); + if (err < 0) + return err; + + /* val_len == 1 and UAC2_CS_MEM */ + err = add_new_ctl(mixer, &usb_scarlett_ctl_sync, 0x3c, 0x00, 2, + USB_MIXER_U8, 1, "Sample Clock Sync Status", + &opt_sync, &elem); + if (err < 0) + return err; + + /* initialize sampling rate to 48000 */ + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | + USB_DIR_OUT, 0x0100, snd_usb_ctrl_intf(mixer->chip) | + (0x29 << 8), sample_rate_buffer, 4); + if (err < 0) + return err; + + return err; +} diff --git a/sound/usb/mixer_scarlett.h b/sound/usb/mixer_scarlett.h new file mode 100644 index 0000000..19c592a --- /dev/null +++ b/sound/usb/mixer_scarlett.h @@ -0,0 +1,6 @@ +#ifndef __USB_MIXER_SCARLETT_H +#define __USB_MIXER_SCARLETT_H + +int snd_scarlett_controls_create(struct usb_mixer_interface *mixer); + +#endif /* __USB_MIXER_SCARLETT_H */ -- cgit v1.1 From 850577db99dbc4fdebe62d30d380de1878f77d2a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 13 Nov 2014 09:55:22 +0800 Subject: ASoC: rt5645: add register setting for TDM We need to set extra register to avoid a recording issue in TDM mode. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 5 ++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 665f8b6..57afa12 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2112,8 +2112,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, struct snd_soc_codec *codec = dai->codec; unsigned int val = 0; - if (rx_mask || tx_mask) + if (rx_mask || tx_mask) { val |= (1 << 14); + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + } switch (slots) { case 4: diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 82f681b..196daf0 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1855,6 +1855,7 @@ #define RT5645_M_BB_HPF_R_SFT 6 #define RT5645_G_BB_BST_MASK (0x3f) #define RT5645_G_BB_BST_SFT 0 +#define RT5645_G_BB_BST_25DB 0x14 /* MP3 Plus Control 1 (0xd0) */ #define RT5645_M_MP3_L_MASK (0x1 << 15) -- cgit v1.1 From 9e6280cd44c1cceaaac921567ee8c5731b7cc72b Mon Sep 17 00:00:00 2001 From: Krishna Mohan Dani Date: Thu, 13 Nov 2014 17:44:21 +0530 Subject: ASoC: rt5631: Add device tree binding documentation Document the device tree binding for the ALC5631 codec and update vendor specific prefix for the Realtek. Signed-off-by: Krishna Mohan Dani Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++++++++++++++++++ 1 file changed, 48 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt new file mode 100644 index 0000000..92b986c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5631.txt @@ -0,0 +1,48 @@ +ALC5631/RT5631 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5631" or "realtek,rt5631" + + - reg : the I2C address of the device. + +Pins on the device (for linking into audio routes): + + * SPK_OUT_R_P + * SPK_OUT_R_N + * SPK_OUT_L_P + * SPK_OUT_L_N + * HP_OUT_L + * HP_OUT_R + * AUX_OUT2_LP + * AUX_OUT2_RN + * AUX_OUT1_LP + * AUX_OUT1_RN + * AUX_IN_L_JD + * AUX_IN_R_JD + * MONO_IN_P + * MONO_IN_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MONO_OUT_P + * MONO_OUT_N + * MICBIAS1 + * MICBIAS2 + +Example: + +alc5631: alc5631@1a { + compatible = "realtek,alc5631"; + reg = <0x1a>; +}; + +or + +rt5631: rt5631@1a { + compatible = "realtek,rt5631"; + reg = <0x1a>; +}; -- cgit v1.1 From 86707f7fece1d3a34aeb1e9c7f2178fd5ff4e788 Mon Sep 17 00:00:00 2001 From: Krishna Mohan Dani Date: Thu, 13 Nov 2014 17:44:23 +0530 Subject: ASoC: rt5631: Adding the description of the codec Signed-off-by: Krishna Mohan Dani Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..7e43e97 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -487,7 +487,8 @@ config SND_SOC_RT286 depends on I2C config SND_SOC_RT5631 - tristate + tristate "Realtek ALC5631/RT5631 CODEC" + depends on I2C config SND_SOC_RT5640 tristate -- cgit v1.1 From 189c88ced169d5197c806828e275c6f063b1d499 Mon Sep 17 00:00:00 2001 From: Krishna Mohan Dani Date: Thu, 13 Nov 2014 17:44:24 +0530 Subject: ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver Signed-off-by: Krishna Mohan Dani Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 1ba27db..3b7d5e4 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1686,10 +1686,18 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { static const struct i2c_device_id rt5631_i2c_id[] = { { "rt5631", 0 }, + { "alc5631", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +static struct of_device_id rt5631_i2c_dt_ids[] = { + { .compatible = "realtek,rt5631"}, + { .compatible = "realtek,alc5631"}, + { } +}; +MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids); + static const struct regmap_config rt5631_regmap_config = { .reg_bits = 8, .val_bits = 16, @@ -1734,6 +1742,7 @@ static struct i2c_driver rt5631_i2c_driver = { .driver = { .name = "rt5631", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5631_i2c_dt_ids), }, .probe = rt5631_i2c_probe, .remove = rt5631_i2c_remove, -- cgit v1.1 From 9547c0999e50fd624cab52f94a79f0fd27a7cb84 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 28 Oct 2014 14:22:49 -0700 Subject: ALSA: 6fire: Convert byte_rev_table uses to bitrev8 Use the inline function instead of directly indexing the array. This allows some architectures with hardware instructions for bit reversals to eliminate the array. Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b02e54..62c25e7 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -316,7 +316,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = byte_rev_table[(u8) *c]; + buffer[i] = bitrev8((u8)*c); ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) { -- cgit v1.1 From 0605815e7ec21e048febcebb691d7f0cc3bdc36c Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 14 Nov 2014 15:51:34 +0800 Subject: ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms This patch adds the ACPI match ID for rt5670/5672 codec. So on Intel CherryTrail/Braswell platforms, the codec can be enumerated from ACPI and depends on ACPI to get platform-specific info and power saving. Signed-off-by: Mengdong Lin Reviewed-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 066b583..b0aabd4 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -2503,6 +2504,14 @@ static const struct i2c_device_id rt5670_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5670_acpi_match[] = { + { "10EC5670", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); +#endif + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2691,6 +2700,7 @@ static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5670_acpi_match), }, .probe = rt5670_i2c_probe, .remove = rt5670_i2c_remove, -- cgit v1.1 From 471f208af987a3741757c169c4e2ad984359000b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 14 Nov 2014 14:25:37 +0800 Subject: ASoC: rt5645: two jacks for hp and mic Some OS need headphone and microphone to be separated. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 17 ++++++++--------- sound/soc/codecs/rt5645.h | 5 +++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 57afa12..ef88b50 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2201,8 +2201,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) +static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); int gpio_state, jack_type = 0; @@ -2245,19 +2244,19 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, snd_soc_dapm_sync(&codec->dapm); } - snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET); - + snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); return 0; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - rt5645->jack = jack; - - rt5645_jack_detect(codec, rt5645->jack); + rt5645->hp_jack = hp_jack; + rt5645->mic_jack = mic_jack; + rt5645_jack_detect(codec); return 0; } @@ -2268,7 +2267,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec, rt5645->jack); + rt5645_jack_detect(rt5645->codec); } static irqreturn_t rt5645_irq(int irq, void *data) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 196daf0..c72220a 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2173,7 +2173,8 @@ struct rt5645_priv { struct rt5645_platform_data pdata; struct regmap *regmap; struct i2c_client *i2c; - struct snd_soc_jack *jack; + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; int sysclk; @@ -2188,6 +2189,6 @@ struct rt5645_priv { }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack); + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); #endif /* __RT5645_H__ */ -- cgit v1.1 From 2880fc877971d6c14b0c76ac09744e3ff5b126d5 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 13 Nov 2014 11:18:29 -0800 Subject: ASoC: add TI ts3a227e headset chip driver The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 314 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 364 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..1d12057 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,314 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_component_driver ts3a227e_soc_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..e2acf9c --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif -- cgit v1.1 From 336cfbb05edf7b122ea927dad6c746608723eb25 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 11 Nov 2014 16:36:28 +0530 Subject: ASoC: Intel: mrfld- add ACPI module Add the last ACPI module support which also uses core module like the PCI part Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 5 + sound/soc/intel/sst/Makefile | 4 +- sound/soc/intel/sst/sst.c | 22 ++- sound/soc/intel/sst/sst.h | 1 + sound/soc/intel/sst/sst_acpi.c | 362 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 6 files changed, 391 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c963a5d..a992e85 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,11 @@ config SND_SST_IPC_PCI tristate select SND_SST_IPC +config SND_SST_IPC_ACPI + tristate + select SND_SST_IPC + depends on ACPI + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile index b8aa1d3..fd21726 100644 --- a/sound/soc/intel/sst/Makefile +++ b/sound/soc/intel/sst/Makefile @@ -1,7 +1,7 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o snd-intel-sst-pci-objs += sst_pci.o - +snd-intel-sst-acpi-objs += sst_acpi.o obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o - +obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b97c231..b2b5604 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -181,6 +182,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; return 0; @@ -323,7 +325,7 @@ EXPORT_SYMBOL_GPL(sst_context_init); void sst_context_cleanup(struct intel_sst_drv *ctx) { pm_runtime_get_noresume(ctx->dev); - pm_runtime_forbid(ctx->dev); + pm_runtime_disable(ctx->dev); sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); @@ -371,8 +373,19 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); pm_runtime_use_autosuspend(ctx->dev); - pm_runtime_allow(ctx->dev); - pm_runtime_put_noidle(ctx->dev); + /* + * For acpi devices, the actual physical device state is + * initially active. So change the state to active before + * enabling the pm + */ + if (acpi_disabled) { + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); + } else { + pm_runtime_allow(ctx->dev); + pm_runtime_put_noidle(ctx->dev); + } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); @@ -395,6 +408,9 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + /* save the shim registers because PMC doesn't save state */ + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); + return ret; } diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 2dcbf47..683dc71 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -29,6 +29,7 @@ /* driver names */ #define SST_DRV_NAME "intel_sst_driver" #define SST_MRFLD_PCI_ID 0x119A +#define SST_BYT_ACPI_ID 0x80860F28 #define SST_SUSPEND_DELAY 2000 #define FW_CONTEXT_MEM (64*1024) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c new file mode 100644 index 0000000..2b1c5d9 --- /dev/null +++ b/sound/soc/intel/sst/sst_acpi.c @@ -0,0 +1,362 @@ +/* + * sst_acpi.c - SST (LPE) driver init file for ACPI enumeration. + * + * Copyright (c) 2013, Intel Corporation. + * + * Authors: Ramesh Babu K V + * Authors: Omair Mohammed Abdullah + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../sst-mfld-platform.h" +#include "../sst-dsp.h" +#include "sst.h" + +struct sst_machines { + char codec_id[32]; + char board[32]; + char machine[32]; + void (*machine_quirk)(void); + char firmware[32]; + struct sst_platform_info *pdata; + +}; + +/* LPE viewpoint addresses */ +#define SST_BYT_IRAM_PHY_START 0xff2c0000 +#define SST_BYT_IRAM_PHY_END 0xff2d4000 +#define SST_BYT_DRAM_PHY_START 0xff300000 +#define SST_BYT_DRAM_PHY_END 0xff320000 +#define SST_BYT_IMR_VIRT_START 0xc0000000 /* virtual addr in LPE */ +#define SST_BYT_IMR_VIRT_END 0xc01fffff +#define SST_BYT_SHIM_PHY_ADDR 0xff340000 +#define SST_BYT_MBOX_PHY_ADDR 0xff344000 +#define SST_BYT_DMA0_PHY_ADDR 0xff298000 +#define SST_BYT_DMA1_PHY_ADDR 0xff29c000 +#define SST_BYT_SSP0_PHY_ADDR 0xff2a0000 +#define SST_BYT_SSP2_PHY_ADDR 0xff2a2000 + +#define BYT_FW_MOD_TABLE_OFFSET 0x80000 +#define BYT_FW_MOD_TABLE_SIZE 0x100 +#define BYT_FW_MOD_OFFSET (BYT_FW_MOD_TABLE_OFFSET + BYT_FW_MOD_TABLE_SIZE) + +static const struct sst_info byt_fwparse_info = { + .use_elf = false, + .max_streams = 25, + .iram_start = SST_BYT_IRAM_PHY_START, + .iram_end = SST_BYT_IRAM_PHY_END, + .iram_use = true, + .dram_start = SST_BYT_DRAM_PHY_START, + .dram_end = SST_BYT_DRAM_PHY_END, + .dram_use = true, + .imr_start = SST_BYT_IMR_VIRT_START, + .imr_end = SST_BYT_IMR_VIRT_END, + .imr_use = true, + .mailbox_start = SST_BYT_MBOX_PHY_ADDR, + .num_probes = 0, + .lpe_viewpt_rqd = true, +}; + +static const struct sst_ipc_info byt_ipc_info = { + .ipc_offset = 0, + .mbox_recv_off = 0x400, +}; + +static const struct sst_lib_dnld_info byt_lib_dnld_info = { + .mod_base = SST_BYT_IMR_VIRT_START, + .mod_end = SST_BYT_IMR_VIRT_END, + .mod_table_offset = BYT_FW_MOD_TABLE_OFFSET, + .mod_table_size = BYT_FW_MOD_TABLE_SIZE, + .mod_ddr_dnld = false, +}; + +static const struct sst_res_info byt_rvp_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 0, + .acpi_ddr_index = 2, + .acpi_ipc_irq_index = 5, +}; + +struct sst_platform_info byt_rvp_platform_data = { + .probe_data = &byt_fwparse_info, + .ipc_info = &byt_ipc_info, + .lib_info = &byt_lib_dnld_info, + .res_info = &byt_rvp_res_info, + .platform = "sst-mfld-platform", +}; + +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + struct resource *rsrc; + struct platform_device *pdev = to_platform_device(ctx->dev); + + /* All ACPI resource request here */ + /* Get Shim addr */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base from IFWI"); + return -EIO; + } + dev_info(ctx->dev, "LPE base: %#x size:%#x", (unsigned int) rsrc->start, + (unsigned int)resource_size(rsrc)); + + ctx->iram_base = rsrc->start + ctx->pdata->res_info->iram_offset; + ctx->iram_end = ctx->iram_base + ctx->pdata->res_info->iram_size - 1; + dev_info(ctx->dev, "IRAM base: %#x", ctx->iram_base); + ctx->iram = devm_ioremap_nocache(ctx->dev, ctx->iram_base, + ctx->pdata->res_info->iram_size); + if (!ctx->iram) { + dev_err(ctx->dev, "unable to map IRAM"); + return -EIO; + } + + ctx->dram_base = rsrc->start + ctx->pdata->res_info->dram_offset; + ctx->dram_end = ctx->dram_base + ctx->pdata->res_info->dram_size - 1; + dev_info(ctx->dev, "DRAM base: %#x", ctx->dram_base); + ctx->dram = devm_ioremap_nocache(ctx->dev, ctx->dram_base, + ctx->pdata->res_info->dram_size); + if (!ctx->dram) { + dev_err(ctx->dev, "unable to map DRAM"); + return -EIO; + } + + ctx->shim_phy_add = rsrc->start + ctx->pdata->res_info->shim_offset; + dev_info(ctx->dev, "SHIM base: %#x", ctx->shim_phy_add); + ctx->shim = devm_ioremap_nocache(ctx->dev, ctx->shim_phy_add, + ctx->pdata->res_info->shim_size); + if (!ctx->shim) { + dev_err(ctx->dev, "unable to map SHIM"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->shim_phy_add = ctx->pdata->res_info->shim_phy_addr; + + /* Get mailbox addr */ + ctx->mailbox_add = rsrc->start + ctx->pdata->res_info->mbox_offset; + dev_info(ctx->dev, "Mailbox base: %#x", ctx->mailbox_add); + ctx->mailbox = devm_ioremap_nocache(ctx->dev, ctx->mailbox_add, + ctx->pdata->res_info->mbox_size); + if (!ctx->mailbox) { + dev_err(ctx->dev, "unable to map mailbox"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->mailbox_add = ctx->info.mailbox_start; + + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_ddr_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid DDR base from IFWI"); + return -EIO; + } + ctx->ddr_base = rsrc->start; + ctx->ddr_end = rsrc->end; + dev_info(ctx->dev, "DDR base: %#x", ctx->ddr_base); + ctx->ddr = devm_ioremap_nocache(ctx->dev, ctx->ddr_base, + resource_size(rsrc)); + if (!ctx->ddr) { + dev_err(ctx->dev, "unable to map DDR"); + return -EIO; + } + + /* Find the IRQ */ + ctx->irq_num = platform_get_irq(pdev, + ctx->pdata->res_info->acpi_ipc_irq_index); + return 0; +} + +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_machines *sst_acpi_find_machine( + struct sst_machines *machines) +{ + struct sst_machines *mach; + bool found = false; + + for (mach = machines; mach->codec_id; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + +int sst_acpi_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + int ret = 0; + struct intel_sst_drv *ctx; + const struct acpi_device_id *id; + struct sst_machines *mach; + struct platform_device *mdev; + struct platform_device *plat_dev; + unsigned int dev_id; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + dev_dbg(dev, "for %s", id->id); + + mach = (struct sst_machines *)id->driver_data; + mach = sst_acpi_find_machine(mach); + if (mach == NULL) { + dev_err(dev, "No matching machine driver found\n"); + return -ENODEV; + } + + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + dev_dbg(dev, "ACPI device id: %x\n", dev_id); + + plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); + if (plat_dev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); + return -ENODEV; + } + + /* Create platform device for sst machine driver */ + mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); + if (mdev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->machine); + return -ENODEV; + } + + ret = sst_alloc_drv_context(&ctx, dev, dev_id); + if (ret < 0) + return ret; + + /* Fill sst platform data */ + ctx->pdata = mach->pdata; + strcpy(ctx->firmware_name, mach->firmware); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + + /* need to save shim registers in BYT */ + ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), + GFP_KERNEL); + if (!ctx->shim_regs64) { + return -ENOMEM; + goto do_sst_cleanup; + } + + sst_configure_runtime_pm(ctx); + platform_set_drvdata(pdev, ctx); + return ret; + +do_sst_cleanup: + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + dev_err(ctx->dev, "failed with %d\n", ret); + return ret; +} + +/** +* intel_sst_remove - remove function +* +* @pdev: platform device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +int sst_acpi_remove(struct platform_device *pdev) +{ + struct intel_sst_drv *ctx; + + ctx = platform_get_drvdata(pdev); + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct sst_machines sst_acpi_bytcr[] = { + {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + &byt_rvp_platform_data }, + {}, +}; + +static const struct acpi_device_id sst_acpi_ids[] = { + { "80860F28", (unsigned long)&sst_acpi_bytcr}, + { }, +}; + +static struct platform_driver sst_acpi_driver = { + .driver = { + .name = "intel_sst_acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_ids), + .pm = &intel_sst_pm, + }, + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, +}; + +module_platform_driver(sst_acpi_driver); + +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine ACPI Driver"); +MODULE_AUTHOR("Ramesh Babu K V"); +MODULE_AUTHOR("Omair Mohammed Abdullah"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 9a5df19..4b77208 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -117,6 +117,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: val = sst_shim_read64(sst->shim, addr); break; } @@ -128,6 +129,7 @@ void write_shim_data(struct intel_sst_drv *sst, int addr, { switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst_shim_write64(sst->shim, addr, (u64) data); break; } -- cgit v1.1 From 044b724ada4448174f3f7510b791df0bdcb834ee Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 12 Nov 2014 19:54:30 +0800 Subject: ASoC: rt5670: make bias level more reasonable This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index b0aabd4..5e54ac9 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2310,6 +2310,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2331,16 +2333,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- cgit v1.1 From 387417b56295ef93d7cb38e1721826c85dfe897c Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Fri, 14 Nov 2014 16:09:05 +0530 Subject: ALSA: ice1712: remove unneeded return statement the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/ice1712/hoontech.c | 6 +----- sound/pci/ice1712/ice1712.c | 19 ++++++------------- sound/pci/ice1712/ice1724.c | 7 ++----- sound/pci/ice1712/juli.c | 5 +---- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 5 +---- 6 files changed, 12 insertions(+), 34 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..a40001c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -309,11 +309,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) return err; /* ak4524 controls */ - err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return snd_ice1712_akm4xxx_build_controls(ice); } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..5975334 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1295,10 +1295,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd return err; } - err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return snd_ice1712_build_pro_mixer(ice); } /* @@ -1545,10 +1542,9 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) dev_warn(ice->card->dev, "cannot initialize ac97 for consumer, skipped\n"); else { - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, + ice)); } } @@ -2497,11 +2493,8 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice)); if (err < 0) return err; - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..ea53167 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2497,11 +2497,8 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) return err; } - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..a1536c1 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -475,11 +475,8 @@ static int juli_add_controls(struct snd_ice1712 *ice) return err; /* only capture SPDIF over AK4114 */ - err = snd_ak4114_build(spec->ak4114, NULL, + return snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..6f55e02 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -833,11 +833,8 @@ static int qtet_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; /* only capture SPDIF over AK4113 */ - err = snd_ak4113_build(spec->ak4113, + return snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- cgit v1.1 From b393df0145e271724fee10f93c023662f8557bb9 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Fri, 14 Nov 2014 16:09:07 +0530 Subject: ALSA: ice1712: remove unused variable buf_size was initialized with snd_pcm_lib_buffer_bytes, but never used. and so it is safe to be deleted. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5975334..6525191 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -620,10 +620,9 @@ static int snd_ice1712_playback_ds_prepare(struct snd_pcm_substream *substream) { struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - u32 period_size, buf_size, rate, tmp, chn; + u32 period_size, rate, tmp, chn; period_size = snd_pcm_lib_period_bytes(substream) - 1; - buf_size = snd_pcm_lib_buffer_bytes(substream) - 1; tmp = 0x0064; if (snd_pcm_format_width(runtime->format) == 16) tmp &= ~0x04; -- cgit v1.1 From b8eca77e54525c818f35f51afb64fc13205443a3 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Fri, 14 Nov 2014 18:12:21 +0530 Subject: ALSA: ice1712: consider error value earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee Signed-off-by: Takashi Iwai --- sound/pci/ice1712/revo.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..1d81ae6 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -494,11 +494,13 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return 0; } static int revo_init(struct snd_ice1712 *ice) -- cgit v1.1 From cdcd7f7287532131d2075dd45f15aaf39dcfe983 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 14 Nov 2014 15:40:45 +0000 Subject: ASoC: wm_adsp: Use vmalloc to allocate firmware download buffer Use vmalloc to allocate the buffer for firmware/coefficient download and rely on the SPI core to split this up into DMA-able chunks. This should give better performance and means we no longer need to manually split the download into page size chunks to avoid allocating overly large continuous memory regions. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 56 ++++++++++++++++++---------------------------- 1 file changed, 22 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f412a99..0a08ef5e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -169,11 +170,12 @@ static struct wm_adsp_buf *wm_adsp_buf_alloc(const void *src, size_t len, if (buf == NULL) return NULL; - buf->buf = kmemdup(src, len, GFP_KERNEL | GFP_DMA); + buf->buf = vmalloc(len); if (!buf->buf) { - kfree(buf); + vfree(buf); return NULL; } + memcpy(buf->buf, src, len); if (list) list_add_tail(&buf->list, list); @@ -188,7 +190,7 @@ static void wm_adsp_buf_free(struct list_head *list) struct wm_adsp_buf, list); list_del(&buf->list); - kfree(buf->buf); + vfree(buf->buf); kfree(buf); } } @@ -684,38 +686,24 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - size_t to_write = PAGE_SIZE; - size_t remain = le32_to_cpu(region->len); - const u8 *data = region->data; - - while (remain > 0) { - if (remain < PAGE_SIZE) - to_write = remain; - - buf = wm_adsp_buf_alloc(data, - to_write, - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } - - ret = regmap_raw_write_async(regmap, reg, - buf->buf, - to_write); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", - file, regions, - to_write, offset, - region_name, ret); - goto out_fw; - } + buf = wm_adsp_buf_alloc(region->data, + le32_to_cpu(region->len), + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - data += to_write; - reg += to_write / 2; - remain -= to_write; + ret = regmap_raw_write_async(regmap, reg, buf->buf, + le32_to_cpu(region->len)); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + file, regions, + le32_to_cpu(region->len), offset, + region_name, ret); + goto out_fw; } } -- cgit v1.1 From 6fdaac1c1ab4fee1619145487c5aaf1bd44acc7b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Nov 2014 09:37:34 +0100 Subject: ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index ce3cdca..b67480f 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); const char *clk; @@ -236,7 +236,7 @@ static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; -- cgit v1.1 From de172051af78883a4a2e7897e7af58ba49353b99 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Nov 2014 09:37:35 +0100 Subject: ASoC: adau1373: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 7c784ad..783dcb5 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -551,7 +551,7 @@ static const struct snd_kcontrol_new adau1373_drc_controls[] = { static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -823,7 +823,7 @@ static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = { static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dai; const char *clk; @@ -844,7 +844,7 @@ static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, static int adau1373_check_src(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dai; -- cgit v1.1 From d69db7f7cd57fdfc6ac64c4c8679eb7b80c84fc7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Nov 2014 09:37:36 +0100 Subject: ASoC: adau17x1: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1761.c | 3 ++- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 3 ++- 3 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5518ebd..3dddb28 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -255,7 +255,8 @@ static const struct snd_kcontrol_new adau1761_input_mux_control = static int adau1761_dejitter_fixup(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct adau *adau = snd_soc_codec_get_drvdata(codec); /* After any power changes have been made the dejitter circuit * has to be reinitialized. */ diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index e9fc00f..aa6a37c 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -174,7 +174,7 @@ static const struct snd_kcontrol_new adau1781_mono_mixer_controls[] = { static int adau1781_dejitter_fixup(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct adau *adau = snd_soc_codec_get_drvdata(codec); /* After any power changes have been made the dejitter circuit diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 3e16c1c..427ad77 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -61,7 +61,8 @@ static const struct snd_kcontrol_new adau17x1_controls[] = { static int adau17x1_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct adau *adau = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct adau *adau = snd_soc_codec_get_drvdata(codec); int ret; if (SND_SOC_DAPM_EVENT_ON(event)) { -- cgit v1.1 From eb826a35d2578106cf6fbfb2a83eedd1c0c2c415 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 17 Nov 2014 14:36:40 +0800 Subject: ASoC: Intel: add missing ACPI device table The ACPI device table will generate the driver module alias for Intel audio devices enumerated from ACPI. Signed-off-by: Mengdong Lin Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2b1c5d9..b261821 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -342,6 +342,8 @@ static const struct acpi_device_id sst_acpi_ids[] = { { }, }; +MODULE_DEVICE_TABLE(acpi, sst_acpi_ids); + static struct platform_driver sst_acpi_driver = { .driver = { .name = "intel_sst_acpi", -- cgit v1.1 From 6d3efa40790ad1286cfa032df6d3c9a2748a695e Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Sat, 15 Nov 2014 22:51:46 +0300 Subject: ASoC: pxa: prepare/unprepare clocks in pxa-ssp Change clk_enable/disable() calls to clk_prepare_enable() and clk_disable_unrepapre(). Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a8e0974..cbba063 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -97,7 +97,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); pxa_ssp_disable(ssp); } @@ -121,7 +121,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); } kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); @@ -136,7 +136,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); @@ -144,7 +144,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->psp = __raw_readl(ssp->mmio_base + SSPSP); pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -154,7 +154,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (cpu_dai->active) pxa_ssp_enable(ssp); else - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -256,11 +256,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (ssp->type != PXA3xx_SSP) - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); if (ssp->type != PXA3xx_SSP) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); return 0; } -- cgit v1.1 From f0acd28c87ad2a5d1b40403fdd5defda2961b2a1 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 10:44:33 +0100 Subject: ALSA: hda: Deletion of unnecessary checks before two function calls The functions kfree() and release_firmware() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_intel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ca98f52..b2d5899 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -834,8 +834,7 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - if (bus->unsol) - kfree(bus->unsol); + kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9ab1e63..91fa959 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1130,8 +1130,7 @@ static int azx_free(struct azx *chip) pci_disable_device(chip->pci); kfree(chip->azx_dev); #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (chip->fw) - release_firmware(chip->fw); + release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); -- cgit v1.1 From ae1b22658e6d3ebc6af07a225c221d84fe8cb91f Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 11:28:02 +0100 Subject: ALSA: ice17xx: Deletion of unnecessary checks before the function call "snd_ac97_resume" The snd_ac97_resume() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/ice1712/ice1724.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 6525191..b039b46 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2878,8 +2878,7 @@ static int snd_ice1712_resume(struct device *dev) outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT)); outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ea53167..d73da15 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2875,8 +2875,7 @@ static int snd_vt1724_resume(struct device *dev) outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; -- cgit v1.1 From 6da95e1ea8e2492530eac9c51b293226e3f4ce94 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 12:42:16 +0100 Subject: ALSA: lola: Deletion of an unnecessary check before the function call "vfree" The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_mixer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..e7fe15d 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -108,8 +108,7 @@ int lola_init_pins(struct lola *chip, int dir, int *nidp) void lola_free_mixer(struct lola *chip) { - if (chip->mixer.array_saved) - vfree(chip->mixer.array_saved); + vfree(chip->mixer.array_saved); } int lola_init_mixer_widget(struct lola *chip, int nid) -- cgit v1.1 From c283661018e347bc72633969411974df8ec2ac92 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 13:04:14 +0100 Subject: ALSA: hdsp: Deletion of an unnecessary check before the function call "release_firmware" The release_firmware() function tests whether its argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 2eb8baf..cf5a6c8 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5307,8 +5307,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) snd_hdsp_free_buffers(hdsp); - if (hdsp->firmware) - release_firmware(hdsp->firmware); + release_firmware(hdsp->firmware); vfree(hdsp->fw_uploaded); if (hdsp->iobase) -- cgit v1.1 From 1ea7a568c63a4735872fc091efbd22d2e4d9c972 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 13:35:54 +0100 Subject: ALSA: powermac: Deletion of an unnecessary check before the function call "pci_dev_put" The pci_dev_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/ppc/pmac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 8a431bc..5a13b22 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -887,8 +887,7 @@ static int snd_pmac_free(struct snd_pmac *chip) } } - if (chip->pdev) - pci_dev_put(chip->pdev); + pci_dev_put(chip->pdev); of_node_put(chip->node); kfree(chip); return 0; -- cgit v1.1 From 187024b36c635bd454c1b1587b58c9439d3a46ad Mon Sep 17 00:00:00 2001 From: Krishna Mohan Dani Date: Mon, 17 Nov 2014 19:26:29 +0530 Subject: ASoC: rt5631: Fixing compilation warning when DT is disabled Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn Signed-off-by: Krishna Mohan Dani Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..9425545 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,12 +1691,14 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +#ifdef CONFIG_OF static struct of_device_id rt5631_i2c_dt_ids[] = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } }; MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids); +#endif static const struct regmap_config rt5631_regmap_config = { .reg_bits = 8, -- cgit v1.1 From 86ae04b174152147052adec7b95dba0c9cd7dff0 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 17 Nov 2014 10:18:11 +0800 Subject: ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function Modify the default value of the MX-8E[4] to 1 for ASRC function. It could prevent the pop noise with ASRC function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..9ae2e84 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- cgit v1.1 From 7e35ac81598c9a98dee4faa77b988c4ea919d1cd Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 17 Nov 2014 00:16:19 -0200 Subject: ASoC: fsl_ssi: Remove comment about SSI running only in slave mode Current driver can also run in I2S master mode, so remove the old comment. Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e695517..bc19849 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,8 +67,6 @@ /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * - * This driver currently only supports the SSI running in I2S slave mode. - * * The SSI has a limitation in that the samples must be in the same byte * order as the host CPU. This is because when multiple bytes are written * to the STX register, the bytes and bits must be written in the same -- cgit v1.1 From bb66f2dc197d9cf1daaa82609302204d71c70389 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Mon, 17 Nov 2014 14:05:27 +0100 Subject: ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree" The kfree() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- cgit v1.1 From bb29a93b38610d2adc6ead40b75e1a1991617550 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 12 Nov 2014 00:52:23 +0900 Subject: ASoC: jack: Fix warning while make htmldocs caused by soc-jack.c This patch fix following errors while "make htmldocs" on linux-next-20141110. Warning(.//sound/soc/soc-jack.c:126): No description found for parameter 'zones' Warning(.//sound/soc/soc-jack.c:126): Excess function parameter 'zone' description in 'snd_soc_jack_add_zones' Signed-off-by: Masanari Iida Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea..ef1d42d 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- cgit v1.1 From 35480e3536cdab1ee1976675e798f16d707f5356 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:43 +0100 Subject: ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device The mpc5200_psc_ac97 driver puts a snd_ac97 device on the stack in the driver probe function, initializes the private data member of the device and the never uses the device again. It should be safe to remove it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 24eafa2..640801a 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -282,7 +282,6 @@ static const struct snd_soc_component_driver psc_ac97_component = { static int psc_ac97_of_probe(struct platform_device *op) { int rc; - struct snd_ac97 ac97; struct mpc52xx_psc __iomem *regs; rc = mpc5200_audio_dma_create(op); @@ -304,7 +303,6 @@ static int psc_ac97_of_probe(struct platform_device *op) psc_dma = dev_get_drvdata(&op->dev); regs = psc_dma->psc_regs; - ac97.private_data = psc_dma; psc_dma->imr = 0; out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); -- cgit v1.1 From 65c72efd1ea370f0311a5d89754996fff9fc0747 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:44 +0100 Subject: ASoC: mpc5200_dma: Don't overwrite ac97 device private_data The mpc5200_dma overwrites the private_data field of the CODEC's AC'97 device with the DMA drivers private data, but never actually reads it again. Given that the private_data field is supposed to be owned by the AC'97 driver, overwriting it may cause undefined behavior. This patch removes the code that overwrites the field from the driver. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f2b5d75..0b82e20 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -327,9 +327,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) goto capture_alloc_err; } - if (rtd->codec->ac97) - rtd->codec->ac97->private_data = psc_dma; - return 0; capture_alloc_err: -- cgit v1.1 From 70f3af3ca15affaef3d026a5aa6e44c4627ea6c7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:45 +0100 Subject: ASoC: Properly handle AC'97 device lifetime management The memory that a struct device is contained in must not be freed except from within the device's release callback. The ASoC code currently does not adhere to this rule for the AC'97 device. This patch fixes it by moving the freeing of the AC'97 to the release callback and splitting up the registration and unregistration of the device into separate steps for getting/putting the reference to the device and adding/removing it to the device hierarchy. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..7084c6f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -504,13 +504,10 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) { if (codec->ac97->dev.bus) - device_unregister(&codec->ac97->dev); + device_del(&codec->ac97->dev); return 0; } -/* stop no dev release warning */ -static void soc_ac97_device_release(struct device *dev){} - /* register ac97 codec to bus */ static int soc_ac97_dev_register(struct snd_soc_codec *codec) { @@ -518,12 +515,11 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.bus = &ac97_bus_type; codec->ac97->dev.parent = codec->component.card->dev; - codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", codec->component.card->snd_card->number, 0, codec->component.name); - err = device_register(&codec->ac97->dev); + err = device_add(&codec->ac97->dev); if (err < 0) { dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); codec->ac97->dev.bus = NULL; @@ -1948,6 +1944,11 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; +static void soc_ac97_device_release(struct device *dev) +{ + kfree(to_ac97_t(dev)); +} + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1972,12 +1973,14 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, codec->ac97->bus->ops = ops; codec->ac97->num = num; + codec->ac97->dev.release = soc_ac97_device_release; /* * Mark the AC97 device to be created by us. This way we ensure that the * device will be registered with the device subsystem later on. */ codec->ac97_created = 1; + device_initialize(&codec->ac97->dev); return 0; } @@ -2152,7 +2155,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) soc_unregister_ac97_codec(codec); #endif kfree(codec->ac97->bus); - kfree(codec->ac97); + codec->ac97->bus = NULL; + put_device(&codec->ac97->dev); codec->ac97 = NULL; codec->ac97_created = 0; } -- cgit v1.1 From 336b8423e285174ebecf02a743d69913b83bbc48 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:46 +0100 Subject: ASoC: Move AC'97 support to its own file Currently the AC'97 support is splattered all throughout soc-core.c. Some parts are #ifdef'd some parts are not. This patch moves the AC'97 support to its own file, this should make the code a bit more clearer and also makes it possible to easily not compile it into the kernel when not needed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 28 +++- sound/soc/Makefile | 4 + sound/soc/soc-ac97.c | 382 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 352 +---------------------------------------------- 4 files changed, 416 insertions(+), 350 deletions(-) create mode 100644 sound/soc/soc-ac97.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..adef34f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -366,8 +366,6 @@ struct snd_soc_jack_gpio; typedef int (*hw_write_t)(void *,const char* ,int); -extern struct snd_ac97_bus_ops *soc_ac97_ops; - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, @@ -500,6 +498,7 @@ int snd_soc_update_bits_locked(struct snd_soc_codec *codec, int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); +#ifdef CONFIG_SND_SOC_AC97_BUS int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); @@ -508,6 +507,31 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); +extern struct snd_ac97_bus_ops *soc_ac97_ops; + +int snd_soc_ac97_register_dai_links(struct snd_soc_card *card); +void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd); +#else + +static inline int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) +{ + return 0; +} + +static inline void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd) {} + +static inline int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, + struct platform_device *pdev) +{ + return 0; +} + +static inline int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + return 0; +} +#endif + /* *Controls */ diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..0fded1b 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -5,6 +5,10 @@ ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o endif +ifneq ($(CONFIG_SND_SOC_AC97_BUS),) +snd-soc-core-objs += soc-ac97.o +endif + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += generic/ diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c new file mode 100644 index 0000000..da7b031 --- /dev/null +++ b/sound/soc/soc-ac97.c @@ -0,0 +1,382 @@ +/* + * soc-ac97.c -- ALSA SoC Audio Layer AC97 support + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. + * + * Author: Liam Girdwood + * with code, comments and ideas from :- + * Richard Purdie + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +struct snd_ac97_reset_cfg { + struct pinctrl *pctl; + struct pinctrl_state *pstate_reset; + struct pinctrl_state *pstate_warm_reset; + struct pinctrl_state *pstate_run; + int gpio_sdata; + int gpio_sync; + int gpio_reset; +}; + +/* unregister ac97 codec */ +static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) +{ + if (codec->ac97->dev.bus) + device_del(&codec->ac97->dev); + return 0; +} + +/* register ac97 codec to bus */ +static int soc_ac97_dev_register(struct snd_soc_codec *codec) +{ + int err; + + codec->ac97->dev.bus = &ac97_bus_type; + codec->ac97->dev.parent = codec->component.card->dev; + + dev_set_name(&codec->ac97->dev, "%d-%d:%s", + codec->component.card->snd_card->number, 0, + codec->component.name); + err = device_add(&codec->ac97->dev); + if (err < 0) { + dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); + codec->ac97->dev.bus = NULL; + return err; + } + return 0; +} + +static int soc_register_ac97_codec(struct snd_soc_codec *codec, + struct snd_soc_dai *codec_dai) +{ + int ret; + + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (codec_dai->driver->ac97_control && !codec->ac97_registered) { + /* + * It is possible that the AC97 device is already registered to + * the device subsystem. This happens when the device is created + * via snd_ac97_mixer(). Currently only SoC codec that does so + * is the generic AC97 glue but others migh emerge. + * + * In those cases we don't try to register the device again. + */ + if (!codec->ac97_created) + return 0; + + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + dev_err(codec->dev, + "ASoC: AC97 device register failed: %d\n", ret); + return ret; + } + + codec->ac97_registered = 1; + } + return 0; +} + +static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) +{ + if (codec->ac97_registered) { + soc_ac97_dev_unregister(codec); + codec->ac97_registered = 0; + } +} + +static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +{ + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + ret = soc_register_ac97_codec(codec_dai->codec, codec_dai); + if (ret) { + while (--i >= 0) + soc_unregister_ac97_codec(codec_dai->codec); + return ret; + } + } + + return 0; +} + +static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +{ + int i; + + for (i = 0; i < rtd->num_codecs; i++) + soc_unregister_ac97_codec(rtd->codec_dais[i]->codec); +} + +static void soc_ac97_device_release(struct device *dev) +{ + kfree(to_ac97_t(dev)); +} + +/** + * snd_soc_new_ac97_codec - initailise AC97 device + * @codec: audio codec + * @ops: AC97 bus operations + * @num: AC97 codec number + * + * Initialises AC97 codec resources for use by ad-hoc devices only. + */ +int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, + struct snd_ac97_bus_ops *ops, int num) +{ + codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); + if (codec->ac97 == NULL) + return -ENOMEM; + + codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); + if (codec->ac97->bus == NULL) { + kfree(codec->ac97); + codec->ac97 = NULL; + return -ENOMEM; + } + + codec->ac97->bus->ops = ops; + codec->ac97->num = num; + codec->ac97->dev.release = soc_ac97_device_release; + + /* + * Mark the AC97 device to be created by us. This way we ensure that the + * device will be registered with the device subsystem later on. + */ + codec->ac97_created = 1; + device_initialize(&codec->ac97->dev); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); + +/** + * snd_soc_free_ac97_codec - free AC97 codec device + * @codec: audio codec + * + * Frees AC97 codec device resources. + */ +void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) +{ + soc_unregister_ac97_codec(codec); + kfree(codec->ac97->bus); + codec->ac97->bus = NULL; + put_device(&codec->ac97->dev); + codec->ac97 = NULL; + codec->ac97_created = 0; +} +EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); + +static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; + +static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static void snd_soc_ac97_reset(struct snd_ac97 *ac97) +{ + struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0); + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0); + + udelay(10); + + gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1); + + pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); + msleep(2); +} + +static int snd_soc_ac97_parse_pinctl(struct device *dev, + struct snd_ac97_reset_cfg *cfg) +{ + struct pinctrl *p; + struct pinctrl_state *state; + int gpio; + int ret; + + p = devm_pinctrl_get(dev); + if (IS_ERR(p)) { + dev_err(dev, "Failed to get pinctrl\n"); + return PTR_ERR(p); + } + cfg->pctl = p; + + state = pinctrl_lookup_state(p, "ac97-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-reset\n"); + return PTR_ERR(state); + } + cfg->pstate_reset = state; + + state = pinctrl_lookup_state(p, "ac97-warm-reset"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n"); + return PTR_ERR(state); + } + cfg->pstate_warm_reset = state; + + state = pinctrl_lookup_state(p, "ac97-running"); + if (IS_ERR(state)) { + dev_err(dev, "Can't find pinctrl state ac97-running\n"); + return PTR_ERR(state); + } + cfg->pstate_run = state; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sync gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sync"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sync gpio\n"); + return ret; + } + cfg->gpio_sync = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link sdata"); + if (ret) { + dev_err(dev, "Failed requesting ac97-sdata gpio\n"); + return ret; + } + cfg->gpio_sdata = gpio; + + gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + if (gpio < 0) { + dev_err(dev, "Can't find ac97-reset gpio\n"); + return gpio; + } + ret = devm_gpio_request(dev, gpio, "AC97 link reset"); + if (ret) { + dev_err(dev, "Failed requesting ac97-reset gpio\n"); + return ret; + } + cfg->gpio_reset = gpio; + + return 0; +} + +struct snd_ac97_bus_ops *soc_ac97_ops; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) +{ + if (ops == soc_ac97_ops) + return 0; + + if (soc_ac97_ops && ops) + return -EBUSY; + + soc_ac97_ops = ops; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); + +/** + * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * + * This function sets the reset and warm_reset properties of ops and parses + * the device node of pdev to get pinctrl states and gpio numbers to use. + */ +int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, + struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct snd_ac97_reset_cfg cfg; + int ret; + + ret = snd_soc_ac97_parse_pinctl(dev, &cfg); + if (ret) + return ret; + + ret = snd_soc_set_ac97_ops(ops); + if (ret) + return ret; + + ops->warm_reset = snd_soc_ac97_warm_reset; + ops->reset = snd_soc_ac97_reset; + + snd_ac97_rst_cfg = cfg; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); + +int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) +{ + int i; + int ret; + + /* register any AC97 codecs */ + for (i = 0; i < card->num_rtd; i++) { + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) + goto err; + } + + return 0; +err: + dev_err(card->dev, + "ASoC: failed to register AC97: %d\n", ret); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + return ret; +} + +void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd) +{ + unsigned int i; + + /* add platform data for AC97 devices */ + for (i = 0; i < rtd->num_codecs; i++) { + if (rtd->codec_dais[i]->driver->ac97_control) + snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97, + rtd->cpu_dai->ac97_pdata); + } +} diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7084c6f..026722f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -34,9 +34,6 @@ #include #include #include -#include -#include -#include #include #include #include @@ -69,16 +66,6 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); -struct snd_ac97_reset_cfg { - struct pinctrl *pctl; - struct pinctrl_state *pstate_reset; - struct pinctrl_state *pstate_warm_reset; - struct pinctrl_state *pstate_run; - int gpio_sdata; - int gpio_sync; - int gpio_reset; -}; - /* returns the minimum number of bytes needed to represent * a particular given value */ static int min_bytes_needed(unsigned long val) @@ -499,36 +486,6 @@ struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); -#ifdef CONFIG_SND_SOC_AC97_BUS -/* unregister ac97 codec */ -static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) -{ - if (codec->ac97->dev.bus) - device_del(&codec->ac97->dev); - return 0; -} - -/* register ac97 codec to bus */ -static int soc_ac97_dev_register(struct snd_soc_codec *codec) -{ - int err; - - codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = codec->component.card->dev; - - dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->component.card->snd_card->number, 0, - codec->component.name); - err = device_add(&codec->ac97->dev); - if (err < 0) { - dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); - codec->ac97->dev.bus = NULL; - return err; - } - return 0; -} -#endif - static void codec2codec_close_delayed_work(struct work_struct *work) { /* Currently nothing to do for c2c links @@ -1418,84 +1375,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } - /* add platform data for AC97 devices */ - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->driver->ac97_control) - snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97, - rtd->cpu_dai->ac97_pdata); - } - - return 0; -} - -#ifdef CONFIG_SND_SOC_AC97_BUS -static int soc_register_ac97_codec(struct snd_soc_codec *codec, - struct snd_soc_dai *codec_dai) -{ - int ret; - - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (codec_dai->driver->ac97_control && !codec->ac97_registered) { - /* - * It is possible that the AC97 device is already registered to - * the device subsystem. This happens when the device is created - * via snd_ac97_mixer(). Currently only SoC codec that does so - * is the generic AC97 glue but others migh emerge. - * - * In those cases we don't try to register the device again. - */ - if (!codec->ac97_created) - return 0; - - ret = soc_ac97_dev_register(codec); - if (ret < 0) { - dev_err(codec->dev, - "ASoC: AC97 device register failed: %d\n", ret); - return ret; - } - - codec->ac97_registered = 1; - } - return 0; -} - -static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) -{ - if (codec->ac97_registered) { - soc_ac97_dev_unregister(codec); - codec->ac97_registered = 0; - } -} - -static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - int i, ret; - - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - - ret = soc_register_ac97_codec(codec_dai->codec, codec_dai); - if (ret) { - while (--i >= 0) - soc_unregister_ac97_codec(codec_dai->codec); - return ret; - } - } + snd_soc_ac97_add_pdata(rtd); return 0; } -static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - int i; - - for (i = 0; i < rtd->num_codecs; i++) - soc_unregister_ac97_codec(rtd->codec_dais[i]->codec); -} -#endif - static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; @@ -1789,19 +1673,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto probe_aux_dev_err; } -#ifdef CONFIG_SND_SOC_AC97_BUS - /* register any AC97 codecs */ - for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - dev_err(card->dev, - "ASoC: failed to register AC97: %d\n", ret); - while (--i >= 0) - soc_unregister_ac97_dai_link(&card->rtd[i]); - goto probe_aux_dev_err; - } - } -#endif + ret = snd_soc_ac97_register_dai_links(card); + if (ret < 0) + goto probe_aux_dev_err; card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); @@ -1944,224 +1818,6 @@ static struct platform_driver soc_driver = { .remove = soc_remove, }; -static void soc_ac97_device_release(struct device *dev) -{ - kfree(to_ac97_t(dev)); -} - -/** - * snd_soc_new_ac97_codec - initailise AC97 device - * @codec: audio codec - * @ops: AC97 bus operations - * @num: AC97 codec number - * - * Initialises AC97 codec resources for use by ad-hoc devices only. - */ -int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, - struct snd_ac97_bus_ops *ops, int num) -{ - codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); - if (codec->ac97 == NULL) - return -ENOMEM; - - codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); - if (codec->ac97->bus == NULL) { - kfree(codec->ac97); - codec->ac97 = NULL; - return -ENOMEM; - } - - codec->ac97->bus->ops = ops; - codec->ac97->num = num; - codec->ac97->dev.release = soc_ac97_device_release; - - /* - * Mark the AC97 device to be created by us. This way we ensure that the - * device will be registered with the device subsystem later on. - */ - codec->ac97_created = 1; - device_initialize(&codec->ac97->dev); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); - -static struct snd_ac97_reset_cfg snd_ac97_rst_cfg; - -static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; - - pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset); - - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1); - - udelay(10); - - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); - - pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); - msleep(2); -} - -static void snd_soc_ac97_reset(struct snd_ac97 *ac97) -{ - struct pinctrl *pctl = snd_ac97_rst_cfg.pctl; - - pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset); - - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); - gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0); - gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0); - - udelay(10); - - gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1); - - pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); - msleep(2); -} - -static int snd_soc_ac97_parse_pinctl(struct device *dev, - struct snd_ac97_reset_cfg *cfg) -{ - struct pinctrl *p; - struct pinctrl_state *state; - int gpio; - int ret; - - p = devm_pinctrl_get(dev); - if (IS_ERR(p)) { - dev_err(dev, "Failed to get pinctrl\n"); - return PTR_ERR(p); - } - cfg->pctl = p; - - state = pinctrl_lookup_state(p, "ac97-reset"); - if (IS_ERR(state)) { - dev_err(dev, "Can't find pinctrl state ac97-reset\n"); - return PTR_ERR(state); - } - cfg->pstate_reset = state; - - state = pinctrl_lookup_state(p, "ac97-warm-reset"); - if (IS_ERR(state)) { - dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n"); - return PTR_ERR(state); - } - cfg->pstate_warm_reset = state; - - state = pinctrl_lookup_state(p, "ac97-running"); - if (IS_ERR(state)) { - dev_err(dev, "Can't find pinctrl state ac97-running\n"); - return PTR_ERR(state); - } - cfg->pstate_run = state; - - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-sync gpio\n"); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link sync"); - if (ret) { - dev_err(dev, "Failed requesting ac97-sync gpio\n"); - return ret; - } - cfg->gpio_sync = gpio; - - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link sdata"); - if (ret) { - dev_err(dev, "Failed requesting ac97-sdata gpio\n"); - return ret; - } - cfg->gpio_sdata = gpio; - - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-reset gpio\n"); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link reset"); - if (ret) { - dev_err(dev, "Failed requesting ac97-reset gpio\n"); - return ret; - } - cfg->gpio_reset = gpio; - - return 0; -} - -struct snd_ac97_bus_ops *soc_ac97_ops; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) -{ - if (ops == soc_ac97_ops) - return 0; - - if (soc_ac97_ops && ops) - return -EBUSY; - - soc_ac97_ops = ops; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); - -/** - * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions - * - * This function sets the reset and warm_reset properties of ops and parses - * the device node of pdev to get pinctrl states and gpio numbers to use. - */ -int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, - struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct snd_ac97_reset_cfg cfg; - int ret; - - ret = snd_soc_ac97_parse_pinctl(dev, &cfg); - if (ret) - return ret; - - ret = snd_soc_set_ac97_ops(ops); - if (ret) - return ret; - - ops->warm_reset = snd_soc_ac97_warm_reset; - ops->reset = snd_soc_ac97_reset; - - snd_ac97_rst_cfg = cfg; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); - -/** - * snd_soc_free_ac97_codec - free AC97 codec device - * @codec: audio codec - * - * Frees AC97 codec device resources. - */ -void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) -{ -#ifdef CONFIG_SND_SOC_AC97_BUS - soc_unregister_ac97_codec(codec); -#endif - kfree(codec->ac97->bus); - codec->ac97->bus = NULL; - put_device(&codec->ac97->dev); - codec->ac97 = NULL; - codec->ac97_created = 0; -} -EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); - /** * snd_soc_cnew - create new control * @_template: control template -- cgit v1.1 From eda1a701fd9589b6ed15b109558bd4f6202e3829 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:47 +0100 Subject: ASoC: ac97: Use static ac97_bus We always pass soc_ac97_ops to snd_soc_new_ac97_codec(). So instead of allocating a snd_ac97_bus in snd_soc_new_ac97_codec() just use a static one that gets initialized when snd_soc_set_ac97_ops() is called. Also drop the device number parameter from snd_soc_new_ac97_codec(). We currently only support one device per bus and all drivers pass 0 for the device number. And if we should ever support multiple devices per bus it wouldn't be up to individual AC'97 device drivers to pick their number, but rather either the AC'97 adapter driver or the core code will assign them. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/soc-ac97.c | 22 ++++++++-------------- 7 files changed, 14 insertions(+), 21 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index adef34f..44b3ce5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -499,8 +499,7 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); #ifdef CONFIG_SND_SOC_AC97_BUS -int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, - struct snd_ac97_bus_ops *ops, int num); +int snd_soc_new_ac97_codec(struct snd_soc_codec *codec); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 9ed4e12..f71cc21 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -220,7 +220,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) u16 vendor_id2; u16 ext_status; - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec); if (ret < 0) { dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 53b810d..45ac4a7 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -336,7 +336,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 1650195..2cb8a31 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -337,7 +337,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec); if (ret < 0) { dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 3fad37e..6b36223 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -666,7 +666,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec); if (ret < 0) { dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 998e4c7..2071df7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1219,7 +1219,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) { int ret = 0, reg; - ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); + ret = snd_soc_new_ac97_codec(codec); if (ret < 0) return ret; diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index da7b031..dbfca7e 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -38,6 +38,10 @@ struct snd_ac97_reset_cfg { int gpio_reset; }; +static struct snd_ac97_bus soc_ac97_bus = { + .ops = NULL, /* Gets initialized in snd_soc_set_ac97_ops() */ +}; + /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) { @@ -140,27 +144,17 @@ static void soc_ac97_device_release(struct device *dev) /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec - * @ops: AC97 bus operations - * @num: AC97 codec number * * Initialises AC97 codec resources for use by ad-hoc devices only. */ -int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, - struct snd_ac97_bus_ops *ops, int num) +int snd_soc_new_ac97_codec(struct snd_soc_codec *codec) { codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (codec->ac97 == NULL) return -ENOMEM; - codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); - if (codec->ac97->bus == NULL) { - kfree(codec->ac97); - codec->ac97 = NULL; - return -ENOMEM; - } - - codec->ac97->bus->ops = ops; - codec->ac97->num = num; + codec->ac97->bus = &soc_ac97_bus; + codec->ac97->num = 0; codec->ac97->dev.release = soc_ac97_device_release; /* @@ -183,7 +177,6 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { soc_unregister_ac97_codec(codec); - kfree(codec->ac97->bus); codec->ac97->bus = NULL; put_device(&codec->ac97->dev); codec->ac97 = NULL; @@ -314,6 +307,7 @@ int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops) return -EBUSY; soc_ac97_ops = ops; + soc_ac97_bus.ops = ops; return 0; } -- cgit v1.1 From bdfd60e3c0affb914549f1d22e8aeef71e7828e6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:48 +0100 Subject: ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() soc_{un,}register_ac97_codec() is just a simple wrapper around soc_ac97_dev_{un,}register(). There is no need to split these up into two different sets of functions. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-ac97.c | 80 ++++++++++++++++++++-------------------------------- 1 file changed, 30 insertions(+), 50 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index dbfca7e..b5d23c9 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -42,18 +42,28 @@ static struct snd_ac97_bus soc_ac97_bus = { .ops = NULL, /* Gets initialized in snd_soc_set_ac97_ops() */ }; -/* unregister ac97 codec */ -static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) -{ - if (codec->ac97->dev.bus) - device_del(&codec->ac97->dev); - return 0; -} - /* register ac97 codec to bus */ -static int soc_ac97_dev_register(struct snd_soc_codec *codec) +static int soc_register_ac97_codec(struct snd_soc_codec *codec, + struct snd_soc_dai *codec_dai) { - int err; + int ret; + + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (!codec_dai->driver->ac97_control || codec->ac97_registered) + return 0; + + /* + * It is possible that the AC97 device is already registered to + * the device subsystem. This happens when the device is created + * via snd_ac97_mixer(). Currently only SoC codec that does so + * is the generic AC97 glue but others migh emerge. + * + * In those cases we don't try to register the device again. + */ + if (!codec->ac97_created) + return 0; codec->ac97->dev.bus = &ac97_bus_type; codec->ac97->dev.parent = codec->component.card->dev; @@ -61,53 +71,23 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) dev_set_name(&codec->ac97->dev, "%d-%d:%s", codec->component.card->snd_card->number, 0, codec->component.name); - err = device_add(&codec->ac97->dev); - if (err < 0) { - dev_err(codec->dev, "ASoC: Can't register ac97 bus\n"); - codec->ac97->dev.bus = NULL; - return err; + ret = device_add(&codec->ac97->dev); + if (ret < 0) { + dev_err(codec->dev, "ASoC: AC97 device register failed: %d\n", + ret); + return ret; } - return 0; -} - -static int soc_register_ac97_codec(struct snd_soc_codec *codec, - struct snd_soc_dai *codec_dai) -{ - int ret; + codec->ac97_registered = 1; - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (codec_dai->driver->ac97_control && !codec->ac97_registered) { - /* - * It is possible that the AC97 device is already registered to - * the device subsystem. This happens when the device is created - * via snd_ac97_mixer(). Currently only SoC codec that does so - * is the generic AC97 glue but others migh emerge. - * - * In those cases we don't try to register the device again. - */ - if (!codec->ac97_created) - return 0; - - ret = soc_ac97_dev_register(codec); - if (ret < 0) { - dev_err(codec->dev, - "ASoC: AC97 device register failed: %d\n", ret); - return ret; - } - - codec->ac97_registered = 1; - } return 0; } static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) { - if (codec->ac97_registered) { - soc_ac97_dev_unregister(codec); - codec->ac97_registered = 0; - } + if (!codec->ac97_registered) + return; + device_del(&codec->ac97->dev); + codec->ac97_registered = 0; } static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -- cgit v1.1 From ca005f324ee38308b319c693f40523d959027acf Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:49 +0100 Subject: ASoC: ac97: Drop support for setting platform data via the CPU DAI This has no users since commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") which was almost 5 years ago. Given that this runs after CODEC probe functions have been run it also doesn't seem to be that useful. So drop it altogether to make the code simpler. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 3 --- sound/soc/soc-ac97.c | 12 ------------ sound/soc/soc-core.c | 2 -- 4 files changed, 18 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index e8b3080..c0e0468 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -241,7 +241,6 @@ struct snd_soc_dai { const char *name; int id; struct device *dev; - void *ac97_pdata; /* platform_data for the ac97 codec */ /* driver ops */ struct snd_soc_dai_driver *driver; diff --git a/include/sound/soc.h b/include/sound/soc.h index 44b3ce5..5b4dec6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -509,7 +509,6 @@ int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, extern struct snd_ac97_bus_ops *soc_ac97_ops; int snd_soc_ac97_register_dai_links(struct snd_soc_card *card); -void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd); #else static inline int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) @@ -517,8 +516,6 @@ static inline int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) return 0; } -static inline void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd) {} - static inline int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev) { diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index b5d23c9..f2ed77b 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -342,15 +342,3 @@ err: soc_unregister_ac97_dai_link(&card->rtd[i]); return ret; } - -void snd_soc_ac97_add_pdata(struct snd_soc_pcm_runtime *rtd) -{ - unsigned int i; - - /* add platform data for AC97 devices */ - for (i = 0; i < rtd->num_codecs; i++) { - if (rtd->codec_dais[i]->driver->ac97_control) - snd_ac97_dev_add_pdata(rtd->codec_dais[i]->codec->ac97, - rtd->cpu_dai->ac97_pdata); - } -} diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 026722f..d883b4a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1375,8 +1375,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } - snd_soc_ac97_add_pdata(rtd); - return 0; } -- cgit v1.1 From 6794f709b7124ff1e574c4f4c9494418ab56c4b4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:50 +0100 Subject: ASoC: ac97: Drop delayed device registration We have all the information and dependencies we need to initialize and register the device available in snd_soc_new_ac97_codec(). So there is no need to delay the device registration until after the card itself as been registered. This makes the code significantly simpler and also makes it possible to use the AC'97 device in the CODECs probe function. The later will be required to be able to convert the AC'97 CODEC drivers to regmap. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 10 ----- sound/soc/soc-ac97.c | 118 ++++++--------------------------------------------- sound/soc/soc-core.c | 4 -- 3 files changed, 14 insertions(+), 118 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5b4dec6..206cc8d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -507,15 +507,7 @@ int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev); extern struct snd_ac97_bus_ops *soc_ac97_ops; - -int snd_soc_ac97_register_dai_links(struct snd_soc_card *card); #else - -static inline int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) -{ - return 0; -} - static inline int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, struct platform_device *pdev) { @@ -808,8 +800,6 @@ struct snd_soc_codec { struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int suspended:1; /* Codec is in suspend PM state */ - unsigned int ac97_registered:1; /* Codec has been AC97 registered */ - unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int cache_init:1; /* codec cache has been initialized */ u32 cache_sync; /* Cache needs to be synced to hardware */ diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index f2ed77b..920d76c 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -42,80 +42,6 @@ static struct snd_ac97_bus soc_ac97_bus = { .ops = NULL, /* Gets initialized in snd_soc_set_ac97_ops() */ }; -/* register ac97 codec to bus */ -static int soc_register_ac97_codec(struct snd_soc_codec *codec, - struct snd_soc_dai *codec_dai) -{ - int ret; - - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (!codec_dai->driver->ac97_control || codec->ac97_registered) - return 0; - - /* - * It is possible that the AC97 device is already registered to - * the device subsystem. This happens when the device is created - * via snd_ac97_mixer(). Currently only SoC codec that does so - * is the generic AC97 glue but others migh emerge. - * - * In those cases we don't try to register the device again. - */ - if (!codec->ac97_created) - return 0; - - codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = codec->component.card->dev; - - dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->component.card->snd_card->number, 0, - codec->component.name); - ret = device_add(&codec->ac97->dev); - if (ret < 0) { - dev_err(codec->dev, "ASoC: AC97 device register failed: %d\n", - ret); - return ret; - } - codec->ac97_registered = 1; - - return 0; -} - -static void soc_unregister_ac97_codec(struct snd_soc_codec *codec) -{ - if (!codec->ac97_registered) - return; - device_del(&codec->ac97->dev); - codec->ac97_registered = 0; -} - -static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - int i, ret; - - for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - - ret = soc_register_ac97_codec(codec_dai->codec, codec_dai); - if (ret) { - while (--i >= 0) - soc_unregister_ac97_codec(codec_dai->codec); - return ret; - } - } - - return 0; -} - -static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) -{ - int i; - - for (i = 0; i < rtd->num_codecs; i++) - soc_unregister_ac97_codec(rtd->codec_dais[i]->codec); -} - static void soc_ac97_device_release(struct device *dev) { kfree(to_ac97_t(dev)); @@ -129,22 +55,28 @@ static void soc_ac97_device_release(struct device *dev) */ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec) { + int ret; + codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); if (codec->ac97 == NULL) return -ENOMEM; codec->ac97->bus = &soc_ac97_bus; codec->ac97->num = 0; + + codec->ac97->dev.bus = &ac97_bus_type; + codec->ac97->dev.parent = codec->component.card->dev; codec->ac97->dev.release = soc_ac97_device_release; - /* - * Mark the AC97 device to be created by us. This way we ensure that the - * device will be registered with the device subsystem later on. - */ - codec->ac97_created = 1; - device_initialize(&codec->ac97->dev); + dev_set_name(&codec->ac97->dev, "%d-%d:%s", + codec->component.card->snd_card->number, 0, + codec->component.name); + + ret = device_register(&codec->ac97->dev); + if (ret) + put_device(&codec->ac97->dev); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); @@ -156,11 +88,10 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); */ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { - soc_unregister_ac97_codec(codec); + device_del(&codec->ac97->dev); codec->ac97->bus = NULL; put_device(&codec->ac97->dev); codec->ac97 = NULL; - codec->ac97_created = 0; } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); @@ -321,24 +252,3 @@ int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, return 0; } EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops_of_reset); - -int snd_soc_ac97_register_dai_links(struct snd_soc_card *card) -{ - int i; - int ret; - - /* register any AC97 codecs */ - for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) - goto err; - } - - return 0; -err: - dev_err(card->dev, - "ASoC: failed to register AC97: %d\n", ret); - while (--i >= 0) - soc_unregister_ac97_dai_link(&card->rtd[i]); - return ret; -} diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d883b4a..fba6e28 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1671,10 +1671,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto probe_aux_dev_err; } - ret = snd_soc_ac97_register_dai_links(card); - if (ret < 0) - goto probe_aux_dev_err; - card->instantiated = 1; snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); -- cgit v1.1 From 4bafcf074aca3bd191e4d93c6a140ca52654f192 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:51 +0100 Subject: ASoC: Drop ac97_control initialization from CODEC driver DAIs This is no longer necessary as there is no code anymore that uses this for CODEC DAIs. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 1 - sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/stac9766.c | 2 -- sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - 6 files changed, 7 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index bd9b183..5d90924 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -53,7 +53,6 @@ static const struct snd_soc_dai_ops ac97_dai_ops = { static struct snd_soc_dai_driver ac97_dai = { .name = "ac97-hifi", - .ac97_control = 1, .playback = { .stream_name = "AC97 Playback", .channels_min = 1, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index f71cc21..c6cb101 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -170,7 +170,6 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static struct snd_soc_dai_driver ad1980_dai = { .name = "ad1980-hifi", - .ac97_control = 1, .playback = { .stream_name = "Playback", .channels_min = 2, diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 45ac4a7..c080806 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -294,7 +294,6 @@ static const struct snd_soc_dai_ops stac9766_dai_ops_digital = { static struct snd_soc_dai_driver stac9766_dai[] = { { .name = "stac9766-hifi-analog", - .ac97_control = 1, /* stream cababilities */ .playback = { @@ -316,7 +315,6 @@ static struct snd_soc_dai_driver stac9766_dai[] = { }, { .name = "stac9766-hifi-IEC958", - .ac97_control = 1, /* stream cababilities */ .playback = { diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 2cb8a31..5b5118b 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -263,7 +263,6 @@ static const struct snd_soc_dai_ops wm9705_dai_ops = { static struct snd_soc_dai_driver wm9705_dai[] = { { .name = "wm9705-hifi", - .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 6b36223..9fa794b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -565,7 +565,6 @@ static const struct snd_soc_dai_ops wm9712_dai_ops_aux = { static struct snd_soc_dai_driver wm9712_dai[] = { { .name = "wm9712-hifi", - .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2071df7..cd1b266 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,7 +1076,6 @@ static const struct snd_soc_dai_ops wm9713_dai_ops_voice = { static struct snd_soc_dai_driver wm9713_dai[] = { { .name = "wm9713-hifi", - .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, -- cgit v1.1 From bc2632140435cc84f9817f1c362479b23dbdfebc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:52 +0100 Subject: ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control Setting the ac97_control field on a CPU DAI tells the ASoC core that this DAI in addition to audio data also transports control data to the CODEC. This causes the core to suspend the DAI after the CODEC and resume it before the CODEC so communication to the CODEC is still possible. This is not necessarily something that is specific to AC'97 and can be used by other buses with the same requirement. This patch renames the flag from ac97_control to bus_control to make this explicit. While we are at it also change the type from int to bool. The following semantich patch was used for automatic conversion of the drivers: // @@ identifier drv; @@ struct snd_soc_dai_driver drv = { - .ac97_control + .bus_control = - 1 + true }; // Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 ++- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 4 ++-- sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +++--- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/sh/hac.c | 2 +- sound/soc/soc-core.c | 26 ++++++++++++++------------ sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- 15 files changed, 33 insertions(+), 30 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index c0e0468..a3738be 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -206,7 +206,6 @@ struct snd_soc_dai_driver { /* DAI description */ const char *name; unsigned int id; - int ac97_control; unsigned int base; /* DAI driver callbacks */ @@ -216,6 +215,8 @@ struct snd_soc_dai_driver { int (*resume)(struct snd_soc_dai *dai); /* compress dai */ bool compress_dai; + /* DAI is also used for the control bus */ + bool bus_control; /* ops */ const struct snd_soc_dai_ops *ops; diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index c8a2de1..5159a50 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -205,7 +205,7 @@ static int au1xac97c_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver au1xac97c_dai_driver = { .name = "alchemy-ac97c", - .ac97_control = 1, + .bus_control = true, .probe = au1xac97c_dai_probe, .playback = { .rates = AC97_RATES, diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 84f31e1..c6daec9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -343,7 +343,7 @@ static const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { }; static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { - .ac97_control = 1, + .bus_control = true, .probe = au1xpsc_ac97_probe, .playback = { .rates = AC97_RATES, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index e82eb37..6bf21a6 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -260,7 +260,7 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) #endif static struct snd_soc_dai_driver bfin_ac97_dai = { - .ac97_control = 1, + .bus_control = true, .suspend = bf5xx_ac97_suspend, .resume = bf5xx_ac97_resume, .playback = { diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index f30dadf..6b8a366 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -338,7 +338,7 @@ static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { static struct snd_soc_dai_driver ep93xx_ac97_dai = { .name = "ep93xx-ac97", .id = 0, - .ac97_control = 1, + .bus_control = true, .probe = ep93xx_ac97_dai_probe, .playback = { .stream_name = "AC97 Playback", diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e695517..7fd3cbc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1099,7 +1099,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { }; static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index ab2fdd7..60b0a5b 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -382,7 +382,7 @@ static struct snd_soc_dai_driver imx_ssi_dai = { static struct snd_soc_dai_driver imx_ac97_dai = { .probe = imx_ssi_dai_probe, - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 640801a..c6ed6ba 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -237,7 +237,7 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = { static struct snd_soc_dai_driver psc_ac97_dai[] = { { .name = "mpc5200-psc-ac97.0", - .ac97_control = 1, + .bus_control = true, .probe = psc_ac97_probe, .playback = { .stream_name = "AC97 Playback", @@ -257,7 +257,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { }, { .name = "mpc5200-psc-ac97.1", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 SPDIF", .channels_min = 1, diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index f2f67942..dff443e 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -298,7 +298,7 @@ static const struct snd_soc_dai_ops nuc900_ac97_dai_ops = { static struct snd_soc_dai_driver nuc900_ac97_dai = { .probe = nuc900_ac97_probe, .remove = nuc900_ac97_remove, - .ac97_control = 1, + .bus_control = true, .playback = { .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index ae956e3..73ca282 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -157,7 +157,7 @@ static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -174,7 +174,7 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-aux", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -191,7 +191,7 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { }, { .name = "pxa2xx-ac97-mic", - .ac97_control = 1, + .bus_control = true, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index e161511..7952a62 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -288,7 +288,7 @@ static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -306,7 +306,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { }, [S3C_AC97_DAI_MIC] = { .name = "samsung-ac97-mic", - .ac97_control = 1, + .bus_control = true, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 0af2e4d..d5f567e 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -272,7 +272,7 @@ static const struct snd_soc_dai_ops hac_dai_ops = { static struct snd_soc_dai_driver sh4_hac_dai[] = { { .name = "hac-dai.0", - .ac97_control = 1, + .bus_control = true, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fba6e28..f5bebca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -550,7 +550,7 @@ int snd_soc_suspend(struct device *dev) if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) + if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control) cpu_dai->driver->suspend(cpu_dai); if (platform->driver->suspend && !platform->suspended) { platform->driver->suspend(cpu_dai); @@ -629,7 +629,7 @@ int snd_soc_suspend(struct device *dev) if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) + if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control) cpu_dai->driver->suspend(cpu_dai); /* deactivate pins to sleep state */ @@ -665,14 +665,14 @@ static void soc_resume_deferred(struct work_struct *work) if (card->resume_pre) card->resume_pre(card); - /* resume AC97 DAIs */ + /* resume control bus DAIs */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->driver->resume && cpu_dai->driver->ac97_control) + if (cpu_dai->driver->resume && cpu_dai->driver->bus_control) cpu_dai->driver->resume(cpu_dai); } @@ -733,7 +733,7 @@ static void soc_resume_deferred(struct work_struct *work) if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) + if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control) cpu_dai->driver->resume(cpu_dai); if (platform->driver->resume && platform->suspended) { platform->driver->resume(cpu_dai); @@ -758,7 +758,8 @@ static void soc_resume_deferred(struct work_struct *work) int snd_soc_resume(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); - int i, ac97_control = 0; + bool bus_control = false; + int i; /* If the card is not initialized yet there is nothing to do */ if (!card->instantiated) @@ -781,17 +782,18 @@ int snd_soc_resume(struct device *dev) } } - /* AC97 devices might have other drivers hanging off them so - * need to resume immediately. Other drivers don't have that - * problem and may take a substantial amount of time to resume + /* + * DAIs that also act as the control bus master might have other drivers + * hanging off them so need to resume immediately. Other drivers don't + * have that problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - ac97_control |= cpu_dai->driver->ac97_control; + bus_control |= cpu_dai->driver->bus_control; } - if (ac97_control) { - dev_dbg(dev, "ASoC: Resuming AC97 immediately\n"); + if (bus_control) { + dev_dbg(dev, "ASoC: Resuming control bus master immediately\n"); soc_resume_deferred(&card->deferred_resume_work); } else { dev_dbg(dev, "ASoC: Scheduling resume work\n"); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 3b0fa12..29a9957 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -228,7 +228,7 @@ static int tegra20_ac97_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver tegra20_ac97_dai = { .name = "tegra-ac97-pcm", - .ac97_control = 1, + .bus_control = true, .probe = tegra20_ac97_probe, .playback = { .stream_name = "PCM Playback", diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 9edd68d..f7135cd 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -152,7 +152,7 @@ static int txx9aclc_ac97_remove(struct snd_soc_dai *dai) } static struct snd_soc_dai_driver txx9aclc_ac97_dai = { - .ac97_control = 1, + .bus_control = true, .probe = txx9aclc_ac97_probe, .remove = txx9aclc_ac97_remove, .playback = { -- cgit v1.1 From 358a8bb5628420529e4f0b77068155ca8fa8973b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 10 Nov 2014 22:41:53 +0100 Subject: ASoC: ac97: Push snd_ac97 pointer to the driver level Now that the ASoC core no longer needs a handle to the AC'97 device that is associated with a CODEC we can remove it from the snd_soc_codec struct and push it into the individual driver state structs like we do for other communication buses. Doing so creates a clean separation between the AC'97 bus support and the ASoC core. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc.h | 5 ++--- sound/soc/codecs/ac97.c | 17 +++++++++++++---- sound/soc/codecs/ad1980.c | 27 ++++++++++++++++++--------- sound/soc/codecs/stac9766.c | 38 ++++++++++++++++++++++++-------------- sound/soc/codecs/wm9705.c | 31 ++++++++++++++++++++++--------- sound/soc/codecs/wm9712.c | 32 +++++++++++++++++++++----------- sound/soc/codecs/wm9713.c | 31 ++++++++++++++++++++----------- sound/soc/soc-ac97.c | 40 +++++++++++++++++++++------------------- 8 files changed, 141 insertions(+), 80 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 206cc8d..9e513ae 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -499,8 +499,8 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); #ifdef CONFIG_SND_SOC_AC97_BUS -int snd_soc_new_ac97_codec(struct snd_soc_codec *codec); -void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec); +void snd_soc_free_ac97_codec(struct snd_ac97 *ac97); int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops); int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops, @@ -797,7 +797,6 @@ struct snd_soc_codec { struct list_head card_list; /* runtime */ - struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int cache_bypass:1; /* Suppress access to the cache */ unsigned int suspended:1; /* Codec is in suspend PM state */ unsigned int cache_init:1; /* codec cache has been initialized */ diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 5d90924..c6e5a31 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -37,10 +37,11 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(codec->ac97, reg, substream->runtime->rate); + return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ @@ -70,6 +71,7 @@ static struct snd_soc_dai_driver ac97_dai = { static int ac97_soc_probe(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97; struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; int ret; @@ -81,24 +83,31 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return ret; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); - ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); + ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97); if (ret < 0) return ret; + snd_soc_codec_set_drvdata(codec, ac97); + return 0; } #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { - snd_ac97_suspend(codec->ac97); + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_ac97_suspend(ac97); return 0; } static int ac97_soc_resume(struct snd_soc_codec *codec) { - snd_ac97_resume(codec->ac97); + + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_ac97_resume(ac97); return 0; } diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index c6cb101..93bd47d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -135,6 +135,7 @@ static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; switch (reg) { @@ -144,7 +145,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, case AC97_EXTENDED_STATUS: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops->read(codec->ac97, reg); + return soc_ac97_ops->read(ac97, reg); default: reg = reg >> 1; @@ -158,9 +159,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(ac97, reg, val); reg = reg >> 1; if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; @@ -186,16 +188,17 @@ static struct snd_soc_dai_driver ad1980_dai = { static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); unsigned int retry_cnt = 0; do { if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(ac97); if (ac97_read(codec, AC97_RESET) == 0x0090) return 1; } - soc_ac97_ops->reset(codec->ac97); + soc_ac97_ops->reset(ac97); /* * Set bit 16slot in register 74h, then every slot will has only * 16 bits. This command is sent out in 20bit mode, in which @@ -215,16 +218,20 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) static int ad1980_soc_probe(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97; int ret; u16 vendor_id2; u16 ext_status; - ret = snd_soc_new_ac97_codec(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to register AC97 codec\n"); + ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(ac97)) { + ret = PTR_ERR(ac97); + dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } + snd_soc_codec_set_drvdata(codec, ac97); + ret = ad1980_reset(codec, 0); if (ret < 0) goto reset_err; @@ -261,13 +268,15 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) return 0; reset_err: - snd_soc_free_ac97_codec(codec); + snd_soc_free_ac97_codec(ac97); return ret; } static int ad1980_soc_remove(struct snd_soc_codec *codec) { - snd_soc_free_ac97_codec(codec); + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(ac97); return 0; } diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index c080806..f37a79e 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -139,18 +139,19 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(ac97, reg, val); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(ac97, reg, val); cache[reg / 2] = val; return 0; } @@ -158,11 +159,12 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 val = 0, *cache = codec->reg_cache; if (reg > AC97_STAC_PAGE0) { stac9766_ac97_write(codec, AC97_INT_PAGING, 0); - val = soc_ac97_ops->read(codec->ac97, reg - AC97_STAC_PAGE0); + val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0); stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } @@ -173,7 +175,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { - val = soc_ac97_ops->read(codec->ac97, reg); + val = soc_ac97_ops->read(ac97, reg); return val; } return cache[reg / 2]; @@ -240,15 +242,17 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec, static int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } - soc_ac97_ops->reset(codec->ac97); + soc_ac97_ops->reset(ac97); if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; @@ -262,6 +266,7 @@ static int stac9766_codec_suspend(struct snd_soc_codec *codec) static int stac9766_codec_resume(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 id, reset; reset = 0; @@ -271,8 +276,8 @@ reset: printk(KERN_ERR "stac9766 failed to resume"); return -EIO; } - codec->ac97->bus->ops->warm_reset(codec->ac97); - id = soc_ac97_ops->read(codec->ac97, AC97_VENDOR_ID2); + ac97->bus->ops->warm_reset(ac97); + id = soc_ac97_ops->read(ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { stac9766_reset(codec, 0); reset++; @@ -332,11 +337,14 @@ static struct snd_soc_dai_driver stac9766_dai[] = { static int stac9766_codec_probe(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97; int ret = 0; - ret = snd_soc_new_ac97_codec(codec); - if (ret < 0) - goto codec_err; + ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(ac97)) + return PTR_ERR(ac97); + + snd_soc_codec_set_drvdata(codec, ac97); /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ @@ -355,13 +363,15 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) return 0; codec_err: - snd_soc_free_ac97_codec(codec); + snd_soc_free_ac97_codec(ac97); return ret; } static int stac9766_codec_remove(struct snd_soc_codec *codec) { - snd_soc_free_ac97_codec(codec); + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(ac97); return 0; } diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 5b5118b..d3a800f 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -203,13 +203,14 @@ static const struct snd_soc_dapm_route wm9705_audio_map[] = { /* We use a register cache to enhance read performance. */ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; switch (reg) { case AC97_RESET: case AC97_VENDOR_ID1: case AC97_VENDOR_ID2: - return soc_ac97_ops->read(codec->ac97, reg); + return soc_ac97_ops->read(ac97, reg); default: reg = reg >> 1; @@ -223,9 +224,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9705_reg))) cache[reg] = val; @@ -293,8 +295,10 @@ static struct snd_soc_dai_driver wm9705_dai[] = { static int wm9705_reset(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + if (soc_ac97_ops->reset) { - soc_ac97_ops->reset(codec->ac97); + soc_ac97_ops->reset(ac97); if (ac97_read(codec, 0) == wm9705_reg[0]) return 0; /* Success */ } @@ -307,13 +311,16 @@ static int wm9705_reset(struct snd_soc_codec *codec) #ifdef CONFIG_PM static int wm9705_soc_suspend(struct snd_soc_codec *codec) { - soc_ac97_ops->write(codec->ac97, AC97_POWERDOWN, 0xffff); + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + soc_ac97_ops->write(ac97, AC97_POWERDOWN, 0xffff); return 0; } static int wm9705_soc_resume(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); int i, ret; u16 *cache = codec->reg_cache; @@ -322,7 +329,7 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) return ret; for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { - soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(ac97, i, cache[i>>1]); } return 0; @@ -334,14 +341,18 @@ static int wm9705_soc_resume(struct snd_soc_codec *codec) static int wm9705_soc_probe(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97; int ret = 0; - ret = snd_soc_new_ac97_codec(codec); - if (ret < 0) { + ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(ac97)) { + ret = PTR_ERR(ac97); dev_err(codec->dev, "Failed to register AC97 codec\n"); return ret; } + snd_soc_codec_set_drvdata(codec, ac97); + ret = wm9705_reset(codec); if (ret) goto reset_err; @@ -349,13 +360,15 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) return 0; reset_err: - snd_soc_free_ac97_codec(codec); + snd_soc_free_ac97_codec(ac97); return ret; } static int wm9705_soc_remove(struct snd_soc_codec *codec) { - snd_soc_free_ac97_codec(codec); + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(ac97); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 9fa794b..52a211b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -24,6 +24,7 @@ #include "wm9712.h" struct wm9712_priv { + struct snd_ac97 *ac97; unsigned int hp_mixer[2]; struct mutex lock; }; @@ -484,12 +485,13 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_REC_GAIN) - return soc_ac97_ops->read(codec->ac97, reg); + return soc_ac97_ops->read(wm9712->ac97, reg); else { reg = reg >> 1; @@ -503,9 +505,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(wm9712->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; @@ -613,15 +616,17 @@ static int wm9712_set_bias_level(struct snd_soc_codec *codec, static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(wm9712->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } - soc_ac97_ops->reset(codec->ac97); + soc_ac97_ops->reset(wm9712->ac97); if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(wm9712->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; @@ -639,6 +644,7 @@ static int wm9712_soc_suspend(struct snd_soc_codec *codec) static int wm9712_soc_resume(struct snd_soc_codec *codec) { + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int i, ret; u16 *cache = codec->reg_cache; @@ -654,7 +660,7 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || (i > 0x58 && i != 0x5c)) continue; - soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(wm9712->ac97, i, cache[i>>1]); } } @@ -663,11 +669,13 @@ static int wm9712_soc_resume(struct snd_soc_codec *codec) static int wm9712_soc_probe(struct snd_soc_codec *codec) { + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_new_ac97_codec(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to register AC97 codec\n"); + wm9712->ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(wm9712->ac97)) { + ret = PTR_ERR(wm9712->ac97); + dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } @@ -683,13 +691,15 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) return 0; reset_err: - snd_soc_free_ac97_codec(codec); + snd_soc_free_ac97_codec(wm9712->ac97); return ret; } static int wm9712_soc_remove(struct snd_soc_codec *codec) { - snd_soc_free_ac97_codec(codec); + struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(wm9712->ac97); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index cd1b266..6c95d98 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -30,6 +30,7 @@ #include "wm9713.h" struct wm9713_priv { + struct snd_ac97 *ac97; u32 pll_in; /* PLL input frequency */ unsigned int hp_mixer[2]; struct mutex lock; @@ -674,12 +675,13 @@ static const struct snd_soc_dapm_route wm9713_audio_map[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || reg == AC97_CD) - return soc_ac97_ops->read(codec->ac97, reg); + return soc_ac97_ops->read(wm9713->ac97, reg); else { reg = reg >> 1; @@ -693,8 +695,10 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + u16 *cache = codec->reg_cache; - soc_ac97_ops->write(codec->ac97, reg, val); + soc_ac97_ops->write(wm9713->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; @@ -1121,15 +1125,17 @@ static struct snd_soc_dai_driver wm9713_dai[] = { int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(wm9713->ac97); if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } - soc_ac97_ops->reset(codec->ac97); + soc_ac97_ops->reset(wm9713->ac97); if (soc_ac97_ops->warm_reset) - soc_ac97_ops->warm_reset(codec->ac97); + soc_ac97_ops->warm_reset(wm9713->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) { dev_err(codec->dev, "Failed to reset: AC97 link error\n"); return -EIO; @@ -1207,7 +1213,7 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || i == AC97_EXTENDED_MSTATUS || i > 0x66) continue; - soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); + soc_ac97_ops->write(wm9713->ac97, i, cache[i>>1]); } } @@ -1216,11 +1222,12 @@ static int wm9713_soc_resume(struct snd_soc_codec *codec) static int wm9713_soc_probe(struct snd_soc_codec *codec) { + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - ret = snd_soc_new_ac97_codec(codec); - if (ret < 0) - return ret; + wm9713->ac97 = snd_soc_new_ac97_codec(codec); + if (IS_ERR(wm9713->ac97)) + return PTR_ERR(wm9713->ac97); /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ @@ -1238,13 +1245,15 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) return 0; reset_err: - snd_soc_free_ac97_codec(codec); + snd_soc_free_ac97_codec(wm9713->ac97); return ret; } static int wm9713_soc_remove(struct snd_soc_codec *codec) { - snd_soc_free_ac97_codec(codec); + struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); + + snd_soc_free_ac97_codec(wm9713->ac97); return 0; } diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 920d76c..2e10e9a 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -53,30 +53,33 @@ static void soc_ac97_device_release(struct device *dev) * * Initialises AC97 codec resources for use by ad-hoc devices only. */ -int snd_soc_new_ac97_codec(struct snd_soc_codec *codec) +struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec) { + struct snd_ac97 *ac97; int ret; - codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); - if (codec->ac97 == NULL) - return -ENOMEM; + ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); + if (ac97 == NULL) + return ERR_PTR(-ENOMEM); - codec->ac97->bus = &soc_ac97_bus; - codec->ac97->num = 0; + ac97->bus = &soc_ac97_bus; + ac97->num = 0; - codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = codec->component.card->dev; - codec->ac97->dev.release = soc_ac97_device_release; + ac97->dev.bus = &ac97_bus_type; + ac97->dev.parent = codec->component.card->dev; + ac97->dev.release = soc_ac97_device_release; - dev_set_name(&codec->ac97->dev, "%d-%d:%s", + dev_set_name(&ac97->dev, "%d-%d:%s", codec->component.card->snd_card->number, 0, codec->component.name); - ret = device_register(&codec->ac97->dev); - if (ret) - put_device(&codec->ac97->dev); + ret = device_register(&ac97->dev); + if (ret) { + put_device(&ac97->dev); + return ERR_PTR(ret); + } - return ret; + return ac97; } EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); @@ -86,12 +89,11 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); * * Frees AC97 codec device resources. */ -void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) +void snd_soc_free_ac97_codec(struct snd_ac97 *ac97) { - device_del(&codec->ac97->dev); - codec->ac97->bus = NULL; - put_device(&codec->ac97->dev); - codec->ac97 = NULL; + device_del(&ac97->dev); + ac97->bus = NULL; + put_device(&ac97->dev); } EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); -- cgit v1.1 From a5a267cf9ca9937b0ef946b502657ae7638282f6 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Tue, 18 Nov 2014 17:42:54 +0530 Subject: ASoC: rt286: build warning of section mismatch while building we were getting the following build warning: Section mismatch in reference from the function rt286_i2c_probe() to the variable .init.data:force_combo_jack_table The function rt286_i2c_probe() references the variable __initdata force_combo_jack_table. This is often because rt286_i2c_probe lacks a __initdata annotation or the annotation of force_combo_jack_table is wrong. we were getting the warning as force_combo_jack_table was marked with __initdata Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2e818aa..2cd4fe4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); -static struct dmi_system_id force_combo_jack_table[] __initdata = { +static struct dmi_system_id force_combo_jack_table[] = { { .ident = "Intel Wilson Beach", .matches = { -- cgit v1.1 From d6d521799fac14e14dead4e9428158340ff6b95f Mon Sep 17 00:00:00 2001 From: JS Park Date: Tue, 18 Nov 2014 16:07:22 +0000 Subject: ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs Signed-off-by: JS Park Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0a08ef5e..6a2a035 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- cgit v1.1 From 5f217f905bc5e9d609d0aac830736bcfc087c7f5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 18 Nov 2014 23:59:40 +0900 Subject: ALSA: firewire-lib: fix kerneldoc errors Complete missing parameters, correct wrong reference, and add an explaination about the differences between the latest specification and our implementation. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.h | 24 ++++++++++++++++++++++-- sound/firewire/cmp.c | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 4823c08..e6e8926 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -23,7 +23,7 @@ * corresponds to the end of event in the packet. Out of IEC 61883. * @CIP_WRONG_DBS: Only for in-stream. The value of dbs is wrong in in-packets. * The value of data_block_quadlets is used instead of reported value. - * @SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is + * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first * packet is not continuous from an initial value. @@ -43,7 +43,27 @@ enum cip_flags { }; /** - * enum cip_sfc - a stream's sample rate + * enum cip_sfc - supported Sampling Frequency Codes (SFCs) + * @CIP_SFC_32000: 32,000 data blocks + * @CIP_SFC_44100: 44,100 data blocks + * @CIP_SFC_48000: 48,000 data blocks + * @CIP_SFC_88200: 88,200 data blocks + * @CIP_SFC_96000: 96,000 data blocks + * @CIP_SFC_176400: 176,400 data blocks + * @CIP_SFC_192000: 192,000 data blocks + * @CIP_SFC_COUNT: the number of supported SFCs + * + * These values are used to show nominal Sampling Frequency Code in + * Format Dependent Field (FDF) of AMDTP packet header. In IEC 61883-6:2002, + * this code means the number of events per second. Actually the code + * represents the number of data blocks transferred per second in an AMDTP + * stream. + * + * In IEC 61883-6:2005, some extensions were added to support more types of + * data such as 'One Bit LInear Audio', therefore the meaning of SFC became + * different depending on the types. + * + * Currently our implementation is compatible with IEC 61883-6:2002. */ enum cip_sfc { CIP_SFC_32000 = 0, diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..ae3bc19 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -114,6 +114,7 @@ static int pcr_modify(struct cmp_connection *c, * cmp_connection_init - initializes a connection manager * @c: the connection manager to initialize * @unit: a unit of the target device + * @direction: input or output * @pcr_index: the index of the iPCR/oPCR on the target device */ int cmp_connection_init(struct cmp_connection *c, @@ -154,6 +155,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store the result of checking the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- cgit v1.1 From 00e4c3b6e285da90e736fbefff3d9e74a200ee54 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Nov 2014 16:25:27 +0000 Subject: ASoC: wm_adsp: Move core_ena to be co-located with start bit Many firmwares do not wait for the start bit before they begin processing audio, whilst this is a bug on the firmware side there are too many such firmwares in the wild to ignore the situation. This patch moves the core enable to happen at same time as the start, the firmware looses the ability to overlap its own startup with the audio path bring up but we ensure that all firmwares behave. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6712478..cce9020 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1595,13 +1595,6 @@ static void wm_adsp2_boot_work(struct work_struct *work) if (ret != 0) goto err; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA, - ADSP2_CORE_ENA); - if (ret != 0) - goto err; - dsp->running = true; return; @@ -1651,8 +1644,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_START, - ADSP2_START); + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; break; -- cgit v1.1 From 2dfe2b08d280c15cc7266de40412c2a911643148 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Nov 2014 13:52:18 +0800 Subject: ASoC: rt5677: Align the reg_default table with tab character Align the reg_default table with tab character Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9ae2e84..b2d88bb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,13 +55,13 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_ASRC_12, 0x0018}, - {RT5677_PR_BASE + 0x3d, 0x364d}, - {RT5677_PR_BASE + 0x17, 0x4fc0}, - {RT5677_PR_BASE + 0x13, 0x0312}, - {RT5677_PR_BASE + 0x1e, 0x0000}, - {RT5677_PR_BASE + 0x12, 0x0eaa}, - {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_PR_BASE + 0x17, 0x4fc0}, + {RT5677_PR_BASE + 0x13, 0x0312}, + {RT5677_PR_BASE + 0x1e, 0x0000}, + {RT5677_PR_BASE + 0x12, 0x0eaa}, + {RT5677_PR_BASE + 0x14, 0x018a}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- cgit v1.1 From 35d40d10e95f52569570dc4e26da19f072aa256d Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Nov 2014 13:52:19 +0800 Subject: ASoC: rt5677: Follow the gpio naming rule to rename the irq function Follow the gpio naming rule to rename the irq function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index b2d88bb..dd080cd 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4552,7 +4552,7 @@ static struct regmap_irq_chip rt5677_irq_chip = { .mask_invert = 1, }; -static int rt5677_irq_init(struct i2c_client *i2c) +static int rt5677_init_irq(struct i2c_client *i2c) { int ret; struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); @@ -4579,7 +4579,7 @@ static int rt5677_irq_init(struct i2c_client *i2c) return 0; } -static void rt5677_irq_exit(struct i2c_client *i2c) +static void rt5677_free_irq(struct i2c_client *i2c) { struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); @@ -4693,7 +4693,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, } rt5677_init_gpio(i2c); - rt5677_irq_init(i2c); + rt5677_init_irq(i2c); return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677, rt5677_dai, ARRAY_SIZE(rt5677_dai)); @@ -4701,9 +4701,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, static int rt5677_i2c_remove(struct i2c_client *i2c) { - rt5677_irq_exit(i2c); - snd_soc_unregister_codec(&i2c->dev); + rt5677_free_irq(i2c); rt5677_free_gpio(i2c); return 0; -- cgit v1.1 From 683996cb2255373c2055e7b69584ac153eb49f42 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 19 Nov 2014 13:52:20 +0800 Subject: ASoC: rt5677: Set the slow charge of the vref in the end of the power sequences Set the slow charge of the vref in the end of the power sequences Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 56 ++++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/rt5677.h | 1 + 2 files changed, 47 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index dd080cd..f2211f1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2184,6 +2184,31 @@ static int rt5677_if2_adc_tdm_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (codec->dapm.bias_level != SND_SOC_BIAS_ON && + !rt5677->is_vref_slow) { + mdelay(20); + regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_PWR_FV1 | RT5677_PWR_FV2, + RT5677_PWR_FV1 | RT5677_PWR_FV2); + rt5677->is_vref_slow = true; + } + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU), @@ -2669,13 +2694,20 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_MUX("PDM2 R Mux", RT5677_PDM_OUT_CTRL, RT5677_M_PDM2_R_SFT, 1, &rt5677_pdm2_r_mux), - SND_SOC_DAPM_PGA_S("LOUT1 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO1_BIT, + SND_SOC_DAPM_PGA_S("LOUT1 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO1_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("LOUT2 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO2_BIT, + SND_SOC_DAPM_PGA_S("LOUT2 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("LOUT3 amp", 1, RT5677_PWR_ANLG1, RT5677_PWR_LO3_BIT, + SND_SOC_DAPM_PGA_S("LOUT3 amp", 0, RT5677_PWR_ANLG1, RT5677_PWR_LO3_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("LOUT1 vref", 1, SND_SOC_NOPM, 0, 0, + rt5677_vref_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT2 vref", 1, SND_SOC_NOPM, 0, 0, + rt5677_vref_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT3 vref", 1, SND_SOC_NOPM, 0, 0, + rt5677_vref_event, SND_SOC_DAPM_POST_PMU), + /* Output Lines */ SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), @@ -2684,6 +2716,8 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("PDM2L"), SND_SOC_DAPM_OUTPUT("PDM2R"), + + SND_SOC_DAPM_POST("vref", rt5677_vref_event), }; static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { @@ -3572,9 +3606,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "LOUT2 amp", NULL, "DAC 2" }, { "LOUT3 amp", NULL, "DAC 3" }, - { "LOUT1", NULL, "LOUT1 amp" }, - { "LOUT2", NULL, "LOUT2 amp" }, - { "LOUT3", NULL, "LOUT3 amp" }, + { "LOUT1 vref", NULL, "LOUT1 amp" }, + { "LOUT2 vref", NULL, "LOUT2 amp" }, + { "LOUT3 vref", NULL, "LOUT3 amp" }, + + { "LOUT1", NULL, "LOUT1 vref" }, + { "LOUT2", NULL, "LOUT2 vref" }, + { "LOUT3", NULL, "LOUT3 vref" }, { "PDM1L", NULL, "PDM1 L Mux" }, { "PDM1R", NULL, "PDM1 R Mux" }, @@ -3957,14 +3995,12 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0f00); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, + RT5677_PWR_FV1 | RT5677_PWR_FV2 | RT5677_PWR_VREF1 | RT5677_PWR_MB | RT5677_PWR_BG | RT5677_PWR_VREF2, RT5677_PWR_VREF1 | RT5677_PWR_MB | RT5677_PWR_BG | RT5677_PWR_VREF2); - mdelay(20); - regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG1, - RT5677_PWR_FV1 | RT5677_PWR_FV2, - RT5677_PWR_FV1 | RT5677_PWR_FV2); + rt5677->is_vref_slow = false; regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_CORE, RT5677_PWR_CORE); regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 2979d5a..a02f64c 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1687,6 +1687,7 @@ struct rt5677_priv { bool dsp_vad_en; struct regmap_irq_chip_data *irq_data; bool is_dsp_mode; + bool is_vref_slow; }; #endif /* __RT5677_H__ */ -- cgit v1.1 From 20feb881988cdf5f53304c355ae8ee3bf82e80ec Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 18 Nov 2014 19:45:52 +0100 Subject: ASoC: Add helper functions for deferred regmap setup Some drivers (most notably the AC'97 drivers) do not have access to their regmap struct when the component/codec is registered. For those drivers the automatic regmap setup will not work and needs to be done manually, typically from the component/CODEC drivers probe callback. This patch adds a set of helper function to handle deferred regmap initialization as well as early regmap tear-down. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 35 +++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++++++++++++++++---------- 2 files changed, 82 insertions(+), 11 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..342b43b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1277,6 +1277,41 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +void snd_soc_component_init_regmap(struct snd_soc_component *component, + struct regmap *regmap); +void snd_soc_component_exit_regmap(struct snd_soc_component *component); + +/** + * snd_soc_codec_init_regmap() - Initialize regmap instance for the CODEC + * @codec: The CODEC for which to initialize the regmap instance + * @regmap: The regmap instance that should be used by the CODEC + * + * This function allows deferred assignment of the regmap instance that is + * associated with the CODEC. Only use this if the regmap instance is not yet + * ready when the CODEC is registered. The function must also be called before + * the first IO attempt of the CODEC. + */ +static inline void snd_soc_codec_init_regmap(struct snd_soc_codec *codec, + struct regmap *regmap) +{ + snd_soc_component_init_regmap(&codec->component, regmap); +} + +/** + * snd_soc_codec_exit_regmap() - De-initialize regmap instance for the CODEC + * @codec: The CODEC for which to de-initialize the regmap instance + * + * Calls regmap_exit() on the regmap instance associated to the CODEC and + * removes the regmap instance from the CODEC. + * + * This function should only be used if snd_soc_codec_init_regmap() was used to + * initialize the regmap instance. + */ +static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) +{ + snd_soc_component_exit_regmap(&codec->component); +} + /* device driver data */ static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..5fd5f08 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3996,22 +3996,58 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, return 0; } -static void snd_soc_component_init_regmap(struct snd_soc_component *component) +static void snd_soc_component_setup_regmap(struct snd_soc_component *component) { - if (!component->regmap) - component->regmap = dev_get_regmap(component->dev, NULL); - if (component->regmap) { - int val_bytes = regmap_get_val_bytes(component->regmap); - /* Errors are legitimate for non-integer byte multiples */ - if (val_bytes > 0) - component->val_bytes = val_bytes; - } + int val_bytes = regmap_get_val_bytes(component->regmap); + + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; +} + +/** + * snd_soc_component_init_regmap() - Initialize regmap instance for the component + * @component: The component for which to initialize the regmap instance + * @regmap: The regmap instance that should be used by the component + * + * This function allows deferred assignment of the regmap instance that is + * associated with the component. Only use this if the regmap instance is not + * yet ready when the component is registered. The function must also be called + * before the first IO attempt of the component. + */ +void snd_soc_component_init_regmap(struct snd_soc_component *component, + struct regmap *regmap) +{ + component->regmap = regmap; + snd_soc_component_setup_regmap(component); } +EXPORT_SYMBOL_GPL(snd_soc_component_init_regmap); + +/** + * snd_soc_component_exit_regmap() - De-initialize regmap instance for the component + * @component: The component for which to de-initialize the regmap instance + * + * Calls regmap_exit() on the regmap instance associated to the component and + * removes the regmap instance from the component. + * + * This function should only be used if snd_soc_component_init_regmap() was used + * to initialize the regmap instance. + */ +void snd_soc_component_exit_regmap(struct snd_soc_component *component) +{ + regmap_exit(component->regmap); + component->regmap = NULL; +} +EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { - if (!component->write && !component->read) - snd_soc_component_init_regmap(component); + if (!component->write && !component->read) { + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) + snd_soc_component_setup_regmap(component); + } list_add(&component->list, &component_list); } -- cgit v1.1 From 82d14636418299b4f54511a02373796b38747b48 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 18 Nov 2014 19:45:53 +0100 Subject: ASoC: ad1980: Convert to regmap This patch converts the ad1980 driver to use regmap for its IO. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/ad1980.c | 141 ++++++++++++++++++++++++++++------------------ 2 files changed, 88 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..6a66216 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -223,6 +223,7 @@ config SND_SOC_AD193X_I2C select SND_SOC_AD193X config SND_SOC_AD1980 + select REGMAP_AC97 tristate config SND_SOC_AD73311 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 93bd47d..5fd4a29 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -24,32 +24,86 @@ #include #include #include +#include #include #include #include #include #include -/* - * AD1980 register cache - */ -static const u16 ad1980_reg[] = { - 0x0090, 0x8000, 0x8000, 0x8000, /* 0 - 6 */ - 0x0000, 0x0000, 0x8008, 0x8008, /* 8 - e */ - 0x8808, 0x8808, 0x0000, 0x8808, /* 10 - 16 */ - 0x8808, 0x0000, 0x8000, 0x0000, /* 18 - 1e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 20 - 26 */ - 0x03c7, 0x0000, 0xbb80, 0xbb80, /* 28 - 2e */ - 0xbb80, 0xbb80, 0x0000, 0x8080, /* 30 - 36 */ - 0x8080, 0x2000, 0x0000, 0x0000, /* 38 - 3e */ - 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ - 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ - 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ - 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ - 0x8080, 0x0000, 0x0000, 0x0000, /* 60 - 66 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* reserved */ - 0x0000, 0x0000, 0x1001, 0x0000, /* 70 - 76 */ - 0x0000, 0x0000, 0x4144, 0x5370 /* 78 - 7e */ +static const struct reg_default ad1980_reg_defaults[] = { + { 0x02, 0x8000 }, + { 0x04, 0x8000 }, + { 0x06, 0x8000 }, + { 0x0c, 0x8008 }, + { 0x0e, 0x8008 }, + { 0x10, 0x8808 }, + { 0x12, 0x8808 }, + { 0x16, 0x8808 }, + { 0x18, 0x8808 }, + { 0x1a, 0x0000 }, + { 0x1c, 0x8000 }, + { 0x20, 0x0000 }, + { 0x28, 0x03c7 }, + { 0x2c, 0xbb80 }, + { 0x2e, 0xbb80 }, + { 0x30, 0xbb80 }, + { 0x32, 0xbb80 }, + { 0x36, 0x8080 }, + { 0x38, 0x8080 }, + { 0x3a, 0x2000 }, + { 0x60, 0x0000 }, + { 0x62, 0x0000 }, + { 0x72, 0x0000 }, + { 0x74, 0x1001 }, + { 0x76, 0x0000 }, +}; + +static bool ad1980_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AC97_RESET ... AC97_MASTER_MONO: + case AC97_PHONE ... AC97_CD: + case AC97_AUX ... AC97_GENERAL_PURPOSE: + case AC97_POWERDOWN ... AC97_PCM_LR_ADC_RATE: + case AC97_SPDIF: + case AC97_CODEC_CLASS_REV: + case AC97_PCI_SVID: + case AC97_AD_CODEC_CFG: + case AC97_AD_JACK_SPDIF: + case AC97_AD_SERIAL_CFG: + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return true; + default: + return false; + } +} + +static bool ad1980_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return false; + default: + return ad1980_readable_reg(dev, reg); + } +} + +static const struct regmap_config ad1980_regmap_config = { + .reg_bits = 16, + .reg_stride = 2, + .val_bits = 16, + .max_register = 0x7e, + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = regmap_ac97_default_volatile, + .readable_reg = ad1980_readable_reg, + .writeable_reg = ad1980_writeable_reg, + + .reg_defaults = ad1980_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ad1980_reg_defaults), }; static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", @@ -135,39 +189,13 @@ static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - switch (reg) { - case AC97_RESET: - case AC97_INT_PAGING: - case AC97_POWERDOWN: - case AC97_EXTENDED_STATUS: - case AC97_VENDOR_ID1: - case AC97_VENDOR_ID2: - return soc_ac97_ops->read(ac97, reg); - default: - reg = reg >> 1; - - if (reg >= ARRAY_SIZE(ad1980_reg)) - return -EINVAL; - - return cache[reg]; - } + return snd_soc_read(codec, reg); } static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - - soc_ac97_ops->write(ac97, reg, val); - reg = reg >> 1; - if (reg < ARRAY_SIZE(ad1980_reg)) - cache[reg] = val; - - return 0; + return snd_soc_write(codec, reg, val); } static struct snd_soc_dai_driver ad1980_dai = { @@ -219,6 +247,7 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) static int ad1980_soc_probe(struct snd_soc_codec *codec) { struct snd_ac97 *ac97; + struct regmap *regmap; int ret; u16 vendor_id2; u16 ext_status; @@ -230,6 +259,13 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) return ret; } + regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + goto err_free_ac97; + } + + snd_soc_codec_init_regmap(codec, regmap); snd_soc_codec_set_drvdata(codec, ac97); ret = ad1980_reset(codec, 0); @@ -268,6 +304,8 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) return 0; reset_err: + snd_soc_codec_exit_regmap(codec); +err_free_ac97: snd_soc_free_ac97_codec(ac97); return ret; } @@ -276,6 +314,7 @@ static int ad1980_soc_remove(struct snd_soc_codec *codec) { struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(ac97); return 0; } @@ -283,12 +322,6 @@ static int ad1980_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .probe = ad1980_soc_probe, .remove = ad1980_soc_remove, - .reg_cache_size = ARRAY_SIZE(ad1980_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = ad1980_reg, - .reg_cache_step = 2, - .write = ac97_write, - .read = ac97_read, .controls = ad1980_snd_ac97_controls, .num_controls = ARRAY_SIZE(ad1980_snd_ac97_controls), -- cgit v1.1 From 17bb577328a00e7251c8e552471b6583173ca77d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 18 Nov 2014 19:45:54 +0100 Subject: ASoC: ad1980: Remove ac97_read/ac97_write wrappers Since the regmap conversion ac97_read/ac97_write are just simple wrappers around snd_soc_read/snd_soc_write. Use those instead directly and remove the wrappers. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 36 ++++++++++++------------------------ 1 file changed, 12 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 5fd4a29..2860eef 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,18 +186,6 @@ static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { { "HP_OUT_R", NULL, "Playback" }, }; -static unsigned int ac97_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return snd_soc_read(codec, reg); -} - -static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - return snd_soc_write(codec, reg, val); -} - static struct snd_soc_dai_driver ad1980_dai = { .name = "ad1980-hifi", .playback = { @@ -222,7 +210,7 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) do { if (try_warm && soc_ac97_ops->warm_reset) { soc_ac97_ops->warm_reset(ac97); - if (ac97_read(codec, AC97_RESET) == 0x0090) + if (snd_soc_read(codec, AC97_RESET) == 0x0090) return 1; } @@ -233,9 +221,9 @@ static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) * case the first nibble of data is eaten by the addr. (Tag is * always 16 bit) */ - ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); + snd_soc_write(codec, AC97_AD_SERIAL_CFG, 0x9900); - if (ac97_read(codec, AC97_RESET) == 0x0090) + if (snd_soc_read(codec, AC97_RESET) == 0x0090) return 0; } while (retry_cnt++ < 10); @@ -273,12 +261,12 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) goto reset_err; /* Read out vendor ID to make sure it is ad1980 */ - if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) { + if (snd_soc_read(codec, AC97_VENDOR_ID1) != 0x4144) { ret = -ENODEV; goto reset_err; } - vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2); + vendor_id2 = snd_soc_read(codec, AC97_VENDOR_ID2); if (vendor_id2 != 0x5370) { if (vendor_id2 != 0x5374) { @@ -291,15 +279,15 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) } /* unmute captures and playbacks volume */ - ac97_write(codec, AC97_MASTER, 0x0000); - ac97_write(codec, AC97_PCM, 0x0000); - ac97_write(codec, AC97_REC_GAIN, 0x0000); - ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); - ac97_write(codec, AC97_SURROUND_MASTER, 0x0000); + snd_soc_write(codec, AC97_MASTER, 0x0000); + snd_soc_write(codec, AC97_PCM, 0x0000); + snd_soc_write(codec, AC97_REC_GAIN, 0x0000); + snd_soc_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); + snd_soc_write(codec, AC97_SURROUND_MASTER, 0x0000); /*power on LFE/CENTER/Surround DACs*/ - ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); - ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); + ext_status = snd_soc_read(codec, AC97_EXTENDED_STATUS); + snd_soc_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); return 0; -- cgit v1.1 From 92a6e2a227da5fcaa5b31c9124eabf8c64a6d9f9 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 19 Nov 2014 15:13:26 +0530 Subject: ASoC: Intel: cleanup runtime_pm initialization For ACPI we missed to pm_runtime_enable() call which is required to tell PM core that runtime on this device is enabled now. Since this is common to both PCI and APCI move it out. Also for ACPI we do not require pm_runtime_allow() call, so remove that Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b2b5604..9e68a7c 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -378,13 +378,13 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ - if (acpi_disabled) { + pm_runtime_enable(ctx->dev); + + if (acpi_disabled) pm_runtime_set_active(ctx->dev); - pm_runtime_enable(ctx->dev); - } else { - pm_runtime_allow(ctx->dev); + else pm_runtime_put_noidle(ctx->dev); - } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); -- cgit v1.1 From 996cc8494d663cb03c5ec23ced0e09e4bcd845de Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Wed, 19 Nov 2014 15:13:27 +0530 Subject: ASoC: Intel: add BYTCR machine driver with RT5640 Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/bytcr_dpcm_rt5640.c | 230 ++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a992e85..a26e8e8 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -86,3 +86,15 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_BYTCR_RT5640_MACH + tristate "ASoC Audio DSP Support for MID BYT Platform" + depends on X86 + select SND_SOC_RT5640 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) MID Baytrail platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..fbde4b07 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -26,11 +26,13 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c new file mode 100644 index 0000000..f5d0fc1 --- /dev/null +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -0,0 +1,230 @@ +/* + * byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/rt5640.h" +#include "sst-atom-controls.h" + +static const struct snd_soc_dapm_widget byt_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_audio_map[] = { + {"IN2P", NULL, "Headset Mic"}, + {"IN2N", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "MICBIAS1"}, + {"IN1P", NULL, "MICBIAS1"}, + {"LDO2", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new byt_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int byt_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + snd_soc_dai_set_bclk_ratio(codec_dai, 50); + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, + params_rate(params) * 50, + params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream byt_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int byt_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops byt_aif1_ops = { + .startup = byt_aif1_startup, +}; + +static struct snd_soc_ops byt_be_ssp2_ops = { + .hw_params = byt_aif1_hw_params, +}; + +static struct snd_soc_dai_link byt_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Baytrail Audio Port", + .stream_name = "Baytrail Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Baytrail Compressed Port", + .stream_name = "Baytrail Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = byt_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_byt = { + .name = "baytrailcraudio", + .dai_link = byt_dailink, + .num_links = ARRAY_SIZE(byt_dailink), + .dapm_widgets = byt_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets), + .dapm_routes = byt_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_audio_map), + .controls = byt_mc_controls, + .num_controls = ARRAY_SIZE(byt_mc_controls), +}; + +static int snd_byt_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_byt.dev = &pdev->dev; + + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt); + if (ret_val) { + dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_byt); + return ret_val; +} + +static struct platform_driver snd_byt_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "bytt100_rt5640", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_byt_mc_probe, +}; + +module_platform_driver(snd_byt_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytrt5640-audio"); -- cgit v1.1 From 50c0f21b42dd4cd02b51f82274f66912d9a7fa32 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Nov 2014 18:29:02 +0100 Subject: ASoC: sigmadsp: Refuse to load firmware files with a non-supported version Make sure to check the version field of the firmware header to make sure to not accidentally try to parse a firmware file with a different layout. Trying to do so can result in loading invalid firmware code to the device. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sigmadsp.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index f2de7e0..81a38dd 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -159,6 +159,13 @@ int _process_sigma_firmware(struct device *dev, goto done; } + if (ssfw_head->version != 1) { + dev_err(dev, + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + ssfw_head->version); + goto done; + } + crc = crc32(0, fw->data + sizeof(*ssfw_head), fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); -- cgit v1.1 From 6b25730f68073ee95079d241ea6aa7be00805254 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Nov 2014 18:29:04 +0100 Subject: ASoC: sigmadsp: Drop support support SIGMA_ACTION_DELAY The official firmware generation tool never emitted any SIGMA_ACTION_DELAY instructions. Keeping support for it with the new restructured loader that also supports v2 will be difficult, so drop support for it. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 81a38dd..4fd3143 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -7,7 +7,6 @@ */ #include -#include #include #include #include @@ -28,9 +27,6 @@ enum { SIGMA_ACTION_WRITEXBYTES = 0, SIGMA_ACTION_WRITESINGLE, SIGMA_ACTION_WRITESAFELOAD, - SIGMA_ACTION_DELAY, - SIGMA_ACTION_PLLWAIT, - SIGMA_ACTION_NOOP, SIGMA_ACTION_END, }; @@ -79,10 +75,6 @@ process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa) if (ret < 0) return -EINVAL; break; - case SIGMA_ACTION_DELAY: - udelay(len); - len = 0; - break; case SIGMA_ACTION_END: return 0; default: -- cgit v1.1 From d48b088e3ec45eeccf0fce0b75378e41428f47e9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Nov 2014 18:29:05 +0100 Subject: ASoC: sigmadsp: Restructure in preparation for fw v2 support The v2 file format of the SigmaDSP takes a more declarative style compared to the imperative style of the v1 format. In addition some features that are supported with v2 require the driver to keep state around for the firmware. This requires a bit of restructuring of both the firmware loader itself and the drivers making use of the firmware loader. Instead of loading and executing the firmware in place when the DSP is configured the firmware is now loaded at driver probe time. This is required since the new firmware format will in addition to the firmware data itself contain meta information describing the firmware and its requirements and capabilities. Those will for example be used to restrict the supported samplerates advertised by the driver to userspace to the list of samplerates supported for the firmware. This only does the restructuring required by the v2 format, but does not yet add support for the new format itself. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 33 ++++- sound/soc/codecs/adau1761.c | 21 ++- sound/soc/codecs/adau1781.c | 30 ++--- sound/soc/codecs/adau17x1.c | 54 +++++++- sound/soc/codecs/adau17x1.h | 9 +- sound/soc/codecs/sigmadsp-i2c.c | 52 ++++++-- sound/soc/codecs/sigmadsp-regmap.c | 38 ++++-- sound/soc/codecs/sigmadsp.c | 255 +++++++++++++++++++++++++++++++------ sound/soc/codecs/sigmadsp.h | 53 +++++--- 9 files changed, 424 insertions(+), 121 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 370b742..05d5eb5 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -103,6 +103,8 @@ struct adau1701 { unsigned int sysclk; struct regmap *regmap; u8 pin_config[12]; + + struct sigmadsp *sigmadsp; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -238,12 +240,14 @@ static int adau1701_reg_read(void *context, unsigned int reg, return 0; } -static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) +static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv, + unsigned int rate) { struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *client = to_i2c_client(codec->dev); int ret; + sigmadsp_reset(adau1701->sigmadsp); + if (clkdiv != ADAU1707_CLKDIV_UNSET && gpio_is_valid(adau1701->gpio_pll_mode[0]) && gpio_is_valid(adau1701->gpio_pll_mode[1])) { @@ -284,7 +288,7 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) * know the correct PLL setup */ if (clkdiv != ADAU1707_CLKDIV_UNSET) { - ret = process_sigma_firmware(client, ADAU1701_FIRMWARE); + ret = sigmadsp_setup(adau1701->sigmadsp, rate); if (ret) { dev_warn(codec->dev, "Failed to load firmware\n"); return ret; @@ -385,7 +389,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, * firmware upload. */ if (clkdiv != adau1701->pll_clkdiv) { - ret = adau1701_reset(codec, clkdiv); + ret = adau1701_reset(codec, clkdiv, params_rate(params)); if (ret < 0) return ret; } @@ -554,6 +558,14 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return 0; } +static int adau1701_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(dai->codec); + + return sigmadsp_restrict_params(adau1701->sigmadsp, substream); +} + #define ADAU1701_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ SNDRV_PCM_RATE_192000) @@ -564,6 +576,7 @@ static const struct snd_soc_dai_ops adau1701_dai_ops = { .set_fmt = adau1701_set_dai_fmt, .hw_params = adau1701_hw_params, .digital_mute = adau1701_digital_mute, + .startup = adau1701_startup, }; static struct snd_soc_dai_driver adau1701_dai = { @@ -600,6 +613,10 @@ static int adau1701_probe(struct snd_soc_codec *codec) unsigned int val; struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + ret = sigmadsp_attach(adau1701->sigmadsp, &codec->component); + if (ret) + return ret; + /* * Let the pll_clkdiv variable default to something that won't happen * at runtime. That way, we can postpone the firmware download from @@ -609,7 +626,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) adau1701->pll_clkdiv = ADAU1707_CLKDIV_UNSET; /* initalize with pre-configured pll mode settings */ - ret = adau1701_reset(codec, adau1701->pll_clkdiv); + ret = adau1701_reset(codec, adau1701->pll_clkdiv, 0); if (ret < 0) return ret; @@ -722,6 +739,12 @@ static int adau1701_i2c_probe(struct i2c_client *client, adau1701->gpio_pll_mode[1] = gpio_pll_mode[1]; i2c_set_clientdata(client, adau1701); + + adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, NULL, + ADAU1701_FIRMWARE); + if (IS_ERR(adau1701->sigmadsp)) + return PTR_ERR(adau1701->sigmadsp); + ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); return ret; diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5518ebd..0ae1501 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -698,11 +698,6 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1761_dapm_routes)); if (ret) return ret; - - ret = adau17x1_load_firmware(adau, codec->dev, - ADAU1761_FIRMWARE); - if (ret) - dev_warn(codec->dev, "Failed to firmware\n"); } ret = adau17x1_add_routes(codec); @@ -771,16 +766,20 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)) { struct snd_soc_dai_driver *dai_drv; + const char *firmware_name; int ret; - ret = adau17x1_probe(dev, regmap, type, switch_mode); - if (ret) - return ret; - - if (type == ADAU1361) + if (type == ADAU1361) { dai_drv = &adau1361_dai_driver; - else + firmware_name = NULL; + } else { dai_drv = &adau1761_dai_driver; + firmware_name = ADAU1761_FIRMWARE; + } + + ret = adau17x1_probe(dev, regmap, type, switch_mode, firmware_name); + if (ret) + return ret; return snd_soc_register_codec(dev, &adau1761_codec_driver, dai_drv, 1); } diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index e9fc00f..4c8ddc3 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -385,7 +385,6 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) { struct adau1781_platform_data *pdata = dev_get_platdata(codec->dev); struct adau *adau = snd_soc_codec_get_drvdata(codec); - const char *firmware; int ret; ret = adau17x1_add_widgets(codec); @@ -422,25 +421,10 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec) return ret; } - switch (adau->type) { - case ADAU1381: - firmware = ADAU1381_FIRMWARE; - break; - case ADAU1781: - firmware = ADAU1781_FIRMWARE; - break; - default: - return -EINVAL; - } - ret = adau17x1_add_routes(codec); if (ret < 0) return ret; - ret = adau17x1_load_firmware(adau, codec->dev, firmware); - if (ret) - dev_warn(codec->dev, "Failed to load firmware\n"); - return 0; } @@ -495,9 +479,21 @@ EXPORT_SYMBOL_GPL(adau1781_regmap_config); int adau1781_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)) { + const char *firmware_name; int ret; - ret = adau17x1_probe(dev, regmap, type, switch_mode); + switch (type) { + case ADAU1381: + firmware_name = ADAU1381_FIRMWARE; + break; + case ADAU1781: + firmware_name = ADAU1781_FIRMWARE; + break; + default: + return -EINVAL; + } + + ret = adau17x1_probe(dev, regmap, type, switch_mode, firmware_name); if (ret) return ret; diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 3e16c1c..1cab34c 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -307,6 +307,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct adau *adau = snd_soc_codec_get_drvdata(codec); unsigned int val, div, dsp_div; unsigned int freq; + int ret; if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) freq = adau->pll_freq; @@ -356,6 +357,12 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dsp_div); } + if (adau->sigmadsp) { + ret = adau17x1_setup_firmware(adau, params_rate(params)); + if (ret < 0) + return ret; + } + if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J) return 0; @@ -661,12 +668,24 @@ static int adau17x1_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int adau17x1_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + + if (adau->sigmadsp) + return sigmadsp_restrict_params(adau->sigmadsp, substream); + + return 0; +} + const struct snd_soc_dai_ops adau17x1_dai_ops = { .hw_params = adau17x1_hw_params, .set_sysclk = adau17x1_set_dai_sysclk, .set_fmt = adau17x1_set_dai_fmt, .set_pll = adau17x1_set_dai_pll, .set_tdm_slot = adau17x1_set_dai_tdm_slot, + .startup = adau17x1_startup, }; EXPORT_SYMBOL_GPL(adau17x1_dai_ops); @@ -745,8 +764,7 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg) } EXPORT_SYMBOL_GPL(adau17x1_volatile_register); -int adau17x1_load_firmware(struct adau *adau, struct device *dev, - const char *firmware) +int adau17x1_setup_firmware(struct adau *adau, unsigned int rate) { int ret; int dspsr; @@ -758,7 +776,7 @@ int adau17x1_load_firmware(struct adau *adau, struct device *dev, regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1); regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf); - ret = process_sigma_firmware_regmap(dev, adau->regmap, firmware); + ret = sigmadsp_setup(adau->sigmadsp, rate); if (ret) { regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0); return ret; @@ -767,7 +785,7 @@ int adau17x1_load_firmware(struct adau *adau, struct device *dev, return 0; } -EXPORT_SYMBOL_GPL(adau17x1_load_firmware); +EXPORT_SYMBOL_GPL(adau17x1_setup_firmware); int adau17x1_add_widgets(struct snd_soc_codec *codec) { @@ -787,8 +805,21 @@ int adau17x1_add_widgets(struct snd_soc_codec *codec) ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dsp_dapm_widgets, ARRAY_SIZE(adau17x1_dsp_dapm_widgets)); + if (ret) + return ret; + + if (!adau->sigmadsp) + return 0; + + ret = sigmadsp_attach(adau->sigmadsp, &codec->component); + if (ret) { + dev_err(codec->dev, "Failed to attach firmware: %d\n", + ret); + return ret; + } } - return ret; + + return 0; } EXPORT_SYMBOL_GPL(adau17x1_add_widgets); @@ -829,7 +860,8 @@ int adau17x1_resume(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(adau17x1_resume); int adau17x1_probe(struct device *dev, struct regmap *regmap, - enum adau17x1_type type, void (*switch_mode)(struct device *dev)) + enum adau17x1_type type, void (*switch_mode)(struct device *dev), + const char *firmware_name) { struct adau *adau; @@ -846,6 +878,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); + if (firmware_name) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, + firmware_name); + if (IS_ERR(adau->sigmadsp)) { + dev_warn(dev, "Could not find firmware file: %ld\n", + PTR_ERR(adau->sigmadsp)); + adau->sigmadsp = NULL; + } + } + if (switch_mode) switch_mode(dev); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e4a557f..6861aa3 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -4,6 +4,8 @@ #include #include +#include "sigmadsp.h" + enum adau17x1_type { ADAU1361, ADAU1761, @@ -42,12 +44,14 @@ struct adau { bool dsp_bypass[2]; struct regmap *regmap; + struct sigmadsp *sigmadsp; }; int adau17x1_add_widgets(struct snd_soc_codec *codec); int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, - enum adau17x1_type type, void (*switch_mode)(struct device *dev)); + enum adau17x1_type type, void (*switch_mode)(struct device *dev), + const char *firmware_name); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); @@ -56,8 +60,7 @@ int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -int adau17x1_load_firmware(struct adau *adau, struct device *dev, - const char *firmware); +int adau17x1_setup_firmware(struct adau *adau, unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index 246081a..bf6a2be 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -6,29 +6,59 @@ * Licensed under the GPL-2 or later. */ -#include #include +#include #include +#include +#include #include "sigmadsp.h" -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) +static int sigmadsp_write_i2c(void *control_data, + unsigned int addr, const uint8_t data[], size_t len) { - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); + uint8_t *buf; + int ret; + + buf = kzalloc(2 + len, GFP_KERNEL | GFP_DMA); + if (!buf) + return -ENOMEM; + + put_unaligned_be16(addr, buf); + memcpy(buf + 2, data, len); + + ret = i2c_master_send(control_data, buf, len + 2); + + kfree(buf); + + return ret; } -int process_sigma_firmware(struct i2c_client *client, const char *name) +/** + * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance + * @client: The parent I2C device + * @ops: The sigmadsp_ops to use for this instance + * @firmware_name: Name of the firmware file to load + * + * Allocates a SigmaDSP instance and loads the specified firmware file. + * + * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error. + */ +struct sigmadsp *devm_sigmadsp_init_i2c(struct i2c_client *client, + const struct sigmadsp_ops *ops, const char *firmware_name) { - struct sigma_firmware ssfw; + struct sigmadsp *sigmadsp; + + sigmadsp = devm_sigmadsp_init(&client->dev, ops, firmware_name); + if (IS_ERR(sigmadsp)) + return sigmadsp; - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; + sigmadsp->control_data = client; + sigmadsp->write = sigmadsp_write_i2c; - return _process_sigma_firmware(&client->dev, &ssfw, name); + return sigmadsp; } -EXPORT_SYMBOL(process_sigma_firmware); +EXPORT_SYMBOL_GPL(devm_sigmadsp_init_i2c); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("SigmaDSP I2C firmware loader"); diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c index f78ed8d..cdc5dda 100644 --- a/sound/soc/codecs/sigmadsp-regmap.c +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -12,24 +12,40 @@ #include "sigmadsp.h" -static int sigma_action_write_regmap(void *control_data, - const struct sigma_action *sa, size_t len) +static int sigmadsp_write_regmap(void *control_data, + unsigned int addr, const uint8_t data[], size_t len) { - return regmap_raw_write(control_data, be16_to_cpu(sa->addr), - sa->payload, len - 2); + return regmap_raw_write(control_data, addr, + data, len); } -int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, - const char *name) +/** + * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance + * @dev: The parent device + * @regmap: Regmap instance to use + * @ops: The sigmadsp_ops to use for this instance + * @firmware_name: Name of the firmware file to load + * + * Allocates a SigmaDSP instance and loads the specified firmware file. + * + * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error. + */ +struct sigmadsp *devm_sigmadsp_init_regmap(struct device *dev, + struct regmap *regmap, const struct sigmadsp_ops *ops, + const char *firmware_name) { - struct sigma_firmware ssfw; + struct sigmadsp *sigmadsp; + + sigmadsp = devm_sigmadsp_init(dev, ops, firmware_name); + if (IS_ERR(sigmadsp)) + return sigmadsp; - ssfw.control_data = regmap; - ssfw.write = sigma_action_write_regmap; + sigmadsp->control_data = regmap; + sigmadsp->write = sigmadsp_write_regmap; - return _process_sigma_firmware(dev, &ssfw, name); + return sigmadsp; } -EXPORT_SYMBOL(process_sigma_firmware_regmap); +EXPORT_SYMBOL_GPL(devm_sigmadsp_init_regmap); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("SigmaDSP regmap firmware loader"); diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4fd3143..34e63b5 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -1,7 +1,7 @@ /* * Load Analog Devices SigmaStudio firmware files * - * Copyright 2009-2011 Analog Devices Inc. + * Copyright 2009-2014 Analog Devices Inc. * * Licensed under the GPL-2 or later. */ @@ -12,11 +12,21 @@ #include #include #include +#include + +#include #include "sigmadsp.h" #define SIGMA_MAGIC "ADISIGM" +struct sigmadsp_data { + struct list_head head; + unsigned int addr; + unsigned int length; + uint8_t data[]; +}; + struct sigma_firmware_header { unsigned char magic[7]; u8 version; @@ -30,6 +40,20 @@ enum { SIGMA_ACTION_END, }; +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +static int sigmadsp_write(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t data[], size_t len) +{ + return sigmadsp->write(sigmadsp->control_data, addr, data, len); +} + static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -58,11 +82,11 @@ static size_t sigma_action_size(struct sigma_action *sa) * Returns a negative error value in case of an error, 0 if processing of * the firmware should be stopped after this action, 1 otherwise. */ -static int -process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa) +static int process_sigma_action(struct sigmadsp *sigmadsp, + struct sigma_action *sa) { size_t len = sigma_action_len(sa); - int ret; + struct sigmadsp_data *data; pr_debug("%s: instr:%i addr:%#x len:%zu\n", __func__, sa->instr, sa->addr, len); @@ -71,9 +95,17 @@ process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa) case SIGMA_ACTION_WRITEXBYTES: case SIGMA_ACTION_WRITESINGLE: case SIGMA_ACTION_WRITESAFELOAD: - ret = ssfw->write(ssfw->control_data, sa, len); - if (ret < 0) + if (len < 3) return -EINVAL; + + data = kzalloc(sizeof(*data) + len - 2, GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->addr = be16_to_cpu(sa->addr); + data->length = len - 2; + memcpy(data->data, sa->payload, data->length); + list_add_tail(&data->head, &sigmadsp->data_list); break; case SIGMA_ACTION_END: return 0; @@ -84,22 +116,24 @@ process_sigma_action(struct sigma_firmware *ssfw, struct sigma_action *sa) return 1; } -static int -process_sigma_actions(struct sigma_firmware *ssfw) +static int sigmadsp_fw_load_v1(struct sigmadsp *sigmadsp, + const struct firmware *fw) { struct sigma_action *sa; - size_t size; + size_t size, pos; int ret; - while (ssfw->pos + sizeof(*sa) <= ssfw->fw->size) { - sa = (struct sigma_action *)(ssfw->fw->data + ssfw->pos); + pos = sizeof(struct sigma_firmware_header); + + while (pos + sizeof(*sa) <= fw->size) { + sa = (struct sigma_action *)(fw->data + pos); size = sigma_action_size(sa); - ssfw->pos += size; - if (ssfw->pos > ssfw->fw->size || size == 0) + pos += size; + if (pos > fw->size || size == 0) break; - ret = process_sigma_action(ssfw, sa); + ret = process_sigma_action(sigmadsp, sa); pr_debug("%s: action returned %i\n", __func__, ret); @@ -107,29 +141,40 @@ process_sigma_actions(struct sigma_firmware *ssfw) return ret; } - if (ssfw->pos != ssfw->fw->size) + if (pos != fw->size) return -EINVAL; return 0; } -int _process_sigma_firmware(struct device *dev, - struct sigma_firmware *ssfw, const char *name) +static void sigmadsp_firmware_release(struct sigmadsp *sigmadsp) { - int ret; - struct sigma_firmware_header *ssfw_head; + struct sigmadsp_data *data, *_data; + + list_for_each_entry_safe(data, _data, &sigmadsp->data_list, head) + kfree(data); + + INIT_LIST_HEAD(&sigmadsp->data_list); +} + +static void devm_sigmadsp_release(struct device *dev, void *res) +{ + sigmadsp_firmware_release((struct sigmadsp *)res); +} + +static int sigmadsp_firmware_load(struct sigmadsp *sigmadsp, const char *name) +{ + const struct sigma_firmware_header *ssfw_head; const struct firmware *fw; + int ret; u32 crc; - pr_debug("%s: loading firmware %s\n", __func__, name); - /* first load the blob */ - ret = request_firmware(&fw, name, dev); + ret = request_firmware(&fw, name, sigmadsp->dev); if (ret) { pr_debug("%s: request_firmware() failed with %i\n", __func__, ret); - return ret; + goto done; } - ssfw->fw = fw; /* then verify the header */ ret = -EINVAL; @@ -141,20 +186,13 @@ int _process_sigma_firmware(struct device *dev, * overflows later in the loading process. */ if (fw->size < sizeof(*ssfw_head) || fw->size >= 0x4000000) { - dev_err(dev, "Failed to load firmware: Invalid size\n"); + dev_err(sigmadsp->dev, "Failed to load firmware: Invalid size\n"); goto done; } ssfw_head = (void *)fw->data; if (memcmp(ssfw_head->magic, SIGMA_MAGIC, ARRAY_SIZE(ssfw_head->magic))) { - dev_err(dev, "Failed to load firmware: Invalid magic\n"); - goto done; - } - - if (ssfw_head->version != 1) { - dev_err(dev, - "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", - ssfw_head->version); + dev_err(sigmadsp->dev, "Failed to load firmware: Invalid magic\n"); goto done; } @@ -162,23 +200,160 @@ int _process_sigma_firmware(struct device *dev, fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); if (crc != le32_to_cpu(ssfw_head->crc)) { - dev_err(dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n", + dev_err(sigmadsp->dev, "Failed to load firmware: Wrong crc checksum: expected %x got %x\n", le32_to_cpu(ssfw_head->crc), crc); goto done; } - ssfw->pos = sizeof(*ssfw_head); + switch (ssfw_head->version) { + case 1: + ret = sigmadsp_fw_load_v1(sigmadsp, fw); + break; + default: + dev_err(sigmadsp->dev, + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + ssfw_head->version); + ret = -EINVAL; + break; + } - /* finally process all of the actions */ - ret = process_sigma_actions(ssfw); + if (ret) + sigmadsp_firmware_release(sigmadsp); - done: +done: release_firmware(fw); - pr_debug("%s: loaded %s\n", __func__, name); + return ret; +} + +static int sigmadsp_init(struct sigmadsp *sigmadsp, struct device *dev, + const struct sigmadsp_ops *ops, const char *firmware_name) +{ + sigmadsp->ops = ops; + sigmadsp->dev = dev; + + INIT_LIST_HEAD(&sigmadsp->data_list); + + return sigmadsp_firmware_load(sigmadsp, firmware_name); +} + +/** + * devm_sigmadsp_init() - Initialize SigmaDSP instance + * @dev: The parent device + * @ops: The sigmadsp_ops to use for this instance + * @firmware_name: Name of the firmware file to load + * + * Allocates a SigmaDSP instance and loads the specified firmware file. + * + * Returns a pointer to a struct sigmadsp on success, or a PTR_ERR() on error. + */ +struct sigmadsp *devm_sigmadsp_init(struct device *dev, + const struct sigmadsp_ops *ops, const char *firmware_name) +{ + struct sigmadsp *sigmadsp; + int ret; + + sigmadsp = devres_alloc(devm_sigmadsp_release, sizeof(*sigmadsp), + GFP_KERNEL); + if (!sigmadsp) + return ERR_PTR(-ENOMEM); + + ret = sigmadsp_init(sigmadsp, dev, ops, firmware_name); + if (ret) { + devres_free(sigmadsp); + return ERR_PTR(ret); + } + + devres_add(dev, sigmadsp); + + return sigmadsp; +} +EXPORT_SYMBOL_GPL(devm_sigmadsp_init); + +/** + * sigmadsp_attach() - Attach a sigmadsp instance to a ASoC component + * @sigmadsp: The sigmadsp instance to attach + * @component: The component to attach to + * + * Typically called in the components probe callback. + * + * Note, once this function has been called the firmware must not be released + * until after the ALSA snd_card that the component belongs to has been + * disconnected, even if sigmadsp_attach() returns an error. + */ +int sigmadsp_attach(struct sigmadsp *sigmadsp, + struct snd_soc_component *component) +{ + sigmadsp->component = component; + + return 0; +} +EXPORT_SYMBOL_GPL(sigmadsp_attach); + +/** + * sigmadsp_setup() - Setup the DSP for the specified samplerate + * @sigmadsp: The sigmadsp instance to configure + * @samplerate: The samplerate the DSP should be configured for + * + * Loads the appropriate firmware program and parameter memory (if not already + * loaded) and enables the controls for the specified samplerate. Any control + * parameter changes that have been made previously will be restored. + * + * Returns 0 on success, a negative error code otherwise. + */ +int sigmadsp_setup(struct sigmadsp *sigmadsp, unsigned int samplerate) +{ + struct sigmadsp_data *data; + int ret; + + if (sigmadsp->current_samplerate == samplerate) + return 0; + + list_for_each_entry(data, &sigmadsp->data_list, head) { + ret = sigmadsp_write(sigmadsp, data->addr, data->data, + data->length); + if (ret) + goto err; + } + + sigmadsp->current_samplerate = samplerate; + + return 0; +err: + sigmadsp_reset(sigmadsp); return ret; } -EXPORT_SYMBOL_GPL(_process_sigma_firmware); +EXPORT_SYMBOL_GPL(sigmadsp_setup); + +/** + * sigmadsp_reset() - Notify the sigmadsp instance that the DSP has been reset + * @sigmadsp: The sigmadsp instance to reset + * + * Should be called whenever the DSP has been reset and parameter and program + * memory need to be re-loaded. + */ +void sigmadsp_reset(struct sigmadsp *sigmadsp) +{ + sigmadsp->current_samplerate = 0; +} +EXPORT_SYMBOL_GPL(sigmadsp_reset); + +/** + * sigmadsp_restrict_params() - Applies DSP firmware specific constraints + * @sigmadsp: The sigmadsp instance + * @substream: The substream to restrict + * + * Applies samplerate constraints that may be required by the firmware Should + * typically be called from the CODEC/component drivers startup callback. + * + * Returns 0 on success, a negative error code otherwise. + */ +int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, + struct snd_pcm_substream *substream) +{ + return 0; +} +EXPORT_SYMBOL_GPL(sigmadsp_restrict_params); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index c47cd23..a6be91a 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -11,31 +11,50 @@ #include #include +#include -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -} __packed; +#include -struct sigma_firmware { - const struct firmware *fw; - size_t pos; +struct sigmadsp; +struct snd_soc_component; +struct snd_pcm_substream; + +struct sigmadsp_ops { + int (*safeload)(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t *data, size_t len); +}; + +struct sigmadsp { + const struct sigmadsp_ops *ops; + + struct list_head data_list; + + unsigned int current_samplerate; + struct snd_soc_component *component; + struct device *dev; void *control_data; - int (*write)(void *control_data, const struct sigma_action *sa, - size_t len); + int (*write)(void *, unsigned int, const uint8_t *, size_t); }; -int _process_sigma_firmware(struct device *dev, - struct sigma_firmware *ssfw, const char *name); +struct sigmadsp *devm_sigmadsp_init(struct device *dev, + const struct sigmadsp_ops *ops, const char *firmware_name); +void sigmadsp_reset(struct sigmadsp *sigmadsp); + +int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, + struct snd_pcm_substream *substream); struct i2c_client; -extern int process_sigma_firmware(struct i2c_client *client, const char *name); -extern int process_sigma_firmware_regmap(struct device *dev, - struct regmap *regmap, const char *name); +struct sigmadsp *devm_sigmadsp_init_regmap(struct device *dev, + struct regmap *regmap, const struct sigmadsp_ops *ops, + const char *firmware_name); +struct sigmadsp *devm_sigmadsp_init_i2c(struct i2c_client *client, + const struct sigmadsp_ops *ops, const char *firmware_name); + +int sigmadsp_attach(struct sigmadsp *sigmadsp, + struct snd_soc_component *component); +int sigmadsp_setup(struct sigmadsp *sigmadsp, unsigned int rate); +void sigmadsp_reset(struct sigmadsp *sigmadsp); #endif -- cgit v1.1 From a35daac77a0397d4f23b642d3dc0684682e56cc5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Nov 2014 18:29:06 +0100 Subject: ASoC: sigmadsp: Add support for fw v2 This patch adds support for the v2 version of the SigmaDSP firmware file format. The new format has support for having different program and parameter settings for different samplerates. In addition it stores metadata describing the firmware. This metadata includes the set of supported samplerates which will be used to restrict the samplerates that can be selected by userspace. Also included is information about the modifiable parameters. Those will be exposed as ALSA controls so they can be changed at runtime. The new format is based on a binary type-length-value structure that makes it both forward and backwards compatible. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp-i2c.c | 29 +++ sound/soc/codecs/sigmadsp-regmap.c | 8 + sound/soc/codecs/sigmadsp.c | 463 ++++++++++++++++++++++++++++++++++++- sound/soc/codecs/sigmadsp.h | 6 + 4 files changed, 504 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c index bf6a2be..21ca3a5 100644 --- a/sound/soc/codecs/sigmadsp-i2c.c +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -34,6 +34,34 @@ static int sigmadsp_write_i2c(void *control_data, return ret; } +static int sigmadsp_read_i2c(void *control_data, + unsigned int addr, uint8_t data[], size_t len) +{ + struct i2c_client *client = control_data; + struct i2c_msg msgs[2]; + uint8_t buf[2]; + int ret; + + put_unaligned_be16(addr, buf); + + msgs[0].addr = client->addr; + msgs[0].len = sizeof(buf); + msgs[0].buf = buf; + msgs[0].flags = 0; + + msgs[1].addr = client->addr; + msgs[1].len = len; + msgs[1].buf = data; + msgs[1].flags = I2C_M_RD; + + ret = i2c_transfer(client->adapter, msgs, ARRAY_SIZE(msgs)); + if (ret < 0) + return ret; + else if (ret != ARRAY_SIZE(msgs)) + return -EIO; + return 0; +} + /** * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance * @client: The parent I2C device @@ -55,6 +83,7 @@ struct sigmadsp *devm_sigmadsp_init_i2c(struct i2c_client *client, sigmadsp->control_data = client; sigmadsp->write = sigmadsp_write_i2c; + sigmadsp->read = sigmadsp_read_i2c; return sigmadsp; } diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c index cdc5dda..912861b 100644 --- a/sound/soc/codecs/sigmadsp-regmap.c +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -19,6 +19,13 @@ static int sigmadsp_write_regmap(void *control_data, data, len); } +static int sigmadsp_read_regmap(void *control_data, + unsigned int addr, uint8_t data[], size_t len) +{ + return regmap_raw_read(control_data, addr, + data, len); +} + /** * devm_sigmadsp_init_i2c() - Initialize SigmaDSP instance * @dev: The parent device @@ -42,6 +49,7 @@ struct sigmadsp *devm_sigmadsp_init_regmap(struct device *dev, sigmadsp->control_data = regmap; sigmadsp->write = sigmadsp_write_regmap; + sigmadsp->read = sigmadsp_read_regmap; return sigmadsp; } diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 34e63b5..55af596 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -14,19 +14,61 @@ #include #include +#include #include #include "sigmadsp.h" #define SIGMA_MAGIC "ADISIGM" +#define SIGMA_FW_CHUNK_TYPE_DATA 0 +#define SIGMA_FW_CHUNK_TYPE_CONTROL 1 +#define SIGMA_FW_CHUNK_TYPE_SAMPLERATES 2 + +struct sigmadsp_control { + struct list_head head; + uint32_t samplerates; + unsigned int addr; + unsigned int num_bytes; + const char *name; + struct snd_kcontrol *kcontrol; + bool cached; + uint8_t cache[]; +}; + struct sigmadsp_data { struct list_head head; + uint32_t samplerates; unsigned int addr; unsigned int length; uint8_t data[]; }; +struct sigma_fw_chunk { + __le32 length; + __le32 tag; + __le32 samplerates; +} __packed; + +struct sigma_fw_chunk_data { + struct sigma_fw_chunk chunk; + __le16 addr; + uint8_t data[]; +} __packed; + +struct sigma_fw_chunk_control { + struct sigma_fw_chunk chunk; + __le16 type; + __le16 addr; + __le16 num_bytes; + const char name[]; +} __packed; + +struct sigma_fw_chunk_samplerate { + struct sigma_fw_chunk chunk; + __le32 samplerates[]; +} __packed; + struct sigma_firmware_header { unsigned char magic[7]; u8 version; @@ -54,6 +96,269 @@ static int sigmadsp_write(struct sigmadsp *sigmadsp, unsigned int addr, return sigmadsp->write(sigmadsp->control_data, addr, data, len); } +static int sigmadsp_read(struct sigmadsp *sigmadsp, unsigned int addr, + uint8_t data[], size_t len) +{ + return sigmadsp->read(sigmadsp->control_data, addr, data, len); +} + +static int sigmadsp_ctrl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + struct sigmadsp_control *ctrl = (void *)kcontrol->private_value; + + info->type = SNDRV_CTL_ELEM_TYPE_BYTES; + info->count = ctrl->num_bytes; + + return 0; +} + +static int sigmadsp_ctrl_write(struct sigmadsp *sigmadsp, + struct sigmadsp_control *ctrl, void *data) +{ + /* safeload loads up to 20 bytes in a atomic operation */ + if (ctrl->num_bytes > 4 && ctrl->num_bytes <= 20 && sigmadsp->ops && + sigmadsp->ops->safeload) + return sigmadsp->ops->safeload(sigmadsp, ctrl->addr, data, + ctrl->num_bytes); + else + return sigmadsp_write(sigmadsp, ctrl->addr, data, + ctrl->num_bytes); +} + +static int sigmadsp_ctrl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sigmadsp_control *ctrl = (void *)kcontrol->private_value; + struct sigmadsp *sigmadsp = snd_kcontrol_chip(kcontrol); + uint8_t *data; + int ret = 0; + + mutex_lock(&sigmadsp->lock); + + data = ucontrol->value.bytes.data; + + if (!(kcontrol->vd[0].access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) + ret = sigmadsp_ctrl_write(sigmadsp, ctrl, data); + + if (ret == 0) { + memcpy(ctrl->cache, data, ctrl->num_bytes); + ctrl->cached = true; + } + + mutex_unlock(&sigmadsp->lock); + + return ret; +} + +static int sigmadsp_ctrl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct sigmadsp_control *ctrl = (void *)kcontrol->private_value; + struct sigmadsp *sigmadsp = snd_kcontrol_chip(kcontrol); + int ret = 0; + + mutex_lock(&sigmadsp->lock); + + if (!ctrl->cached) { + ret = sigmadsp_read(sigmadsp, ctrl->addr, ctrl->cache, + ctrl->num_bytes); + } + + if (ret == 0) { + ctrl->cached = true; + memcpy(ucontrol->value.bytes.data, ctrl->cache, + ctrl->num_bytes); + } + + mutex_unlock(&sigmadsp->lock); + + return ret; +} + +static void sigmadsp_control_free(struct snd_kcontrol *kcontrol) +{ + struct sigmadsp_control *ctrl = (void *)kcontrol->private_value; + + ctrl->kcontrol = NULL; +} + +static bool sigma_fw_validate_control_name(const char *name, unsigned int len) +{ + unsigned int i; + + for (i = 0; i < len; i++) { + /* Normal ASCII characters are valid */ + if (name[i] < ' ' || name[i] > '~') + return false; + } + + return true; +} + +static int sigma_fw_load_control(struct sigmadsp *sigmadsp, + const struct sigma_fw_chunk *chunk, unsigned int length) +{ + const struct sigma_fw_chunk_control *ctrl_chunk; + struct sigmadsp_control *ctrl; + unsigned int num_bytes; + size_t name_len; + char *name; + int ret; + + if (length <= sizeof(*ctrl_chunk)) + return -EINVAL; + + ctrl_chunk = (const struct sigma_fw_chunk_control *)chunk; + + name_len = length - sizeof(*ctrl_chunk); + if (name_len >= SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + name_len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1; + + /* Make sure there are no non-displayable characaters in the string */ + if (!sigma_fw_validate_control_name(ctrl_chunk->name, name_len)) + return -EINVAL; + + num_bytes = le16_to_cpu(ctrl_chunk->num_bytes); + ctrl = kzalloc(sizeof(*ctrl) + num_bytes, GFP_KERNEL); + if (!ctrl) + return -ENOMEM; + + name = kzalloc(name_len + 1, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto err_free_ctrl; + } + memcpy(name, ctrl_chunk->name, name_len); + name[name_len] = '\0'; + ctrl->name = name; + + ctrl->addr = le16_to_cpu(ctrl_chunk->addr); + ctrl->num_bytes = num_bytes; + ctrl->samplerates = chunk->samplerates; + + list_add_tail(&ctrl->head, &sigmadsp->ctrl_list); + + return 0; + +err_free_ctrl: + kfree(ctrl); + + return ret; +} + +static int sigma_fw_load_data(struct sigmadsp *sigmadsp, + const struct sigma_fw_chunk *chunk, unsigned int length) +{ + const struct sigma_fw_chunk_data *data_chunk; + struct sigmadsp_data *data; + + if (length <= sizeof(*data_chunk)) + return -EINVAL; + + data_chunk = (struct sigma_fw_chunk_data *)chunk; + + length -= sizeof(*data_chunk); + + data = kzalloc(sizeof(*data) + length, GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->addr = le16_to_cpu(data_chunk->addr); + data->length = length; + data->samplerates = chunk->samplerates; + memcpy(data->data, data_chunk->data, length); + list_add_tail(&data->head, &sigmadsp->data_list); + + return 0; +} + +static int sigma_fw_load_samplerates(struct sigmadsp *sigmadsp, + const struct sigma_fw_chunk *chunk, unsigned int length) +{ + const struct sigma_fw_chunk_samplerate *rate_chunk; + unsigned int num_rates; + unsigned int *rates; + unsigned int i; + + rate_chunk = (const struct sigma_fw_chunk_samplerate *)chunk; + + num_rates = (length - sizeof(*rate_chunk)) / sizeof(__le32); + + if (num_rates > 32 || num_rates == 0) + return -EINVAL; + + /* We only allow one samplerates block per file */ + if (sigmadsp->rate_constraints.count) + return -EINVAL; + + rates = kcalloc(num_rates, sizeof(*rates), GFP_KERNEL); + if (!rates) + return -ENOMEM; + + for (i = 0; i < num_rates; i++) + rates[i] = le32_to_cpu(rate_chunk->samplerates[i]); + + sigmadsp->rate_constraints.count = num_rates; + sigmadsp->rate_constraints.list = rates; + + return 0; +} + +static int sigmadsp_fw_load_v2(struct sigmadsp *sigmadsp, + const struct firmware *fw) +{ + struct sigma_fw_chunk *chunk; + unsigned int length, pos; + int ret; + + /* + * Make sure that there is at least one chunk to avoid integer + * underflows later on. Empty firmware is still valid though. + */ + if (fw->size < sizeof(*chunk) + sizeof(struct sigma_firmware_header)) + return 0; + + pos = sizeof(struct sigma_firmware_header); + + while (pos < fw->size - sizeof(*chunk)) { + chunk = (struct sigma_fw_chunk *)(fw->data + pos); + + length = le32_to_cpu(chunk->length); + + if (length > fw->size - pos || length < sizeof(*chunk)) + return -EINVAL; + + switch (chunk->tag) { + case SIGMA_FW_CHUNK_TYPE_DATA: + ret = sigma_fw_load_data(sigmadsp, chunk, length); + break; + case SIGMA_FW_CHUNK_TYPE_CONTROL: + ret = sigma_fw_load_control(sigmadsp, chunk, length); + break; + case SIGMA_FW_CHUNK_TYPE_SAMPLERATES: + ret = sigma_fw_load_samplerates(sigmadsp, chunk, length); + break; + default: + dev_warn(sigmadsp->dev, "Unknown chunk type: %d\n", + chunk->tag); + ret = 0; + break; + } + + if (ret) + return ret; + + /* + * This can not overflow since if length is larger than the + * maximum firmware size (0x4000000) we'll error out earilier. + */ + pos += ALIGN(length, sizeof(__le32)); + } + + return 0; +} + static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -149,11 +454,18 @@ static int sigmadsp_fw_load_v1(struct sigmadsp *sigmadsp, static void sigmadsp_firmware_release(struct sigmadsp *sigmadsp) { + struct sigmadsp_control *ctrl, *_ctrl; struct sigmadsp_data *data, *_data; + list_for_each_entry_safe(ctrl, _ctrl, &sigmadsp->ctrl_list, head) { + kfree(ctrl->name); + kfree(ctrl); + } + list_for_each_entry_safe(data, _data, &sigmadsp->data_list, head) kfree(data); + INIT_LIST_HEAD(&sigmadsp->ctrl_list); INIT_LIST_HEAD(&sigmadsp->data_list); } @@ -209,9 +521,12 @@ static int sigmadsp_firmware_load(struct sigmadsp *sigmadsp, const char *name) case 1: ret = sigmadsp_fw_load_v1(sigmadsp, fw); break; + case 2: + ret = sigmadsp_fw_load_v2(sigmadsp, fw); + break; default: dev_err(sigmadsp->dev, - "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1, 2\n", ssfw_head->version); ret = -EINVAL; break; @@ -232,7 +547,9 @@ static int sigmadsp_init(struct sigmadsp *sigmadsp, struct device *dev, sigmadsp->ops = ops; sigmadsp->dev = dev; + INIT_LIST_HEAD(&sigmadsp->ctrl_list); INIT_LIST_HEAD(&sigmadsp->data_list); + mutex_init(&sigmadsp->lock); return sigmadsp_firmware_load(sigmadsp, firmware_name); } @@ -270,6 +587,114 @@ struct sigmadsp *devm_sigmadsp_init(struct device *dev, } EXPORT_SYMBOL_GPL(devm_sigmadsp_init); +static int sigmadsp_rate_to_index(struct sigmadsp *sigmadsp, unsigned int rate) +{ + unsigned int i; + + for (i = 0; i < sigmadsp->rate_constraints.count; i++) { + if (sigmadsp->rate_constraints.list[i] == rate) + return i; + } + + return -EINVAL; +} + +static unsigned int sigmadsp_get_samplerate_mask(struct sigmadsp *sigmadsp, + unsigned int samplerate) +{ + int samplerate_index; + + if (samplerate == 0) + return 0; + + if (sigmadsp->rate_constraints.count) { + samplerate_index = sigmadsp_rate_to_index(sigmadsp, samplerate); + if (samplerate_index < 0) + return 0; + + return BIT(samplerate_index); + } else { + return ~0; + } +} + +static bool sigmadsp_samplerate_valid(unsigned int supported, + unsigned int requested) +{ + /* All samplerates are supported */ + if (!supported) + return true; + + return supported & requested; +} + +static int sigmadsp_alloc_control(struct sigmadsp *sigmadsp, + struct sigmadsp_control *ctrl, unsigned int samplerate_mask) +{ + struct snd_kcontrol_new template; + struct snd_kcontrol *kcontrol; + int ret; + + memset(&template, 0, sizeof(template)); + template.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + template.name = ctrl->name; + template.info = sigmadsp_ctrl_info; + template.get = sigmadsp_ctrl_get; + template.put = sigmadsp_ctrl_put; + template.private_value = (unsigned long)ctrl; + template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + if (!sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask)) + template.access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + + kcontrol = snd_ctl_new1(&template, sigmadsp); + if (!kcontrol) + return -ENOMEM; + + kcontrol->private_free = sigmadsp_control_free; + ctrl->kcontrol = kcontrol; + + ret = snd_ctl_add(sigmadsp->component->card->snd_card, kcontrol); + if (ret) + return ret; + + return 0; +} + +static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, + struct sigmadsp_control *ctrl, unsigned int samplerate_mask) +{ + struct snd_card *card = sigmadsp->component->card->snd_card; + struct snd_kcontrol_volatile *vd; + struct snd_ctl_elem_id id; + bool active, changed; + + active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask); + + down_write(&card->controls_rwsem); + if (!ctrl->kcontrol) { + up_write(&card->controls_rwsem); + return; + } + + id = ctrl->kcontrol->id; + vd = &ctrl->kcontrol->vd[0]; + if (active == (bool)(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) { + vd->access ^= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + changed = true; + } + up_write(&card->controls_rwsem); + + if (active && changed) { + mutex_lock(&sigmadsp->lock); + if (ctrl->cached) + sigmadsp_ctrl_write(sigmadsp, ctrl, ctrl->cache); + mutex_unlock(&sigmadsp->lock); + } + + if (changed) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &id); +} + /** * sigmadsp_attach() - Attach a sigmadsp instance to a ASoC component * @sigmadsp: The sigmadsp instance to attach @@ -284,8 +709,21 @@ EXPORT_SYMBOL_GPL(devm_sigmadsp_init); int sigmadsp_attach(struct sigmadsp *sigmadsp, struct snd_soc_component *component) { + struct sigmadsp_control *ctrl; + unsigned int samplerate_mask; + int ret; + sigmadsp->component = component; + samplerate_mask = sigmadsp_get_samplerate_mask(sigmadsp, + sigmadsp->current_samplerate); + + list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head) { + ret = sigmadsp_alloc_control(sigmadsp, ctrl, samplerate_mask); + if (ret) + return ret; + } + return 0; } EXPORT_SYMBOL_GPL(sigmadsp_attach); @@ -303,19 +741,31 @@ EXPORT_SYMBOL_GPL(sigmadsp_attach); */ int sigmadsp_setup(struct sigmadsp *sigmadsp, unsigned int samplerate) { + struct sigmadsp_control *ctrl; + unsigned int samplerate_mask; struct sigmadsp_data *data; int ret; if (sigmadsp->current_samplerate == samplerate) return 0; + samplerate_mask = sigmadsp_get_samplerate_mask(sigmadsp, samplerate); + if (samplerate_mask == 0) + return -EINVAL; + list_for_each_entry(data, &sigmadsp->data_list, head) { + if (!sigmadsp_samplerate_valid(data->samplerates, + samplerate_mask)) + continue; ret = sigmadsp_write(sigmadsp, data->addr, data->data, data->length); if (ret) goto err; } + list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head) + sigmadsp_activate_ctrl(sigmadsp, ctrl, samplerate_mask); + sigmadsp->current_samplerate = samplerate; return 0; @@ -335,6 +785,11 @@ EXPORT_SYMBOL_GPL(sigmadsp_setup); */ void sigmadsp_reset(struct sigmadsp *sigmadsp) { + struct sigmadsp_control *ctrl; + + list_for_each_entry(ctrl, &sigmadsp->ctrl_list, head) + sigmadsp_activate_ctrl(sigmadsp, ctrl, false); + sigmadsp->current_samplerate = 0; } EXPORT_SYMBOL_GPL(sigmadsp_reset); @@ -352,7 +807,11 @@ EXPORT_SYMBOL_GPL(sigmadsp_reset); int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, struct snd_pcm_substream *substream) { - return 0; + if (sigmadsp->rate_constraints.count == 0) + return 0; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &sigmadsp->rate_constraints); } EXPORT_SYMBOL_GPL(sigmadsp_restrict_params); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index a6be91a..614475c 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -27,14 +27,20 @@ struct sigmadsp_ops { struct sigmadsp { const struct sigmadsp_ops *ops; + struct list_head ctrl_list; struct list_head data_list; + struct snd_pcm_hw_constraint_list rate_constraints; + unsigned int current_samplerate; struct snd_soc_component *component; struct device *dev; + struct mutex lock; + void *control_data; int (*write)(void *, unsigned int, const uint8_t *, size_t); + int (*read)(void *, unsigned int, uint8_t *, size_t); }; struct sigmadsp *devm_sigmadsp_init(struct device *dev, -- cgit v1.1 From a3a1ec66d6c9320e676fc99dbaf18db4f8dcda93 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 19 Nov 2014 18:29:07 +0100 Subject: ASoC: adau1701: Implement sigmadsp safeload The safeload feature allows to load up to 5 parameter memory registers atomically. This is helpful for switching between e.g. filter settings without causing any glitches. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 57 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 55 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 05d5eb5..d4e219b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -22,9 +22,14 @@ #include #include +#include + #include "sigmadsp.h" #include "adau1701.h" +#define ADAU1701_SAFELOAD_DATA(i) (0x0810 + (i)) +#define ADAU1701_SAFELOAD_ADDR(i) (0x0815 + (i)) + #define ADAU1701_DSPCTRL 0x081c #define ADAU1701_SEROCTL 0x081e #define ADAU1701_SERICTL 0x081f @@ -42,6 +47,7 @@ #define ADAU1701_DSPCTRL_CR (1 << 2) #define ADAU1701_DSPCTRL_DAM (1 << 3) #define ADAU1701_DSPCTRL_ADM (1 << 4) +#define ADAU1701_DSPCTRL_IST (1 << 5) #define ADAU1701_DSPCTRL_SR_48 0x00 #define ADAU1701_DSPCTRL_SR_96 0x01 #define ADAU1701_DSPCTRL_SR_192 0x02 @@ -102,6 +108,7 @@ struct adau1701 { unsigned int pll_clkdiv; unsigned int sysclk; struct regmap *regmap; + struct i2c_client *client; u8 pin_config[12]; struct sigmadsp *sigmadsp; @@ -161,6 +168,7 @@ static bool adau1701_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case ADAU1701_DACSET: + case ADAU1701_DSPCTRL: return true; default: return false; @@ -240,6 +248,50 @@ static int adau1701_reg_read(void *context, unsigned int reg, return 0; } +static int adau1701_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + struct i2c_client *client = to_i2c_client(sigmadsp->dev); + struct adau1701 *adau1701 = i2c_get_clientdata(client); + unsigned int val; + unsigned int i; + uint8_t buf[10]; + int ret; + + ret = regmap_read(adau1701->regmap, ADAU1701_DSPCTRL, &val); + if (ret) + return ret; + + if (val & ADAU1701_DSPCTRL_IST) + msleep(50); + + for (i = 0; i < len / 4; i++) { + put_unaligned_le16(ADAU1701_SAFELOAD_DATA(i), buf); + buf[2] = 0x00; + memcpy(buf + 3, bytes + i * 4, 4); + ret = i2c_master_send(client, buf, 7); + if (ret < 0) + return ret; + else if (ret != 7) + return -EIO; + + put_unaligned_le16(ADAU1701_SAFELOAD_ADDR(i), buf); + put_unaligned_le16(addr + i, buf + 2); + ret = i2c_master_send(client, buf, 4); + if (ret < 0) + return ret; + else if (ret != 4) + return -EIO; + } + + return regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, + ADAU1701_DSPCTRL_IST, ADAU1701_DSPCTRL_IST); +} + +static const struct sigmadsp_ops adau1701_sigmadsp_ops = { + .safeload = adau1701_safeload, +}; + static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv, unsigned int rate) { @@ -684,6 +736,7 @@ static int adau1701_i2c_probe(struct i2c_client *client, if (!adau1701) return -ENOMEM; + adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, &adau1701_regmap); if (IS_ERR(adau1701->regmap)) @@ -740,8 +793,8 @@ static int adau1701_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, adau1701); - adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, NULL, - ADAU1701_FIRMWARE); + adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, + &adau1701_sigmadsp_ops, ADAU1701_FIRMWARE); if (IS_ERR(adau1701->sigmadsp)) return PTR_ERR(adau1701->sigmadsp); -- cgit v1.1 From 0f32fd1900e6b972f289416dbd75e92772b630cb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 19 Nov 2014 12:16:14 +0100 Subject: ALSA: hda/realtek - Clean up mute/mic GPIO LED handling There are a few duplicated codes handling the mute and mic-mute LEDs via GPIO pins. Let's consolidate to single helpers. Here we introduced two new fields to alc_spec, gpio_mute_led_mask and gpio_mic_led_mask, to contain the bit mask to set/clear. Also, mute_led_polarity is evaluated as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 81 +++++++++++++++++++++---------------------- 1 file changed, 40 insertions(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1af917f..3c29a55 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -96,6 +96,8 @@ struct alc_spec { hda_nid_t cap_mute_led_nid; unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ + unsigned int gpio_mute_led_mask; + unsigned int gpio_mic_led_mask; hda_nid_t headset_mic_pin; hda_nid_t headphone_mic_pin; @@ -3235,41 +3237,45 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, } } -/* turn on/off mute LED per vmaster hook */ -static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) +/* update LED status via GPIO */ +static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, + bool enabled) { - struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; + if (spec->mute_led_polarity) + enabled = !enabled; + if (enabled) - spec->gpio_led &= ~0x08; + spec->gpio_led &= ~mask; else - spec->gpio_led |= 0x08; + spec->gpio_led |= mask; if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); } -/* turn on/off mic-mute LED per capture hook */ -static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* turn on/off mute LED via GPIO per vmaster hook */ +static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) { + struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - unsigned int oldval = spec->gpio_led; - if (!ucontrol) - return; + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); +} - if (ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]) - spec->gpio_led &= ~0x10; - else - spec->gpio_led |= 0x10; - if (spec->gpio_led != oldval) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); +/* turn on/off mic-mute LED via GPIO per capture hook */ +static void alc_fixup_gpio_mic_mute_hook(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct alc_spec *spec = codec->spec; + + if (ucontrol) + alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]); } static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, @@ -3283,9 +3289,12 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, }; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; - spec->gen.cap_sync_hook = alc269_fixup_hp_gpio_mic_mute_hook; + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x08; + spec->gpio_mic_led_mask = 0x10; snd_hda_add_verbs(codec, gpio_init); } } @@ -3327,9 +3336,11 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec, }; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; snd_hda_add_verbs(codec, gpio_init); codec->power_filter = led_power_filter; @@ -3348,9 +3359,11 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, }; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc269_fixup_hp_gpio_mute_hook; + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; spec->gen.cap_sync_hook = alc269_fixup_hp_cap_mic_mute_hook; spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x08; spec->cap_mute_led_nid = 0x18; snd_hda_add_verbs(codec, gpio_init); codec->power_filter = led_power_filter; @@ -5624,22 +5637,6 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec, } } -/* turn on/off mute LED per vmaster hook */ -static void alc662_led_gpio1_mute_hook(void *private_data, int enabled) -{ - struct hda_codec *codec = private_data; - struct alc_spec *spec = codec->spec; - unsigned int oldval = spec->gpio_led; - - if (enabled) - spec->gpio_led |= 0x01; - else - spec->gpio_led &= ~0x01; - if (spec->gpio_led != oldval) - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); -} - /* avoid D3 for keeping GPIO up */ static unsigned int gpio_led_power_filter(struct hda_codec *codec, hda_nid_t nid, @@ -5662,8 +5659,10 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, }; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->gen.vmaster_mute.hook = alc662_led_gpio1_mute_hook; + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; spec->gpio_led = 0; + spec->mute_led_polarity = 1; + spec->gpio_mute_led_mask = 0x01; snd_hda_add_verbs(codec, gpio_init); codec->power_filter = gpio_led_power_filter; } -- cgit v1.1 From 5aeee3424fa5926a53885b9defb593971e446d77 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 11:02:14 +0100 Subject: ALSA: usb-audio: Refactor ignore_ctl_error checks Introduce an internal helper macro for avoiding many open codes. The only slight behavior change is in a couple of get ballcks where the value is reset at error no matter whether ignore_ctl_error is set or not. Actually this is even safer than before. Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 39 ++++++++++++++++----------------------- 1 file changed, 16 insertions(+), 23 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c6209ce..a4fbfff 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -136,6 +136,10 @@ check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) return strlcpy(buf, p->name, buflen); } +/* ignore the error value if ignore_ctl_error flag is set */ +#define filter_error(cval, err) \ + ((cval)->mixer->ignore_ctl_error ? 0 : (err)) + /* check whether the control should be ignored */ static inline int check_ignored_ctl(const struct usbmix_name_map *p) @@ -1088,7 +1092,7 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, continue; err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &val); if (err < 0) - return cval->mixer->ignore_ctl_error ? 0 : err; + return filter_error(cval, err); val = get_relative_value(cval, val); ucontrol->value.integer.value[cnt] = val; cnt++; @@ -1098,7 +1102,7 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, /* master channel */ err = snd_usb_get_cur_mix_value(cval, 0, 0, &val); if (err < 0) - return cval->mixer->ignore_ctl_error ? 0 : err; + return filter_error(cval, err); val = get_relative_value(cval, val); ucontrol->value.integer.value[0] = val; } @@ -1120,7 +1124,7 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, continue; err = snd_usb_get_cur_mix_value(cval, c + 1, cnt, &oval); if (err < 0) - return cval->mixer->ignore_ctl_error ? 0 : err; + return filter_error(cval, err); val = ucontrol->value.integer.value[cnt]; val = get_abs_value(cval, val); if (oval != val) { @@ -1133,7 +1137,7 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, /* master channel */ err = snd_usb_get_cur_mix_value(cval, 0, 0, &oval); if (err < 0) - return cval->mixer->ignore_ctl_error ? 0 : err; + return filter_error(cval, err); val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { @@ -1628,12 +1632,10 @@ static int mixer_ctl_procunit_get(struct snd_kcontrol *kcontrol, int err, val; err = get_cur_ctl_value(cval, cval->control << 8, &val); - if (err < 0 && cval->mixer->ignore_ctl_error) { + if (err < 0) { ucontrol->value.integer.value[0] = cval->min; - return 0; + return filter_error(cval, err); } - if (err < 0) - return err; val = get_relative_value(cval, val); ucontrol->value.integer.value[0] = val; return 0; @@ -1647,11 +1649,8 @@ static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, int val, oval, err; err = get_cur_ctl_value(cval, cval->control << 8, &oval); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) - return 0; - return err; - } + if (err < 0) + return filter_error(cval, err); val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { @@ -1923,11 +1922,8 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, err = get_cur_ctl_value(cval, cval->control << 8, &val); if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.enumerated.item[0] = 0; - return 0; - } - return err; + ucontrol->value.enumerated.item[0] = 0; + return filter_error(cval, err); } val = get_relative_value(cval, val); ucontrol->value.enumerated.item[0] = val; @@ -1942,11 +1938,8 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, int val, oval, err; err = get_cur_ctl_value(cval, cval->control << 8, &oval); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) - return 0; - return err; - } + if (err < 0) + return filter_error(cval, err); val = ucontrol->value.enumerated.item[0]; val = get_abs_value(cval, val); if (val != oval) { -- cgit v1.1 From eaa8e5ef18fa9e09286482a4ded3a3cad36e44b2 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 21 Nov 2014 15:49:11 +0800 Subject: ALSA: hda/realtek - Supported HP mute Led for ALC286 New HP machine supported output mute led and input mute led. ALC286: GPIO1 to control output mute led. GPIO5 to control input mute led. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c29a55..3fcb7d9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3299,6 +3299,27 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc286_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct hda_verb gpio_init[] = { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x22 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x22 }, + {} + }; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook; + spec->gen.cap_sync_hook = alc_fixup_gpio_mic_mute_hook; + spec->gpio_led = 0; + spec->mute_led_polarity = 0; + spec->gpio_mute_led_mask = 0x02; + spec->gpio_mic_led_mask = 0x20; + snd_hda_add_verbs(codec, gpio_init); + } +} + /* turn on/off mic-mute LED per capture hook */ static void alc269_fixup_hp_cap_mic_mute_hook(struct hda_codec *codec, struct snd_kcontrol *kcontrol, @@ -4238,6 +4259,7 @@ enum { ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, + ALC286_FIXUP_HP_GPIO_LED, }; static const struct hda_fixup alc269_fixups[] = { @@ -4705,6 +4727,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio4, }, + [ALC286_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc286_fixup_hp_gpio_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4745,6 +4771,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- cgit v1.1 From 3360b84b8ed1f08bfb39743465b858a04492fcc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 11:47:04 +0100 Subject: ALSA: usb-audio: Allow quirks to handle own resume and proc dump So far, we blindly assumed that the all usb-audio mixer elements follow the standard and apply the standard resume method for the registered elements in the id_elems[] list. However, some quirks really need the own resume and it's incomplete for now. This patch enhances the resume handling in two folds: - split some fields in struct usb_mixer_elem_info into a smaller header struct (usb_mixer_elem_list) for keeping the minimal information in the linked-list; the usb_mixer_elem_info embeds this header struct instead - add resume and dump callbacks to usb_mixer_elem_list struct to allow quirks providing the own methods For the standard mixer elements, these new callbacks are set to the standard ones as default, thus there is no functional change by this patch yet. The dump and resume callbacks are typedef'ed for ease of later patches using arrays of such function pointers. Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 156 +++++++++++++++++++++++++-------------------- sound/usb/mixer.h | 26 ++++++-- sound/usb/mixer_quirks.c | 12 +--- sound/usb/mixer_scarlett.c | 8 +-- 4 files changed, 114 insertions(+), 88 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a4fbfff..41650d5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -138,7 +138,7 @@ check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) /* ignore the error value if ignore_ctl_error flag is set */ #define filter_error(cval, err) \ - ((cval)->mixer->ignore_ctl_error ? 0 : (err)) + ((cval)->head.mixer->ignore_ctl_error ? 0 : (err)) /* check whether the control should be ignored */ static inline int @@ -290,13 +290,13 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { - struct snd_usb_audio *chip = cval->mixer->chip; + struct snd_usb_audio *chip = cval->head.mixer->chip; unsigned char buf[2]; int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; int timeout = 10; int idx = 0, err; - err = snd_usb_autoresume(cval->mixer->chip); + err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -304,7 +304,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, while (timeout-- > 0) { if (chip->shutdown) break; - idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); + idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, val_len) >= val_len) { @@ -320,14 +320,14 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, out: up_read(&chip->shutdown_rwsem); - snd_usb_autosuspend(cval->mixer->chip); + snd_usb_autosuspend(chip); return err; } static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { - struct snd_usb_audio *chip = cval->mixer->chip; + struct snd_usb_audio *chip = cval->head.mixer->chip; unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; @@ -351,7 +351,7 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, if (chip->shutdown) { ret = -ENODEV; } else { - idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); + idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, idx, buf, size); @@ -396,7 +396,7 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, { validx += cval->idx_off; - return (cval->mixer->protocol == UAC_VERSION_1) ? + return (cval->head.mixer->protocol == UAC_VERSION_1) ? get_ctl_value_v1(cval, request, validx, value_ret) : get_ctl_value_v2(cval, request, validx, value_ret); } @@ -427,8 +427,8 @@ int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, } err = get_cur_mix_raw(cval, channel, value); if (err < 0) { - if (!cval->mixer->ignore_ctl_error) - usb_audio_dbg(cval->mixer->chip, + if (!cval->head.mixer->ignore_ctl_error) + usb_audio_dbg(cval->head.mixer->chip, "cannot get current value for control %d ch %d: err = %d\n", cval->control, channel, err); return err; @@ -445,13 +445,13 @@ int snd_usb_get_cur_mix_value(struct usb_mixer_elem_info *cval, int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { - struct snd_usb_audio *chip = cval->mixer->chip; + struct snd_usb_audio *chip = cval->head.mixer->chip; unsigned char buf[2]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; - if (cval->mixer->protocol == UAC_VERSION_1) { + if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ /* audio class v2 controls are always 2 bytes in size */ @@ -476,7 +476,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, while (timeout-- > 0) { if (chip->shutdown) break; - idx = snd_usb_ctrl_intf(chip) | (cval->id << 8); + idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); if (snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, @@ -510,7 +510,7 @@ int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, cval->ch_readonly & (1 << (channel - 1)); if (read_only) { - usb_audio_dbg(cval->mixer->chip, + usb_audio_dbg(cval->head.mixer->chip, "%s(): channel %d of control %d is read_only\n", __func__, channel, cval->control); return 0; @@ -569,10 +569,10 @@ static int check_matrix_bitmap(unsigned char *bmap, * if failed, give up and free the control instance. */ -int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, +int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, struct snd_kcontrol *kctl) { - struct usb_mixer_elem_info *cval = kctl->private_data; + struct usb_mixer_interface *mixer = list->mixer; int err; while (snd_ctl_find_id(mixer->chip->card, &kctl->id)) @@ -582,9 +582,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, err); return err; } - cval->elem_id = &kctl->id; - cval->next_id_elem = mixer->id_elems[cval->id]; - mixer->id_elems[cval->id] = cval; + list->kctl = kctl; + list->next_id_elem = mixer->id_elems[list->id]; + mixer->id_elems[list->id] = list; return 0; } @@ -833,7 +833,7 @@ void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) static void volume_control_quirks(struct usb_mixer_elem_info *cval, struct snd_kcontrol *kctl) { - struct snd_usb_audio *chip = cval->mixer->chip; + struct snd_usb_audio *chip = cval->head.mixer->chip; switch (chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ @@ -958,10 +958,10 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, } if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { - usb_audio_err(cval->mixer->chip, + usb_audio_err(cval->head.mixer->chip, "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->id, snd_usb_ctrl_intf(cval->mixer->chip), - cval->control, cval->id); + cval->head.id, snd_usb_ctrl_intf(cval->head.mixer->chip), + cval->control, cval->head.id); return -EINVAL; } if (get_ctl_value(cval, UAC_GET_RES, @@ -1065,7 +1065,7 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, kcontrol->vd[0].access &= ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); - snd_ctl_notify(cval->mixer->chip->card, + snd_ctl_notify(cval->head.mixer->chip->card, SNDRV_CTL_EVENT_MASK_INFO, &kcontrol->id); } @@ -1235,8 +1235,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return; - cval->mixer = state->mixer; - cval->id = unitid; + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; cval->val_type = audio_feature_info[control-1].type; @@ -1347,14 +1346,14 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, range); usb_audio_warn(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d", - cval->id, kctl->id.name, cval->channels, + cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); } usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", - cval->id, kctl->id.name, cval->channels, + cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); - snd_usb_mixer_add_control(state->mixer, kctl); + snd_usb_mixer_add_control(&cval->head, kctl); } /* @@ -1528,8 +1527,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, if (!cval) return; - cval->mixer = state->mixer; - cval->id = unitid; + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = in_ch + 1; /* based on 1 */ cval->val_type = USB_MIXER_S16; for (i = 0; i < num_outs; i++) { @@ -1561,8 +1559,8 @@ static void build_mixer_unit_ctl(struct mixer_build *state, append_ctl_name(kctl, " Volume"); usb_audio_dbg(state->chip, "[%d] MU [%s] ch = %d, val = %d/%d\n", - cval->id, kctl->id.name, cval->channels, cval->min, cval->max); - snd_usb_mixer_add_control(state->mixer, kctl); + cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max); + snd_usb_mixer_add_control(&cval->head, kctl); } /* @@ -1812,8 +1810,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return -ENOMEM; - cval->mixer = state->mixer; - cval->id = unitid; + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = valinfo->control; cval->val_type = valinfo->val_type; cval->channels = 1; @@ -1866,10 +1863,10 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, usb_audio_dbg(state->chip, "[%d] PU [%s] ch = %d, val = %d/%d\n", - cval->id, kctl->id.name, cval->channels, + cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max); - err = snd_usb_mixer_add_control(state->mixer, kctl); + err = snd_usb_mixer_add_control(&cval->head, kctl); if (err < 0) return err; } @@ -2016,8 +2013,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (!cval) return -ENOMEM; - cval->mixer = state->mixer; - cval->id = unitid; + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->val_type = USB_MIXER_U8; cval->channels = 1; cval->min = 1; @@ -2088,11 +2084,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, } usb_audio_dbg(state->chip, "[%d] SU [%s] items = %d\n", - cval->id, kctl->id.name, desc->bNrInPins); - if ((err = snd_usb_mixer_add_control(state->mixer, kctl)) < 0) - return err; - - return 0; + cval->head.id, kctl->id.name, desc->bNrInPins); + return snd_usb_mixer_add_control(&cval->head, kctl); } /* @@ -2237,25 +2230,21 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { - struct usb_mixer_elem_info *info; + struct usb_mixer_elem_list *list; - for (info = mixer->id_elems[unitid]; info; info = info->next_id_elem) + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - info->elem_id); + &list->kctl->id); } static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, - int unitid, - struct usb_mixer_elem_info *cval) + struct usb_mixer_elem_list *list) { + struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; - snd_iprintf(buffer, " Unit: %i\n", unitid); - if (cval->elem_id) - snd_iprintf(buffer, " Control: name=\"%s\", index=%i\n", - cval->elem_id->name, cval->elem_id->index); snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " - "channels=%i, type=\"%s\"\n", cval->id, + "channels=%i, type=\"%s\"\n", cval->head.id, cval->control, cval->cmask, cval->channels, val_types[cval->val_type]); snd_iprintf(buffer, " Volume: min=%i, max=%i, dBmin=%i, dBmax=%i\n", @@ -2267,7 +2256,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, { struct snd_usb_audio *chip = entry->private_data; struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *cval; + struct usb_mixer_elem_list *list; int unitid; list_for_each_entry(mixer, &chip->mixer_list, list) { @@ -2277,9 +2266,17 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (cval = mixer->id_elems[unitid]; cval; - cval = cval->next_id_elem) - snd_usb_mixer_dump_cval(buffer, unitid, cval); + for (list = mixer->id_elems[unitid]; list; + list = list->next_id_elem) { + snd_iprintf(buffer, " Unit: %i\n", list->id); + if (list->kctl) + snd_iprintf(buffer, + " Control: name=\"%s\", index=%i\n", + list->kctl->id.name, + list->kctl->id.index); + if (list->dump) + list->dump(buffer, list); + } } } } @@ -2287,7 +2284,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, int attribute, int value, int index) { - struct usb_mixer_elem_info *info; + struct usb_mixer_elem_list *list; __u8 unitid = (index >> 8) & 0xff; __u8 control = (value >> 8) & 0xff; __u8 channel = value & 0xff; @@ -2299,7 +2296,13 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (info = mixer->id_elems[unitid]; info; info = info->next_id_elem) { + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + struct usb_mixer_elem_info *info; + + if (!list->kctl) + continue; + + info = (struct usb_mixer_elem_info *)list; if (info->control != control) continue; @@ -2312,7 +2315,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - info->elem_id); + &info->head.kctl->id); break; case UAC2_CS_RANGE: @@ -2510,8 +2513,9 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) return 0; } -static int restore_mixer_value(struct usb_mixer_elem_info *cval) +static int restore_mixer_value(struct usb_mixer_elem_list *list) { + struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; int c, err, idx; if (cval->cmask) { @@ -2541,19 +2545,19 @@ static int restore_mixer_value(struct usb_mixer_elem_info *cval) int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) { - struct usb_mixer_elem_info *cval; + struct usb_mixer_elem_list *list; int id, err; - /* FIXME: any mixer quirks? */ - if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (cval = mixer->id_elems[id]; cval; - cval = cval->next_id_elem) { - err = restore_mixer_value(cval); - if (err < 0) - return err; + for (list = mixer->id_elems[id]; list; + list = list->next_id_elem) { + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } } } } @@ -2561,3 +2565,15 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) return snd_usb_mixer_activate(mixer); } #endif + +void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, + struct usb_mixer_interface *mixer, + int unitid) +{ + list->mixer = mixer; + list->id = unitid; + list->dump = snd_usb_mixer_dump_cval; +#ifdef CONFIG_PM + list->resume = restore_mixer_value; +#endif +} diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 2478a84..0fe87b7 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -1,6 +1,8 @@ #ifndef __USBMIXER_H #define __USBMIXER_H +#include + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -8,7 +10,7 @@ struct usb_mixer_interface { unsigned int ignore_ctl_error; struct urb *urb; /* array[MAX_ID_ELEMS], indexed by unit id */ - struct usb_mixer_elem_info **id_elems; + struct usb_mixer_elem_list **id_elems; /* the usb audio specification version this interface complies to */ int protocol; @@ -36,11 +38,21 @@ enum { USB_MIXER_U16, }; -struct usb_mixer_elem_info { +typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, + struct usb_mixer_elem_list *list); +typedef int (*usb_mixer_elem_resume_func_t)(struct usb_mixer_elem_list *elem); + +struct usb_mixer_elem_list { struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ - struct snd_ctl_elem_id *elem_id; + struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */ + struct snd_kcontrol *kctl; unsigned int id; + usb_mixer_elem_dump_func_t dump; + usb_mixer_elem_resume_func_t resume; +}; + +struct usb_mixer_elem_info { + struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ unsigned int cmask; /* channel mask bitmap: 0 = master */ unsigned int idx_off; /* Control index offset */ @@ -65,9 +77,13 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer, +int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, struct snd_kcontrol *kctl); +void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, + struct usb_mixer_interface *mixer, + int unitid); + int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 8b55c06..88a408c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -69,7 +69,6 @@ static int snd_create_std_mono_ctl_offset(struct usb_mixer_interface *mixer, const char *name, snd_kcontrol_tlv_rw_t *tlv_callback) { - int err; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; @@ -77,8 +76,7 @@ static int snd_create_std_mono_ctl_offset(struct usb_mixer_interface *mixer, if (!cval) return -ENOMEM; - cval->id = unitid; - cval->mixer = mixer; + snd_usb_mixer_elem_init_std(&cval->head, mixer, unitid); cval->val_type = val_type; cval->channels = 1; cval->control = control; @@ -112,11 +110,7 @@ static int snd_create_std_mono_ctl_offset(struct usb_mixer_interface *mixer, SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; } /* Add control to mixer */ - err = snd_usb_mixer_add_control(mixer, kctl); - if (err < 0) - return err; - - return 0; + return snd_usb_mixer_add_control(&cval->head, kctl); } static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, @@ -1206,7 +1200,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer->id_elems[unitid]; + cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index a0a8745..92dba35 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -436,10 +436,10 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct snd_usb_audio *chip = elem->mixer->chip; + struct snd_usb_audio *chip = elem->head.mixer->chip; unsigned char buf[2 * MAX_CHANNELS] = {0, }; int wValue = (elem->control << 8) | elem->idx_off; - int idx = snd_usb_ctrl_intf(chip) | (elem->id << 8); + int idx = snd_usb_ctrl_intf(chip) | (elem->head.id << 8); int err; err = snd_usb_ctl_msg(chip->dev, @@ -528,10 +528,10 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, if (!elem) return -ENOMEM; - elem->mixer = mixer; + elem->head.mixer = mixer; elem->control = offset; elem->idx_off = num; - elem->id = index; + elem->head.id = index; elem->val_type = val_type; elem->channels = channels; -- cgit v1.1 From 9cf3689bfe0784b6f6afb301bece95d3fc3eeb64 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 12:58:51 +0100 Subject: ALSA: usb-audio: Add audigy2nx resume support Rewrite the code to handle LEDs on audigy2nx and co for supporting the proper resume. A new internal helper function add_single_ctl_with_resume() is introduced to manage the usb_mixer_elem_list more easily. Also while we're at it, move audigy2nx_leds[] in usb_mixer_interface struct into the private_value of each kctl, too. Signed-off-by: Takashi Iwai --- sound/usb/mixer.h | 1 - sound/usb/mixer_quirks.c | 148 +++++++++++++++++++++++++++++------------------ 2 files changed, 92 insertions(+), 57 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 0fe87b7..0b3740e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -23,7 +23,6 @@ struct usb_mixer_interface { struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - u8 audigy2nx_leds[3]; u8 xonar_u1_status; }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 88a408c..41cacf8 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -144,6 +144,32 @@ static int snd_create_std_mono_table(struct usb_mixer_interface *mixer, return 0; } +static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, + int id, + usb_mixer_elem_resume_func_t resume, + const struct snd_kcontrol_new *knew, + struct usb_mixer_elem_list **listp) +{ + struct usb_mixer_elem_list *list; + struct snd_kcontrol *kctl; + + list = kzalloc(sizeof(*list), GFP_KERNEL); + if (!list) + return -ENOMEM; + if (listp) + *listp = list; + list->mixer = mixer; + list->id = id; + list->resume = resume; + kctl = snd_ctl_new1(knew, list); + if (!kctl) { + kfree(list); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + return snd_usb_mixer_add_control(list, kctl); +} + /* * Sound Blaster remote control configuration * @@ -271,84 +297,90 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) static int snd_audigy2nx_led_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - int index = kcontrol->private_value; - - ucontrol->value.integer.value[0] = mixer->audigy2nx_leds[index]; + ucontrol->value.integer.value[0] = kcontrol->private_value >> 8; return 0; } -static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_audigy2nx_led_update(struct usb_mixer_interface *mixer, + int value, int index) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - int index = kcontrol->private_value; - int value = ucontrol->value.integer.value[0]; - int err, changed; + struct snd_usb_audio *chip = mixer->chip; + int err; - if (value > 1) - return -EINVAL; - changed = value != mixer->audigy2nx_leds[index]; - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) { + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { err = -ENODEV; goto out; } - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + if (chip->usb_id == USB_ID(0x041e, 0x3042)) + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, !value, 0, NULL, 0); /* USB X-Fi S51 Pro */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + if (chip->usb_id == USB_ID(0x041e, 0x30df)) + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, !value, 0, NULL, 0); else - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, value, index + 2, NULL, 0); out: - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; - mixer->audigy2nx_leds[index] = value; - return changed; + up_read(&chip->shutdown_rwsem); + return err; } -static struct snd_kcontrol_new snd_audigy2nx_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CMSS LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 0, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Power LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 1, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Dolby Digital LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 2, - }, +static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + int index = kcontrol->private_value & 0xff; + int value = ucontrol->value.integer.value[0]; + int old_value = kcontrol->private_value >> 8; + int err; + + if (value > 1) + return -EINVAL; + if (value == old_value) + return 0; + kcontrol->private_value = (value << 8) | index; + err = snd_audigy2nx_led_update(mixer, value, index); + return err < 0 ? err : 1; +} + +static int snd_audigy2nx_led_resume(struct usb_mixer_elem_list *list) +{ + int priv_value = list->kctl->private_value; + + return snd_audigy2nx_led_update(list->mixer, priv_value >> 8, + priv_value & 0xff); +} + +/* name and private_value are set dynamically */ +static struct snd_kcontrol_new snd_audigy2nx_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_audigy2nx_led_info, + .get = snd_audigy2nx_led_get, + .put = snd_audigy2nx_led_put, +}; + +static const char * const snd_audigy2nx_led_names[] = { + "CMSS LED Switch", + "Power LED Switch", + "Dolby Digital LED Switch", }; static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) { int i, err; - for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_led_names); ++i) { + struct snd_kcontrol_new knew; + /* USB X-Fi S51 doesn't have a CMSS LED */ if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) continue; @@ -361,12 +393,16 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) mixer->chip->usb_id == USB_ID(0x041e, 0x30df) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); + + knew = snd_audigy2nx_control; + knew.name = snd_audigy2nx_led_names[i]; + knew.private_value = (1 << 8) | i; /* LED on as default */ + err = add_single_ctl_with_resume(mixer, 0, + snd_audigy2nx_led_resume, + &knew, NULL); if (err < 0) return err; } - mixer->audigy2nx_leds[1] = 1; /* Power LED is on by default */ return 0; } -- cgit v1.1 From 5f503ee9e270f8251ba9024bafa8d698050066cb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 16:11:37 +0100 Subject: ALSA: usb-audio: Add Emu0204 channel switch resume support Similar as the previous fix, this adds the proper resume support to Emu0202 "Front Jack Channels" enum mixer element. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 84 ++++++++++++++++++++++++++---------------------- 1 file changed, 46 insertions(+), 38 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 41cacf8..f2b1c0d 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -473,63 +473,71 @@ static int snd_emu0204_ch_switch_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_emu0204_ch_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_emu0204_ch_switch_update(struct usb_mixer_interface *mixer, + int value) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - unsigned int value = ucontrol->value.enumerated.item[0]; - int err, changed; + struct snd_usb_audio *chip = mixer->chip; + int err; unsigned char buf[2]; - if (value > 1) - return -EINVAL; - - buf[0] = 0x01; - buf[1] = value ? 0x02 : 0x01; - - changed = value != kcontrol->private_value; - down_read(&mixer->chip->shutdown_rwsem); + down_read(&chip->shutdown_rwsem); if (mixer->chip->shutdown) { err = -ENODEV; goto out; } - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), UAC_SET_CUR, + + buf[0] = 0x01; + buf[1] = value ? 0x02 : 0x01; + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, 0x0400, 0x0e00, buf, 2); out: - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; + up_read(&chip->shutdown_rwsem); + return err; +} + +static int snd_emu0204_ch_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + unsigned int value = ucontrol->value.enumerated.item[0]; + int err; + + if (value > 1) + return -EINVAL; + + if (value == kcontrol->private_value) + return 0; + kcontrol->private_value = value; - return changed; + err = snd_emu0204_ch_switch_update(mixer, value); + return err < 0 ? err : 1; } +static int snd_emu0204_ch_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_emu0204_ch_switch_update(list->mixer, + list->kctl->private_value); +} -static struct snd_kcontrol_new snd_emu0204_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Front Jack Channels", - .info = snd_emu0204_ch_switch_info, - .get = snd_emu0204_ch_switch_get, - .put = snd_emu0204_ch_switch_put, - .private_value = 0, - }, +static struct snd_kcontrol_new snd_emu0204_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Front Jack Channels", + .info = snd_emu0204_ch_switch_info, + .get = snd_emu0204_ch_switch_get, + .put = snd_emu0204_ch_switch_put, + .private_value = 0, }; static int snd_emu0204_controls_create(struct usb_mixer_interface *mixer) { - int i, err; - - for (i = 0; i < ARRAY_SIZE(snd_emu0204_controls); ++i) { - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_emu0204_controls[i], mixer)); - if (err < 0) - return err; - } - - return 0; + return add_single_ctl_with_resume(mixer, 0, + snd_emu0204_ch_switch_resume, + &snd_emu0204_control, NULL); } + /* ASUS Xonar U1 / U3 controls */ static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, -- cgit v1.1 From 2bfb14c3b8fbc787ff4478f9d77ecee78cb922fe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 16:18:15 +0100 Subject: ALSA: usb-audio: Add Xonar U1 resume support This time it's about Xonar U1: add the proper resume support for "Digital Playback Switch" element. Also, the status is moved into kcontrol private_value from usb_mixer_interface struct field. One more cut. Signed-off-by: Takashi Iwai --- sound/usb/mixer.h | 2 -- sound/usb/mixer_quirks.c | 66 ++++++++++++++++++++++++++++-------------------- 2 files changed, 38 insertions(+), 30 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 0b3740e..d3268f0 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,8 +22,6 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - - u8 xonar_u1_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f2b1c0d..4afcf09 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -543,38 +543,52 @@ static int snd_emu0204_controls_create(struct usb_mixer_interface *mixer) static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(kcontrol->private_value & 0x02); return 0; } +static int snd_xonar_u1_switch_update(struct usb_mixer_interface *mixer, + unsigned char status) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) + err = -ENODEV; + else + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x08, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + 50, 0, &status, 1); + up_read(&chip->shutdown_rwsem); + return err; +} + static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; - int err, changed; + int err; - old_status = mixer->xonar_u1_status; + old_status = kcontrol->private_value; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else new_status = old_status & ~0x02; - changed = new_status != old_status; - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) - err = -ENODEV; - else - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 50, 0, &new_status, 1); - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; - mixer->xonar_u1_status = new_status; - return changed; + if (new_status == old_status) + return 0; + + kcontrol->private_value = new_status; + err = snd_xonar_u1_switch_update(list->mixer, new_status); + return err < 0 ? err : 1; +} + +static int snd_xonar_u1_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_xonar_u1_switch_update(list->mixer, + list->kctl->private_value); } static struct snd_kcontrol_new snd_xonar_u1_output_switch = { @@ -583,18 +597,14 @@ static struct snd_kcontrol_new snd_xonar_u1_output_switch = { .info = snd_ctl_boolean_mono_info, .get = snd_xonar_u1_switch_get, .put = snd_xonar_u1_switch_put, + .private_value = 0x05, }; static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) { - int err; - - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); - if (err < 0) - return err; - mixer->xonar_u1_status = 0x05; - return 0; + return add_single_ctl_with_resume(mixer, 0, + snd_xonar_u1_switch_resume, + &snd_xonar_u1_output_switch, NULL); } /* Digidesign Mbox 1 clock source switch (internal/spdif) */ -- cgit v1.1 From 25a9a4f91b909822fa07cbc9939c99a8c67d8960 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 16:31:35 +0100 Subject: ALSA: usb-audio: Add Digidesign Mbox 1 resume support Again another quirk fix, just convert to usb_mixer_elem_list with the resume callback for Mbox 1 stuff. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 52 +++++++++++++++++++++++++++--------------------- 1 file changed, 29 insertions(+), 23 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4afcf09..f7ad207 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -616,29 +616,12 @@ static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, return 0; } -static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) { - struct snd_usb_audio *chip; - struct usb_mixer_interface *mixer; + struct snd_usb_audio *chip = mixer->chip; int err; - bool cur_val, new_val; unsigned char buff[3]; - cur_val = kctl->private_value; - new_val = ucontrol->value.enumerated.item[0]; - - mixer = snd_kcontrol_chip(kctl); - if (snd_BUG_ON(!mixer)) - return -EINVAL; - - chip = mixer->chip; - if (snd_BUG_ON(!chip)) - return -EINVAL; - - if (cur_val == new_val) - return 0; - down_read(&chip->shutdown_rwsem); if (chip->shutdown) { err = -ENODEV; @@ -668,7 +651,7 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, * while S/PDIF sync is enabled and confusing * this configuration. */ - if (new_val == 0) { + if (val == 0) { buff[0] = 0x80; buff[1] = 0xbb; buff[2] = 0x00; @@ -697,10 +680,27 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, USB_RECIP_ENDPOINT, 0x100, 0x2, buff, 3); if (err < 0) goto err; - kctl->private_value = new_val; err: up_read(&chip->shutdown_rwsem); + return err; +} + +static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + int err; + bool cur_val, new_val; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; + + kctl->private_value = new_val; + err = snd_mbox1_switch_update(mixer, new_val); return err < 0 ? err : 1; } @@ -715,6 +715,11 @@ static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } +static int snd_mbox1_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_mbox1_switch_update(list->mixer, list->kctl->private_value); +} + static struct snd_kcontrol_new snd_mbox1_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Source", @@ -728,8 +733,9 @@ static struct snd_kcontrol_new snd_mbox1_switch = { static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) { - return snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_mbox1_switch, mixer)); + return add_single_ctl_with_resume(mixer, 0, + snd_mbox1_switch_resume, + &snd_mbox1_switch, NULL); } /* Native Instruments device quirks */ -- cgit v1.1 From da6d276957ea56b9514aa5c8d885edf22f0b3e65 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 16:59:47 +0100 Subject: ALSA: usb-audio: Add resume support for Native Instruments controls The changes at this time are a bit more wider than previous ones. Firstly, the NI controls didn't cache the values, so I had to implement the caching. It's stored in bit 24 of private_value. In addition to that, the initial values have to be read from registers. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 99 +++++++++++++++++++++++++----------------------- 1 file changed, 52 insertions(+), 47 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f7ad207..e11b4f3 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -742,64 +742,68 @@ static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) #define _MAKE_NI_CONTROL(bRequest,wIndex) ((bRequest) << 16 | (wIndex)) -static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_ni_control_init_val(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); struct usb_device *dev = mixer->chip->dev; - u8 bRequest = (kcontrol->private_value >> 16) & 0xff; - u16 wIndex = kcontrol->private_value & 0xffff; - u8 tmp; - int ret; - - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) - ret = -ENODEV; - else - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, - USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, wIndex, - &tmp, sizeof(tmp)); - up_read(&mixer->chip->shutdown_rwsem); + unsigned int pval = kctl->private_value; + u8 value; + int err; - if (ret < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + (pval >> 16) & 0xff, + USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, + 0, pval & 0xffff, &value, 1); + if (err < 0) { dev_err(&dev->dev, - "unable to issue vendor read request (ret = %d)", ret); - return ret; + "unable to issue vendor read request (ret = %d)", err); + return err; } - ucontrol->value.integer.value[0] = tmp; - + kctl->private_value |= (value << 24); return 0; } -static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, +static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - struct usb_device *dev = mixer->chip->dev; - u8 bRequest = (kcontrol->private_value >> 16) & 0xff; - u16 wIndex = kcontrol->private_value & 0xffff; - u16 wValue = ucontrol->value.integer.value[0]; - int ret; + ucontrol->value.integer.value[0] = kcontrol->private_value >> 24; + return 0; +} - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) - ret = -ENODEV; +static int snd_ni_update_cur_val(struct usb_mixer_elem_list *list) +{ + struct snd_usb_audio *chip = list->mixer->chip; + unsigned int pval = list->kctl->private_value; + int err; + + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) + err = -ENODEV; else - ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, - USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - wValue, wIndex, - NULL, 0, 1000); - up_read(&mixer->chip->shutdown_rwsem); + err = usb_control_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), + (pval >> 16) & 0xff, + USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, + pval >> 24, pval & 0xffff, NULL, 0, 1000); + up_read(&chip->shutdown_rwsem); + return err; +} - if (ret < 0) { - dev_err(&dev->dev, - "unable to issue vendor write request (ret = %d)", ret); - return ret; - } +static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + u8 oldval = (kcontrol->private_value >> 24) & 0xff; + u8 newval = ucontrol->value.integer.value[0]; + int err; - return 0; + if (oldval == newval) + return 0; + + kcontrol->private_value &= ~(0xff << 24); + kcontrol->private_value |= newval; + err = snd_ni_update_cur_val(list); + return err < 0 ? err : 1; } static struct snd_kcontrol_new snd_nativeinstruments_ta6_mixers[] = { @@ -870,16 +874,17 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, }; for (i = 0; i < count; i++) { - struct snd_kcontrol *c; + struct usb_mixer_elem_list *list; template.name = kc[i].name; template.private_value = kc[i].private_value; - c = snd_ctl_new1(&template, mixer); - err = snd_ctl_add(mixer->chip->card, c); - + err = add_single_ctl_with_resume(mixer, 0, + snd_ni_update_cur_val, + &template, &list); if (err < 0) break; + snd_ni_control_init_val(mixer, list->kctl); } return err; -- cgit v1.1 From 0b4e9cfcef055a1be9bee5a47262e9cbcf17e8cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 17:37:40 +0100 Subject: ALSA: usb-audio: Add resume support for FTU controls A few FTU mixer controls have the own value handling, so they have to be rewritten to follow the support for resume callbacks. This ended up in a fair amount of refactoring. Its own struct is now removed, instead the values are embedded in kctl private_value totally. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 194 +++++++++++++---------------------------------- 1 file changed, 54 insertions(+), 140 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e11b4f3..fcb7ae5 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -892,14 +892,6 @@ static int snd_nativeinstruments_create_mixer(struct usb_mixer_interface *mixer, /* M-Audio FastTrack Ultra quirks */ /* FTU Effect switch (also used by C400/C600) */ -struct snd_ftu_eff_switch_priv_val { - struct usb_mixer_interface *mixer; - int cached_value; - int is_cached; - int bUnitID; - int validx; -}; - static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -911,138 +903,77 @@ static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol, return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } -static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_ftu_eff_switch_init(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) { - struct snd_usb_audio *chip; - struct usb_mixer_interface *mixer; - struct snd_ftu_eff_switch_priv_val *pval; + struct usb_device *dev = mixer->chip->dev; + unsigned int pval = kctl->private_value; int err; unsigned char value[2]; - int id, validx; - - const int val_len = 2; value[0] = 0x00; value[1] = 0x00; - pval = (struct snd_ftu_eff_switch_priv_val *) - kctl->private_value; + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + pval & 0xff00, + snd_usb_ctrl_intf(mixer->chip) | ((pval & 0xff) << 8), + value, 2); + if (err < 0) + return err; - if (pval->is_cached) { - ucontrol->value.enumerated.item[0] = pval->cached_value; - return 0; - } + kctl->private_value |= value[0] << 24; + return 0; +} - mixer = (struct usb_mixer_interface *) pval->mixer; - if (snd_BUG_ON(!mixer)) - return -EINVAL; +static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kctl->private_value >> 24; + return 0; +} - chip = (struct snd_usb_audio *) mixer->chip; - if (snd_BUG_ON(!chip)) - return -EINVAL; +static int snd_ftu_eff_switch_update(struct usb_mixer_elem_list *list) +{ + struct snd_usb_audio *chip = list->mixer->chip; + unsigned int pval = list->kctl->private_value; + unsigned char value[2]; + int err; - id = pval->bUnitID; - validx = pval->validx; + value[0] = pval >> 24; + value[1] = 0; - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) err = -ENODEV; else err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), - value, val_len); - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; - - ucontrol->value.enumerated.item[0] = value[0]; - pval->cached_value = value[0]; - pval->is_cached = 1; - - return 0; + usb_sndctrlpipe(chip->dev, 0), + UAC_SET_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, + pval & 0xff00, + snd_usb_ctrl_intf(chip) | ((pval & 0xff) << 8), + value, 2); + up_read(&chip->shutdown_rwsem); + return err; } static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct snd_usb_audio *chip; - struct snd_ftu_eff_switch_priv_val *pval; - - struct usb_mixer_interface *mixer; - int changed, cur_val, err, new_val; - unsigned char value[2]; - int id, validx; - - const int val_len = 2; - - changed = 0; + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + unsigned int pval = list->kctl->private_value; + int cur_val, err, new_val; - pval = (struct snd_ftu_eff_switch_priv_val *) - kctl->private_value; - cur_val = pval->cached_value; + cur_val = pval >> 24; new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; - mixer = (struct usb_mixer_interface *) pval->mixer; - if (snd_BUG_ON(!mixer)) - return -EINVAL; - - chip = (struct snd_usb_audio *) mixer->chip; - if (snd_BUG_ON(!chip)) - return -EINVAL; - - id = pval->bUnitID; - validx = pval->validx; - - if (!pval->is_cached) { - /* Read current value */ - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) - err = -ENODEV; - else - err = snd_usb_ctl_msg(chip->dev, - usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, - validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), - value, val_len); - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; - - cur_val = value[0]; - pval->cached_value = cur_val; - pval->is_cached = 1; - } - /* update value if needed */ - if (cur_val != new_val) { - value[0] = new_val; - value[1] = 0; - down_read(&mixer->chip->shutdown_rwsem); - if (mixer->chip->shutdown) - err = -ENODEV; - else - err = snd_usb_ctl_msg(chip->dev, - usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, - USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, - validx << 8, snd_usb_ctrl_intf(chip) | (id << 8), - value, val_len); - up_read(&mixer->chip->shutdown_rwsem); - if (err < 0) - return err; - - pval->cached_value = new_val; - pval->is_cached = 1; - changed = 1; - } - - return changed; -} - -static void kctl_private_value_free(struct snd_kcontrol *kctl) -{ - kfree((void *)kctl->private_value); + kctl->private_value &= ~(0xff << 24); + kctl->private_value |= new_val << 24; + err = snd_ftu_eff_switch_update(list); + return err < 0 ? err : 1; } static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, @@ -1057,33 +988,16 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, .get = snd_ftu_eff_switch_get, .put = snd_ftu_eff_switch_put }; - + struct usb_mixer_elem_list *list; int err; - struct snd_kcontrol *kctl; - struct snd_ftu_eff_switch_priv_val *pval; - - pval = kzalloc(sizeof(*pval), GFP_KERNEL); - if (!pval) - return -ENOMEM; - - pval->cached_value = 0; - pval->is_cached = 0; - pval->mixer = mixer; - pval->bUnitID = bUnitID; - pval->validx = validx; - template.private_value = (unsigned long) pval; - kctl = snd_ctl_new1(&template, mixer->chip); - if (!kctl) { - kfree(pval); - return -ENOMEM; - } - - kctl->private_free = kctl_private_value_free; - err = snd_ctl_add(mixer->chip->card, kctl); + err = add_single_ctl_with_resume(mixer, bUnitID, + snd_ftu_eff_switch_update, + &template, &list); if (err < 0) return err; - + list->kctl->private_value = (validx << 8) | bUnitID; + snd_ftu_eff_switch_init(mixer, list->kctl); return 0; } -- cgit v1.1 From 288673beae6cbd8198be94284adbaeb5cba7dbda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Nov 2014 18:06:17 +0100 Subject: ALSA: usb-audio: Add resume support for MicroII SPDIF ctls Like the previous fixes, the mixer accessors are converted to use usb_mixer_elem_list objects. In addition, the proper shutdown check are put in get and put callbacks. Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 135 ++++++++++++++++++++++++++++++++--------------- 1 file changed, 93 insertions(+), 42 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index fcb7ae5..dc9df00 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1509,7 +1509,8 @@ static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; int err; struct usb_interface *iface; struct usb_host_interface *alts; @@ -1517,17 +1518,23 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, unsigned char data[3]; int rate; + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { + err = -ENODEV; + goto end; + } + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; ucontrol->value.iec958.status[2] = 0x00; /* use known values for that card: interface#1 altsetting#1 */ - iface = usb_ifnum_to_if(mixer->chip->dev, 1); + iface = usb_ifnum_to_if(chip->dev, 1); alts = &iface->altsetting[1]; ep = get_endpoint(alts, 0)->bEndpointAddress; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_rcvctrlpipe(mixer->chip->dev, 0), + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, UAC_EP_CS_ATTR_SAMPLE_RATE << 8, @@ -1542,22 +1549,27 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; err = 0; -end: + end: + up_read(&chip->shutdown_rwsem); return err; } -static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_microii_spdif_default_update(struct usb_mixer_elem_list *list) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - int err; + struct snd_usb_audio *chip = list->mixer->chip; + unsigned int pval = list->kctl->private_value; u8 reg; - unsigned long priv_backup = kcontrol->private_value; + int err; + + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { + err = -ENODEV; + goto end; + } - reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | - (ucontrol->value.iec958.status[0] & 0x0f); - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), + reg = ((pval >> 4) & 0xf0) | (pval & 0x0f); + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, reg, @@ -1567,15 +1579,10 @@ static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, if (err < 0) goto end; - kcontrol->private_value &= 0xfffff0f0; - kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; - kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); - - reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? - 0xa0 : 0x20; - reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), + reg = (pval & IEC958_AES0_NONAUDIO) ? 0xa0 : 0x20; + reg |= (pval >> 12) & 0x0f; + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, reg, @@ -1585,16 +1592,36 @@ static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, if (err < 0) goto end; - kcontrol->private_value &= 0xffff0fff; - kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + end: + up_read(&chip->shutdown_rwsem); + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + unsigned int pval, pval_old; + int err; + + pval = pval_old = kcontrol->private_value; + pval &= 0xfffff0f0; + pval |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + pval |= (ucontrol->value.iec958.status[0] & 0x0f); + + pval &= 0xffff0fff; + pval |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; /* The frequency bits in AES3 cannot be set via register access. */ /* Silently ignore any bits from the request that cannot be set. */ - err = (priv_backup != kcontrol->private_value); -end: - return err; + if (pval == pval_old) + return 0; + + kcontrol->private_value = pval; + err = snd_microii_spdif_default_update(list); + return err < 0 ? err : 1; } static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, @@ -1616,15 +1643,20 @@ static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, return 0; } -static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_microii_spdif_switch_update(struct usb_mixer_elem_list *list) { - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + u8 reg = list->kctl->private_value; int err; - u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), + down_read(&chip->shutdown_rwsem); + if (chip->shutdown) { + err = -ENODEV; + goto end; + } + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, reg, @@ -1632,15 +1664,27 @@ static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, NULL, 0); - if (!err) { - err = (reg != (kcontrol->private_value & 0x0ff)); - if (err) - kcontrol->private_value = reg; - } - + end: + up_read(&chip->shutdown_rwsem); return err; } +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + u8 reg; + int err; + + reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + if (reg != list->kctl->private_value) + return 0; + + kcontrol->private_value = reg; + err = snd_microii_spdif_switch_update(list); + return err < 0 ? err : 1; +} + static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { { .iface = SNDRV_CTL_ELEM_IFACE_PCM, @@ -1670,10 +1714,17 @@ static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { static int snd_microii_controls_create(struct usb_mixer_interface *mixer) { int err, i; + static usb_mixer_elem_resume_func_t resume_funcs[] = { + snd_microii_spdif_default_update, + NULL, + snd_microii_spdif_switch_update + }; for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + err = add_single_ctl_with_resume(mixer, 0, + resume_funcs[i], + &snd_microii_mixer_spdif[i], + NULL); if (err < 0) return err; } -- cgit v1.1 From b61f90eac1ff9d1b30497e611aba4651d4066706 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 20 Nov 2014 17:20:46 +0100 Subject: ALSA: usb-audio: Add resume support for Scarlett mixers Scarlett driver uses almost compatible usb_mixer_elem_info struct, so we just need to add a couple of simple resume callbacks to handle them accordingly. Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett.c | 65 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 92dba35..9109652 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -285,6 +285,19 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, return changed; } +static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *elem = + container_of(list, struct usb_mixer_elem_info, head); + int i; + + for (i = 0; i < elem->channels; i++) + if (elem->cached & (1 << i)) + snd_usb_set_cur_mix_value(elem, i, i, + elem->cache_val[i]); + return 0; +} + static int scarlett_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { @@ -432,6 +445,16 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, return 0; } +static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *elem = + container_of(list, struct usb_mixer_elem_info, head); + + if (elem->cached) + snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); + return 0; +} + static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -514,6 +537,7 @@ static struct snd_kcontrol_new usb_scarlett_ctl_sync = { static int add_new_ctl(struct usb_mixer_interface *mixer, const struct snd_kcontrol_new *ncontrol, + usb_mixer_elem_resume_func_t resume, int index, int offset, int num, int val_type, int channels, const char *name, const struct scarlett_mixer_elem_enum_info *opt, @@ -529,6 +553,7 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, return -ENOMEM; elem->head.mixer = mixer; + elem->head.resume = resume; elem->control = offset; elem->idx_off = num; elem->head.id = index; @@ -548,7 +573,7 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); - err = snd_ctl_add(mixer->chip->card, kctl); + err = snd_usb_mixer_add_control(&elem->head, kctl); if (err < 0) return err; @@ -569,7 +594,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, /* Add mute switch */ snprintf(mx, sizeof(mx), "Master %d (%s) Playback Switch", index + 1, name); - err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, 0x0a, 0x01, + err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, + scarlett_ctl_resume, 0x0a, 0x01, 2*index+1, USB_MIXER_S16, 2, mx, NULL, &elem); if (err < 0) return err; @@ -577,7 +603,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, /* Add volume control and initialize to 0 */ snprintf(mx, sizeof(mx), "Master %d (%s) Playback Volume", index + 1, name); - err = add_new_ctl(mixer, &usb_scarlett_ctl_master, 0x0a, 0x02, + err = add_new_ctl(mixer, &usb_scarlett_ctl_master, + scarlett_ctl_resume, 0x0a, 0x02, 2*index+1, USB_MIXER_S16, 2, mx, NULL, &elem); if (err < 0) return err; @@ -585,7 +612,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, /* Add L channel source playback enumeration */ snprintf(mx, sizeof(mx), "Master %dL (%s) Source Playback Enum", index + 1, name); - err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x33, 0x00, + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, + scarlett_ctl_enum_resume, 0x33, 0x00, 2*index, USB_MIXER_S16, 1, mx, &info->opt_master, &elem); if (err < 0) @@ -594,7 +622,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, /* Add R channel source playback enumeration */ snprintf(mx, sizeof(mx), "Master %dR (%s) Source Playback Enum", index + 1, name); - err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x33, 0x00, + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, + scarlett_ctl_enum_resume, 0x33, 0x00, 2*index+1, USB_MIXER_S16, 1, mx, &info->opt_master, &elem); if (err < 0) @@ -824,13 +853,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *elem; /* create master switch and playback volume */ - err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, 0x0a, 0x01, 0, + err = add_new_ctl(mixer, &usb_scarlett_ctl_switch, + scarlett_ctl_resume, 0x0a, 0x01, 0, USB_MIXER_S16, 1, "Master Playback Switch", NULL, &elem); if (err < 0) return err; - err = add_new_ctl(mixer, &usb_scarlett_ctl_master, 0x0a, 0x02, 0, + err = add_new_ctl(mixer, &usb_scarlett_ctl_master, + scarlett_ctl_resume, 0x0a, 0x02, 0, USB_MIXER_S16, 1, "Master Playback Volume", NULL, &elem); if (err < 0) @@ -848,7 +879,8 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, break; case SCARLETT_SWITCH_IMPEDANCE: sprintf(mx, "Input %d Impedance Switch", ctl->num); - err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x01, + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, 0x09, ctl->num, USB_MIXER_S16, 1, mx, &opt_impedance, &elem); if (err < 0) @@ -856,7 +888,8 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, break; case SCARLETT_SWITCH_PAD: sprintf(mx, "Input %d Pad Switch", ctl->num); - err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x01, + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, 0x0b, ctl->num, USB_MIXER_S16, 1, mx, &opt_pad, &elem); if (err < 0) @@ -912,7 +945,8 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) for (i = 0; i < info->matrix_in; i++) { snprintf(mx, sizeof(mx), "Matrix %02d Input Playback Route", i+1); - err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x32, + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, + scarlett_ctl_enum_resume, 0x32, 0x06, i, USB_MIXER_S16, 1, mx, &info->opt_matrix, &elem); if (err < 0) @@ -921,7 +955,8 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) for (o = 0; o < info->matrix_out; o++) { sprintf(mx, "Matrix %02d Mix %c Playback Volume", i+1, o+'A'); - err = add_new_ctl(mixer, &usb_scarlett_ctl, 0x3c, 0x00, + err = add_new_ctl(mixer, &usb_scarlett_ctl, + scarlett_ctl_resume, 0x3c, 0x00, (i << 3) + (o & 0x07), USB_MIXER_S16, 1, mx, NULL, &elem); if (err < 0) @@ -933,7 +968,8 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) for (i = 0; i < info->input_len; i++) { snprintf(mx, sizeof(mx), "Input Source %02d Capture Route", i+1); - err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, 0x34, + err = add_new_ctl(mixer, &usb_scarlett_ctl_dynamic_enum, + scarlett_ctl_enum_resume, 0x34, 0x00, i, USB_MIXER_S16, 1, mx, &info->opt_master, &elem); if (err < 0) @@ -941,14 +977,15 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) } /* val_len == 1 needed here */ - err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, 0x28, 0x01, 0, + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x28, 0x01, 0, USB_MIXER_U8, 1, "Sample Clock Source", &opt_clock, &elem); if (err < 0) return err; /* val_len == 1 and UAC2_CS_MEM */ - err = add_new_ctl(mixer, &usb_scarlett_ctl_sync, 0x3c, 0x00, 2, + err = add_new_ctl(mixer, &usb_scarlett_ctl_sync, NULL, 0x3c, 0x00, 2, USB_MIXER_U8, 1, "Sample Clock Sync Status", &opt_sync, &elem); if (err < 0) -- cgit v1.1 From 1a28fc190c60e9bb04649384826f3224c8463efc Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Fri, 21 Nov 2014 19:06:15 +0800 Subject: ASoC: Intel: byt_rvp_platform_data can be static sound/soc/intel/sst/sst_acpi.c:124:26: sparse: symbol 'byt_rvp_platform_data' was not declared. Should it be static? Signed-off-by: Fengguang Wu Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index b261821..f94f007 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -121,7 +121,7 @@ static const struct sst_res_info byt_rvp_res_info = { .acpi_ipc_irq_index = 5, }; -struct sst_platform_info byt_rvp_platform_data = { +static struct sst_platform_info byt_rvp_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, .lib_info = &byt_lib_dnld_info, -- cgit v1.1 From b2de1d20a05d8691cb1889c859de2ab56938b82a Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Thu, 20 Nov 2014 15:33:17 +0530 Subject: ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk() In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register in i2s_set_sysclk. It's due to an incorrect mask used for the AND operation which is introduced in commit a5a56871f804edac93a53b5e871c0e9818fb9033 (ASoC: samsung: add support for exynos7 I2S controller) and also adds the missing break statement. Signed-off-by: Sylwester Nawrocki Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..0d76bc1 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; @@ -551,10 +551,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= 0 << i2s_regs->rclksrc_off; + mod &= ~(1 << i2s_regs->rclksrc_off); else mod |= 1 << i2s_regs->rclksrc_off; + break; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); return -EINVAL; -- cgit v1.1 From 335ca471eebf130d88cb94c1192568b6c75aa9b0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:28:15 +0100 Subject: ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. While we are at it also replace dev_get_drvdata() with snd_soc_codec_get_drvdata(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sirf-audio-codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 06ba492..07eea20 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -120,7 +120,8 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define ATLAS6_CODEC_ENABLE_BITS (1 << 29) #define ATLAS6_CODEC_RESET_BITS (1 << 28) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, @@ -142,7 +143,8 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define PRIMA2_CODEC_ENABLE_BITS (1 << 27) #define PRIMA2_CODEC_RESET_BITS (1 << 26) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, -- cgit v1.1 From 0b5155bbca8b5a8a1456ae462a47eeaedf8ce091 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:21:53 +0100 Subject: ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 2cd3e54..abf3832 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -875,7 +875,7 @@ static const struct snd_kcontrol_new max98088_right_ADC_mixer_controls[] = { static int max98088_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -905,7 +905,7 @@ static int max98088_mic_event(struct snd_soc_dapm_widget *w, static int max98088_line_pga(struct snd_soc_dapm_widget *w, int event, int line, u8 channel) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); u8 *state; -- cgit v1.1 From 24445f8c5eae926e402335bbe0292f09b1deb7a7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:21:54 +0100 Subject: ASoC: max98090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index a65861c..2ad381c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -806,7 +806,7 @@ static const struct snd_kcontrol_new max98091_snd_controls[] = { static int max98090_micinput_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); unsigned int val = snd_soc_read(codec, w->reg); -- cgit v1.1 From 0db5dc943e7649bbfbc2d2de8f5cb778b05ea5bd Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:21:55 +0100 Subject: ASoC: max98095: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 42103ca..d911d4c 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -864,7 +864,7 @@ static const struct snd_kcontrol_new max98095_right_ADC_mixer_controls[] = { static int max98095_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -894,7 +894,7 @@ static int max98095_mic_event(struct snd_soc_dapm_widget *w, static int max98095_line_pga(struct snd_soc_dapm_widget *w, int event, u8 channel) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); u8 *state; @@ -942,7 +942,7 @@ static int max98095_pga_in2_event(struct snd_soc_dapm_widget *w, static int max98095_lineout_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); switch (event) { case SND_SOC_DAPM_POST_PMU: -- cgit v1.1 From dee9cec42fc9cc4635ea2f45939e443210a638f8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 21 Nov 2014 18:53:51 +0100 Subject: ASoC: adau17x1: Mark DSP parameter memory as readable and precious To be able to read back data from the DSP parameter memory the register range needs to be marked as readable. At the same time we do not want them to e.g. appear in debugfs output so mark them as precious as well. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1761.c | 1 + sound/soc/codecs/adau1781.c | 1 + sound/soc/codecs/adau17x1.c | 14 ++++++++++++++ sound/soc/codecs/adau17x1.h | 1 + 4 files changed, 17 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 0ae1501..4c018c5 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -793,6 +793,7 @@ const struct regmap_config adau1761_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1761_reg_defaults), .readable_reg = adau1761_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1761_regmap_config); diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 4c8ddc3..926fc99 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -472,6 +472,7 @@ const struct regmap_config adau1781_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1781_reg_defaults), .readable_reg = adau1781_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1781_regmap_config); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 1cab34c..5000047 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -706,8 +706,22 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); +bool adau17x1_precious_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_precious_register); + bool adau17x1_readable_register(struct device *dev, unsigned int reg) { + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + switch (reg) { case ADAU17X1_CLOCK_CONTROL: case ADAU17X1_PLL_CONTROL: diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 6861aa3..e13583e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -56,6 +56,7 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +bool adau17x1_precious_register(struct device *dev, unsigned int reg); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -- cgit v1.1 From 1fc10044d76e86b71f724988c7cbd8205bb903a8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 21 Nov 2014 18:53:52 +0100 Subject: ASoC: sigmadsp: Fix endianness conversion Make sure to always convert the firmware data to local endianness before using it. Reported-by: kbuild test robot Fixes: a35daac77a03 ("ASoC: sigmadsp: Add support for fw v2") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 55af596..6abefd2 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -235,7 +235,7 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, ctrl->addr = le16_to_cpu(ctrl_chunk->addr); ctrl->num_bytes = num_bytes; - ctrl->samplerates = chunk->samplerates; + ctrl->samplerates = le32_to_cpu(chunk->samplerates); list_add_tail(&ctrl->head, &sigmadsp->ctrl_list); @@ -266,7 +266,7 @@ static int sigma_fw_load_data(struct sigmadsp *sigmadsp, data->addr = le16_to_cpu(data_chunk->addr); data->length = length; - data->samplerates = chunk->samplerates; + data->samplerates = le32_to_cpu(chunk->samplerates); memcpy(data->data, data_chunk->data, length); list_add_tail(&data->head, &sigmadsp->data_list); @@ -329,7 +329,7 @@ static int sigmadsp_fw_load_v2(struct sigmadsp *sigmadsp, if (length > fw->size - pos || length < sizeof(*chunk)) return -EINVAL; - switch (chunk->tag) { + switch (le32_to_cpu(chunk->tag)) { case SIGMA_FW_CHUNK_TYPE_DATA: ret = sigma_fw_load_data(sigmadsp, chunk, length); break; -- cgit v1.1 From e2280c9040d8bc5039617af35ccf7b8ac4abb428 Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Thu, 20 Nov 2014 19:07:48 +0800 Subject: ASoC: wm8960: Add device tree support Document the device tree binding for the WM8960 codec, and modify the driver to extract the platform data from device tree, if present. Signed-off-by: Zidan Wang Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/wm8960.txt | 31 ++++++++++++++++ sound/soc/codecs/wm8960.c | 41 ++++++++++++++++------ 2 files changed, 62 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt new file mode 100644 index 0000000..2deb8a3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,31 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + When wm8960 works on synchronize mode and DACLRC pin is used to supply + frame clock, it will no frame clock for captrue unless enable DAC to enable + DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue. + + - wlf,capless: This is a boolean property. If present, OUT3 pin will be + enabled and disabled together with HP_L and HP_R pins in response to jack + detect events. + +Example: + +codec: wm8960@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + wlf,shared-lrclk; +}; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..99d6457 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = { static int wm8960_add_widgets(struct snd_soc_codec *codec) { - struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct wm8960_data *pdata = &wm8960->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "wlf,capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "wlf,shared-lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (pdata && pdata->shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); @@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); +static const struct of_device_id wm8960_of_match[] = { + { .compatible = "wlf,wm8960", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8960_of_match); + static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", .owner = THIS_MODULE, + .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- cgit v1.1 From ceb3c0683cfc5dcc2b627985143105f6dfb0b324 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:05:38 +0100 Subject: ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 09488d9..3142baf 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -153,15 +153,17 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, CS42L51_POWER_CTL1_PDN); break; default: case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, 0); break; } -- cgit v1.1 From 6e2793b98e23372cc80d9b5d981ab2467e90acea Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 20 Nov 2014 21:05:39 +0100 Subject: ASoC: cs42l73: Replace w->codec snd_soc_dapm_to_codec(w->dapm) The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 2f8b946..7c55537 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -584,7 +584,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMD: @@ -600,7 +600,7 @@ static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w, static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMD: @@ -618,7 +618,7 @@ static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w, static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMD: -- cgit v1.1 From d712eaf29d3fe5928d891a4a90ac58644ad595ed Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Fri, 21 Nov 2014 18:34:48 +0100 Subject: ALSA: core: Deletion of unnecessary checks before two function calls The functions snd_seq_oss_timer_delete() and vunmap() perform also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 9 +++------ sound/core/sgbuf.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b9184d2..b0e32e1 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -403,14 +403,11 @@ free_devinfo(void *private) { struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private; - if (dp->timer) - snd_seq_oss_timer_delete(dp->timer); + snd_seq_oss_timer_delete(dp->timer); - if (dp->writeq) - snd_seq_oss_writeq_delete(dp->writeq); + snd_seq_oss_writeq_delete(dp->writeq); - if (dp->readq) - snd_seq_oss_readq_delete(dp->readq); + snd_seq_oss_readq_delete(dp->readq); kfree(dp); } diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 0a41850..84fffab 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -39,8 +39,7 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; - if (dmab->area) - vunmap(dmab->area); + vunmap(dmab->area); dmab->area = NULL; tmpb.dev.type = SNDRV_DMA_TYPE_DEV; -- cgit v1.1 From 42d772101274492ca15258ea346e6ebbc2bc9bd0 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Fri, 21 Nov 2014 19:05:50 +0100 Subject: ALSA: es1688_lib: Deletion of an unnecessary check before the function call "release_and_free_resource" The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688_lib.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index de810e4..b545014 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -614,8 +614,7 @@ static int snd_es1688_free(struct snd_es1688 *chip) { if (chip->hardware != ES1688_HW_UNDEF) snd_es1688_init(chip, 0); - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); if (chip->dma8 >= 0) { -- cgit v1.1 From 966b7bc9354ab8f59f0ef2d96306615157e0f76e Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Fri, 21 Nov 2014 19:32:02 +0100 Subject: ALSA: sb: Deletion of unnecessary checks before two function calls The functions release_and_free_resource() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_synth.c | 3 +-- sound/isa/sb/sb_common.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 4e3fcfb..95b39be 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -105,8 +105,7 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) snd_device_free(dev->card, hw->pcm); if (hw->emu) snd_emux_free(hw->emu); - if (hw->memhdr) - snd_util_memhdr_free(hw->memhdr); + snd_util_memhdr_free(hw->memhdr); hw->emu = NULL; hw->memhdr = NULL; return 0; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3ef9906..f22b448 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -184,8 +184,7 @@ static int snd_sbdsp_probe(struct snd_sb * chip) static int snd_sbdsp_free(struct snd_sb *chip) { - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); #ifdef CONFIG_ISA -- cgit v1.1 From 026da220c512f6ab706cc9f738439f900b564967 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 21 Nov 2014 16:08:59 +0800 Subject: ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672 Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 12 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/cht_bsw_rt5672.c | 285 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 299 insertions(+) create mode 100644 sound/soc/intel/cht_bsw_rt5672.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a26e8e8..e989ecf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -98,3 +98,15 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index fbde4b07..e928ec3 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -27,12 +27,14 @@ snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c new file mode 100644 index 0000000..9b8b561 --- /dev/null +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -0,0 +1,285 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty + * Mengdong Lin + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include "../codecs/rt5670.h" +#include "sst-atom-controls.h" + +/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */ +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI "rt5670-aif1" + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd; + + rtd = card->rtd + i; + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, + strlen(CHT_CODEC_DAI))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + + return 0; +} + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + {"Headphone", NULL, "Platform Clock"}, + {"Headset Mic", NULL, "Platform Clock"}, + {"Int Mic", NULL, "Platform Clock"}, + {"Ext Spk", NULL, "Platform Clock"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .init = cht_codec_init, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- cgit v1.1 From bd01fdc3aa63b7ba0b035f9196d80551ad03f5d4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 21 Nov 2014 16:09:17 +0800 Subject: ASoC: Intel: add support for Cherrytrail and Braswell in SST driver This patch add ACPI device ID and platform data for two Cherryview-based platforms, Cherrytrail and Braswell. Also reuse mfld driver ops in sst driver. Signed-off-by: Mengdong Lin Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 1 + sound/soc/intel/sst/sst.h | 1 + sound/soc/intel/sst/sst_acpi.c | 19 +++++++++++++++++++ 3 files changed, 21 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 9e68a7c..8a8d56a 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -183,6 +183,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: case SST_BYT_ACPI_ID: + case SST_CHV_ACPI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; return 0; diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 683dc71..7f4bbfc 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -30,6 +30,7 @@ #define SST_DRV_NAME "intel_sst_driver" #define SST_MRFLD_PCI_ID 0x119A #define SST_BYT_ACPI_ID 0x80860F28 +#define SST_CHV_ACPI_ID 0x808622A8 #define SST_SUSPEND_DELAY 2000 #define FW_CONTEXT_MEM (64*1024) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index f94f007..3f29721 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -129,6 +129,17 @@ static struct sst_platform_info byt_rvp_platform_data = { .platform = "sst-mfld-platform", }; +/* Cherryview (Cherrytrail and Braswell) uses same mrfld dpcm fw as Baytrail, + * so pdata is same as Baytrail. + */ +struct sst_platform_info chv_platform_data = { + .probe_data = &byt_fwparse_info, + .ipc_info = &byt_ipc_info, + .lib_info = &byt_lib_dnld_info, + .res_info = &byt_rvp_res_info, + .platform = "sst-mfld-platform", +}; + static int sst_platform_get_resources(struct intel_sst_drv *ctx) { struct resource *rsrc; @@ -337,8 +348,16 @@ static struct sst_machines sst_acpi_bytcr[] = { {}, }; +/* Cherryview-based platforms: CherryTrail and Braswell */ +static struct sst_machines sst_acpi_chv[] = { + {"10EC5670", "cht-bsw", "cht-bsw-rt5672", NULL, "fw_sst_22a8.bin", + &chv_platform_data }, + {}, +}; + static const struct acpi_device_id sst_acpi_ids[] = { { "80860F28", (unsigned long)&sst_acpi_bytcr}, + { "808622A8", (unsigned long) &sst_acpi_chv}, { }, }; -- cgit v1.1 From 075207d24a394bcdb3a864446f391d4014a04cd4 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Mon, 17 Nov 2014 16:02:57 +0800 Subject: ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE Skip dpcm path checking for playback or capture, if corresponding FE doesn't support playback or capture, or currently is not ready. It can reduce the unnecessary cost to search connected widgets. [Tweaked comments for clarity -- broonie] Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..2f4f074 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2248,7 +2248,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) fe->dai_link->name); /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min) + if (!fe->cpu_dai->driver->playback.channels_min + || !fe->codec_dai->driver->playback.channels_min) + goto capture; + + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active + || !fe->codec_dai->playback_active) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2278,7 +2284,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min) + if (!fe->cpu_dai->driver->capture.channels_min + || !fe->codec_dai->driver->capture.channels_min) + continue; + + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active + || !fe->codec_dai->capture_active) continue; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); -- cgit v1.1 From 14cd7923122c7c4473848f7c737604bfe945b81b Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Sat, 22 Nov 2014 04:50:33 +0800 Subject: ASoC: Intel: chv_platform_data can be static sound/soc/intel/sst/sst_acpi.c:135:26: sparse: symbol 'chv_platform_data' was not declared. Should it be static? Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3f29721..31124aa 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -132,7 +132,7 @@ static struct sst_platform_info byt_rvp_platform_data = { /* Cherryview (Cherrytrail and Braswell) uses same mrfld dpcm fw as Baytrail, * so pdata is same as Baytrail. */ -struct sst_platform_info chv_platform_data = { +static struct sst_platform_info chv_platform_data = { .probe_data = &byt_fwparse_info, .ipc_info = &byt_ipc_info, .lib_info = &byt_lib_dnld_info, -- cgit v1.1 From 4fc390a198c41a007c02f82f41ad100c7c67b55e Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 22 Nov 2014 21:29:01 +0300 Subject: sound: oss: uart401: remove unneeded NULL check "devc" can't be NULL here so there is no need to check. Also I removed the "devc = NULL" assignment because devc is stored on stack so it's a no-op. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/uart401.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 279bc56..dae4d43 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -412,13 +412,10 @@ void unload_uart401(struct address_info *hw_config) if (!devc->share_irq) free_irq(devc->irq, devc); - if (devc) - { - kfree(midi_devs[devc->my_dev]->converter); - kfree(midi_devs[devc->my_dev]); - kfree(devc); - devc = NULL; - } + kfree(midi_devs[devc->my_dev]->converter); + kfree(midi_devs[devc->my_dev]); + kfree(devc); + /* This kills midi_devs[x] */ sound_unload_mididev(hw_config->slots[4]); } -- cgit v1.1 From e9886ab06c1ef42451307c9367e344b2d8140e0b Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:48 +1300 Subject: ALSA: asihpi: Minor string and dead code cleanup Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index ff9f9f1c..0e130dd 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -28,7 +28,6 @@ #include "hpioctl.h" #include "hpicmn.h" - #include #include #include @@ -47,7 +46,7 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. "); -MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx " +MODULE_DESCRIPTION("AudioScience ALSA ASI5xxx ASI6xxx ASI87xx ASI89xx " HPI_VER_STRING); #if defined CONFIG_SND_DEBUG_VERBOSE @@ -87,11 +86,11 @@ MODULE_PARM_DESC(enable_hpi_hwdep, #ifdef KERNEL_ALSA_BUILD static char *build_info = "Built using headers from kernel source"; module_param(build_info, charp, S_IRUGO); -MODULE_PARM_DESC(build_info, "built using headers from kernel source"); +MODULE_PARM_DESC(build_info, "Built using headers from kernel source"); #else static char *build_info = "Built within ALSA source"; module_param(build_info, charp, S_IRUGO); -MODULE_PARM_DESC(build_info, "built within ALSA source"); +MODULE_PARM_DESC(build_info, "Built within ALSA source"); #endif /* set to 1 to dump every control from adapter to log */ @@ -538,7 +537,7 @@ static void snd_card_asihpi_pcm_timer_start(struct snd_pcm_substream * int expiry; expiry = HZ / 200; - /*? (dpcm->period_bytes * HZ / dpcm->bytes_per_sec); */ + expiry = max(expiry, 1); /* don't let it be zero! */ dpcm->timer.expires = jiffies + expiry; dpcm->respawn_timer = 1; @@ -2932,10 +2931,6 @@ static struct pci_driver driver = { .id_table = asihpi_pci_tbl, .probe = snd_asihpi_probe, .remove = snd_asihpi_remove, -#ifdef CONFIG_PM_SLEEP -/* .suspend = snd_asihpi_suspend, - .resume = snd_asihpi_resume, */ -#endif }; static int __init snd_asihpi_init(void) -- cgit v1.1 From 3872f19d96a55ec1d1e7af904d84457d91ef5a63 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:49 +1300 Subject: ALSA: asihpi: New I/O types - AVB & BLUlink, DAB Rf receiver Audio cards wth have AVB or BLU Link IO. Tuner card with DAB receiver Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 25 ++++++++++++++++++------- sound/pci/asihpi/hpi.h | 16 ++++++++++++---- 2 files changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 0e130dd..628ef7f 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -109,7 +109,7 @@ static int adapter_fs = DEFAULT_SAMPLERATE; struct clk_source { int source; int index; - char *name; + const char *name; }; struct clk_cache { @@ -1292,8 +1292,9 @@ static const char * const asihpi_tuner_band_names[] = { "TV PAL I", "TV PAL DK", "TV SECAM", + "TV DAB", }; - +/* Number of strings must match the enumerations for HPI_TUNER_BAND in hpi.h */ compile_time_assert( (ARRAY_SIZE(asihpi_tuner_band_names) == (HPI_TUNER_BAND_LAST+1)), @@ -1313,9 +1314,11 @@ static const char * const asihpi_src_names[] = { "Analog", "Adapter", "RTP", - "Internal" + "Internal", + "AVB", + "BLU-Link" }; - +/* Number of strings must match the enumerations for HPI_SOURCENODES in hpi.h */ compile_time_assert( (ARRAY_SIZE(asihpi_src_names) == (HPI_SOURCENODE_LAST_INDEX-HPI_SOURCENODE_NONE+1)), @@ -1331,8 +1334,11 @@ static const char * const asihpi_dst_names[] = { "Net", "Analog", "RTP", + "AVB", + "Internal", + "BLU-Link" }; - +/* Number of strings must match the enumerations for HPI_DESTNODES in hpi.h */ compile_time_assert( (ARRAY_SIZE(asihpi_dst_names) == (HPI_DESTNODE_LAST_INDEX-HPI_DESTNODE_NONE+1)), @@ -2288,13 +2294,18 @@ static int snd_asihpi_cmode_add(struct snd_card_asihpi *asihpi, /*------------------------------------------------------------ Sampleclock source controls ------------------------------------------------------------*/ -static char *sampleclock_sources[MAX_CLOCKSOURCES] = { +static const char const *sampleclock_sources[] = { "N/A", "Local PLL", "Digital Sync", "Word External", "Word Header", "SMPTE", "Digital1", "Auto", "Network", "Invalid", - "Prev Module", + "Prev Module", "BLU-Link", "Digital2", "Digital3", "Digital4", "Digital5", "Digital6", "Digital7", "Digital8"}; + /* Number of strings must match expected enumerated values */ + compile_time_assert( + (ARRAY_SIZE(sampleclock_sources) == MAX_CLOCKSOURCES), + assert_sampleclock_sources_size); + static int snd_asihpi_clksrc_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..4466bd2 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -196,8 +196,10 @@ enum HPI_SOURCENODES { packets of RTP audio samples from other devices. */ HPI_SOURCENODE_RTP_DESTINATION = 112, HPI_SOURCENODE_INTERNAL = 113, /**< node internal to the device. */ + HPI_SOURCENODE_AVB = 114, /**< AVB input stream */ + HPI_SOURCENODE_BLULINK = 115, /**< BLU-link input channel */ /* !!!Update this AND hpidebug.h if you add a new sourcenode type!!! */ - HPI_SOURCENODE_LAST_INDEX = 113 /**< largest ID */ + HPI_SOURCENODE_LAST_INDEX = 115 /**< largest ID */ /* AX6 max sourcenode types = 15 */ }; @@ -224,8 +226,11 @@ enum HPI_DESTNODES { /** RTP stream output node - This node is a source for packets of RTP audio samples that are sent to other devices. */ HPI_DESTNODE_RTP_SOURCE = 208, + HPI_DESTNODE_AVB = 209, /**< AVB output stream */ + HPI_DESTNODE_INTERNAL = 210, /**< node internal to the device. */ + HPI_DESTNODE_BLULINK = 211, /**< BLU-link output channel. */ /* !!!Update this AND hpidebug.h if you add a new destnode type!!! */ - HPI_DESTNODE_LAST_INDEX = 208 /**< largest ID */ + HPI_DESTNODE_LAST_INDEX = 211 /**< largest ID */ /* AX6 max destnode types = 15 */ }; @@ -752,7 +757,8 @@ enum HPI_TUNER_BAND { HPI_TUNER_BAND_TV_PAL_I = 7, /**< PAL-I TV band*/ HPI_TUNER_BAND_TV_PAL_DK = 8, /**< PAL-D/K TV band*/ HPI_TUNER_BAND_TV_SECAM_L = 9, /**< SECAM-L TV band*/ - HPI_TUNER_BAND_LAST = 9 /**< the index of the last tuner band. */ + HPI_TUNER_BAND_DAB = 10, + HPI_TUNER_BAND_LAST = 10 /**< the index of the last tuner band. */ }; /** Tuner mode attributes @@ -842,8 +848,10 @@ enum HPI_SAMPLECLOCK_SOURCES { HPI_SAMPLECLOCK_SOURCE_NETWORK = 8, /** From previous adjacent module (ASI2416 only)*/ HPI_SAMPLECLOCK_SOURCE_PREV_MODULE = 10, +/** Blu link sample clock*/ + HPI_SAMPLECLOCK_SOURCE_BLULINK = 11, /*! Update this if you add a new clock source.*/ - HPI_SAMPLECLOCK_SOURCE_LAST = 10 + HPI_SAMPLECLOCK_SOURCE_LAST = 11 }; /** Equalizer filter types. Used by HPI_ParametricEq_SetBand() -- cgit v1.1 From 35a8dc1f66a0fa88144fcbcd562eb2b2c1e36f11 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:50 +1300 Subject: ALSA: asihpi: Logging format improvements Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 85 ++++++++++++++++++++++------------------------- 1 file changed, 39 insertions(+), 46 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 628ef7f..c069033 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -288,21 +288,17 @@ static void print_hwparams(struct snd_pcm_substream *substream, { char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printd("%s HWPARAMS\n", name); - snd_printd(" samplerate %d Hz\n", params_rate(p)); - snd_printd(" channels %d\n", params_channels(p)); - snd_printd(" format %d\n", params_format(p)); - snd_printd(" subformat %d\n", params_subformat(p)); - snd_printd(" buffer %d B\n", params_buffer_bytes(p)); - snd_printd(" period %d B\n", params_period_bytes(p)); - snd_printd(" access %d\n", params_access(p)); - snd_printd(" period_size %d\n", params_period_size(p)); - snd_printd(" periods %d\n", params_periods(p)); - snd_printd(" buffer_size %d\n", params_buffer_size(p)); - snd_printd(" %d B/s\n", params_rate(p) * - params_channels(p) * + snd_printdd("%s HWPARAMS\n", name); + snd_printdd(" samplerate=%dHz channels=%d format=%d subformat=%d\n", + params_rate(p), params_channels(p), + params_format(p), params_subformat(p)); + snd_printdd(" buffer=%dB period=%dB period_size=%dB periods=%d\n", + params_buffer_bytes(p), params_period_bytes(p), + params_period_size(p), params_periods(p)); + snd_printdd(" buffer_size=%d access=%d data_rate=%dB/s\n", + params_buffer_size(p), params_access(p), + params_rate(p) * params_channels(p) * snd_pcm_format_width(params_format(p)) / 8); - } static snd_pcm_format_t hpi_to_alsa_formats[] = { @@ -480,7 +476,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, params_buffer_bytes(params), runtime->dma_addr); if (err == 0) { snd_printdd( - "stream_host_buffer_attach succeeded %u %lu\n", + "stream_host_buffer_attach success %u %lu\n", params_buffer_bytes(params), (unsigned long)runtime->dma_addr); } else { @@ -490,12 +486,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, } err = hpi_stream_get_info_ex(dpcm->h_stream, NULL, - &dpcm->hpi_buffer_attached, - NULL, NULL, NULL); - - snd_printdd("stream_host_buffer_attach status 0x%x\n", - dpcm->hpi_buffer_attached); - + &dpcm->hpi_buffer_attached, NULL, NULL, NULL); } bytes_per_sec = params_rate(params) * params_channels(params); width = snd_pcm_format_width(params_format(params)); @@ -563,10 +554,10 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printdd("%s trigger\n", name); switch (cmd) { case SNDRV_PCM_TRIGGER_START: + snd_printdd("%s trigger start\n", name); snd_pcm_group_for_each_entry(s, substream) { struct snd_pcm_runtime *runtime = s->runtime; struct snd_card_asihpi_pcm *ds = runtime->private_data; @@ -587,7 +578,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, * data?? */ unsigned int preload = ds->period_bytes * 1; - snd_printddd("%d preload x%x\n", s->number, preload); + snd_printddd("%d preload %d\n", s->number, preload); hpi_handle_error(hpi_outstream_write_buf( ds->h_stream, &runtime->dma_area[0], @@ -610,7 +601,6 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } else break; } - snd_printdd("start\n"); /* start the master stream */ snd_card_asihpi_pcm_timer_start(substream); if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || @@ -619,6 +609,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_STOP: + snd_printdd("%s trigger stop\n", name); snd_card_asihpi_pcm_timer_stop(substream); snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) != card) @@ -637,7 +628,6 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } else break; } - snd_printdd("stop\n"); /* _prepare and _hwparams reset the stream */ hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); @@ -650,12 +640,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_printdd("pause release\n"); + snd_printdd("%s trigger pause release\n", name); hpi_handle_error(hpi_stream_start(dpcm->h_stream)); snd_card_asihpi_pcm_timer_start(substream); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_printdd("pause\n"); + snd_printdd("%s trigger pause push\n", name); snd_card_asihpi_pcm_timer_stop(substream); hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); break; @@ -730,7 +720,6 @@ static void snd_card_asihpi_timer_function(unsigned long data) snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printdd("%s snd_card_asihpi_timer_function\n", name); /* find minimum newdata and buffer pos in group */ snd_pcm_group_for_each_entry(s, substream) { @@ -790,19 +779,20 @@ static void snd_card_asihpi_timer_function(unsigned long data) newdata); } - snd_printdd("hw_ptr 0x%04lX, appl_ptr 0x%04lX\n", + snd_printddd("timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n", + name, s->number, state, + ds->pcm_buf_elapsed_dma_ofs, + ds->pcm_buf_host_rw_ofs, + pcm_buf_dma_ofs, + (int)bytes_avail, + + (int)on_card_bytes, + buffer_size-bytes_avail, (unsigned long)frames_to_bytes(runtime, runtime->status->hw_ptr), (unsigned long)frames_to_bytes(runtime, - runtime->control->appl_ptr)); - - snd_printdd("%d S=%d, " - "rw=0x%04X, dma=0x%04X, left=0x%04X, " - "aux=0x%04X space=0x%04X\n", - s->number, state, - ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, - (int)bytes_avail, - (int)on_card_bytes, buffer_size-bytes_avail); + runtime->control->appl_ptr) + ); loops++; } pcm_buf_dma_ofs = min_buf_pos; @@ -820,7 +810,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) next_jiffies = max(next_jiffies, 1U); dpcm->timer.expires = jiffies + next_jiffies; - snd_printdd("jif %d buf pos 0x%04X newdata 0x%04X xfer 0x%04X\n", + snd_printddd("timer2, jif=%d, buf_pos=%d, newdata=%d, xfer=%d\n", next_jiffies, pcm_buf_dma_ofs, newdata, xfercount); snd_pcm_group_for_each_entry(s, substream) { @@ -852,7 +842,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) } if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - snd_printddd("P%d write1 0x%04X 0x%04X\n", + snd_printddd("write1, P=%d, xfer=%d, buf_ofs=%d\n", s->number, xfer1, buf_ofs); hpi_handle_error( hpi_outstream_write_buf( @@ -862,7 +852,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) if (xfer2) { pd = s->runtime->dma_area; - snd_printddd("P%d write2 0x%04X 0x%04X\n", + snd_printddd("write2, P=%d, xfer=%d, buf_ofs=%d\n", s->number, xfercount - xfer1, buf_ofs); hpi_handle_error( @@ -872,7 +862,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) &ds->format)); } } else { - snd_printddd("C%d read1 0x%04x\n", + snd_printddd("read1, C=%d, xfer=%d\n", s->number, xfer1); hpi_handle_error( hpi_instream_read_buf( @@ -880,7 +870,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) pd, xfer1)); if (xfer2) { pd = s->runtime->dma_area; - snd_printddd("C%d read2 0x%04x\n", + snd_printddd("read2, C=%d, xfer=%d\n", s->number, xfer2); hpi_handle_error( hpi_instream_read_buf( @@ -933,7 +923,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_debug_name(substream, name, sizeof(name)); ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - snd_printddd("%s pointer = 0x%04lx\n", name, (unsigned long)ptr); + snd_printddd("%s, pointer=%ld\n", name, (unsigned long)ptr); return ptr; } @@ -1081,9 +1071,10 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; + char name[16]; + snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printddd("capture pointer %d=%d\n", - substream->number, dpcm->pcm_buf_dma_ofs); + snd_printddd("%s, pointer=%d\n", name, dpcm->pcm_buf_dma_ofs); /* NOTE Unlike playback can't use actual samples_played for the capture position, because those samples aren't yet in the local buffer available for reading. @@ -2867,6 +2858,8 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->in_min_chans = 1; } + snd_printk(KERN_INFO "update_interval_frames: %d", + asihpi->update_interval_frames); snd_printk(KERN_INFO "Has dma:%d, grouping:%d, mrx:%d\n", asihpi->can_dma, asihpi->support_grouping, -- cgit v1.1 From e51c58c982a0f81baa6fdee7331a8700c8586be5 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:51 +1300 Subject: ALSA: asihpi: Use CONFIG_64BIT directly Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 16 ++++++++++------ sound/pci/asihpi/hpios.h | 4 ---- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index bc86cb7..c9bdc28 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -554,17 +554,21 @@ struct hpi_pci { struct pci_dev *pci_dev; }; +/** Adapter specification resource */ +struct hpi_adapter_specification { + u32 type; + u8 modules[4]; +}; + struct hpi_resource { union { const struct hpi_pci *pci; const char *net_if; + struct hpi_adapter_specification adapter_spec; + const void *sw_if; } r; -#ifndef HPI64BIT /* keep structure size constant */ - u32 pad_to64; -#endif u16 bus_type; /* HPI_BUS_PNPISA, _PCI, _USB etc */ u16 padding; - }; /** Format info used inside struct hpi_message @@ -582,7 +586,7 @@ struct hpi_msg_format { struct hpi_msg_data { struct hpi_msg_format format; u8 *pb_data; -#ifndef HPI64BIT +#ifndef CONFIG_64BIT u32 padding; #endif u32 data_size; @@ -595,7 +599,7 @@ struct hpi_data_legacy32 { u32 data_size; }; -#ifdef HPI64BIT +#ifdef CONFIG_64BIT /* Compatibility version of struct hpi_data*/ struct hpi_data_compat32 { struct hpi_msg_format format; diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index d3fbd0d..d17d017 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -41,10 +41,6 @@ HPI Operating System Specific macros for Linux Kernel driver #define HPI_NO_OS_FILE_OPS -#ifdef CONFIG_64BIT -#define HPI64BIT -#endif - /** Details of a memory area allocated with pci_alloc_consistent Need all info for parameters to pci_free_consistent */ -- cgit v1.1 From c1464a885444dd7e9c4491177ee102b64adc46c5 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:52 +1300 Subject: ALSA: asihpi: Refactor control cache code. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpicmn.c | 107 +++++++++++++++++++++++++++------------------- 1 file changed, 63 insertions(+), 44 deletions(-) diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 7ed5c26..c775124 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -206,6 +206,14 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC) struct hpi_control_cache_info *info = (struct hpi_control_cache_info *) &p_master_cache[byte_count]; + u16 control_index = info->control_index; + + if (control_index >= pC->control_count) { + HPI_DEBUG_LOG(INFO, + "adap %d control index %d out of range, cache not ready?\n", + pC->adap_idx, control_index); + return 0; + } if (!info->size_in32bit_words) { if (!i) { @@ -225,10 +233,10 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC) } if (info->control_type) { - pC->p_info[info->control_index] = info; + pC->p_info[control_index] = info; cached++; } else { /* dummy cache entry */ - pC->p_info[info->control_index] = NULL; + pC->p_info[control_index] = NULL; } byte_count += info->size_in32bit_words * 4; @@ -309,35 +317,18 @@ static const struct pad_ofs_size pad_desc[] = { /** CheckControlCache checks the cache and fills the struct hpi_response * accordingly. It returns one if a cache hit occurred, zero otherwise. */ -short hpi_check_control_cache(struct hpi_control_cache *p_cache, +short hpi_check_control_cache_single(struct hpi_control_cache_single *pC, struct hpi_message *phm, struct hpi_response *phr) { - short found = 1; - struct hpi_control_cache_info *pI; - struct hpi_control_cache_single *pC; size_t response_size; - if (!find_control(phm->obj_index, p_cache, &pI)) { - HPI_DEBUG_LOG(VERBOSE, - "HPICMN find_control() failed for adap %d\n", - phm->adapter_index); - return 0; - } - - phr->error = 0; - phr->specific_error = 0; - phr->version = 0; + short found = 1; /* set the default response size */ response_size = sizeof(struct hpi_response_header) + sizeof(struct hpi_control_res); - /* pC is the default cached control strucure. May be cast to - something else in the following switch statement. - */ - pC = (struct hpi_control_cache_single *)pI; - - switch (pI->control_type) { + switch (pC->u.i.control_type) { case HPI_CONTROL_METER: if (phm->u.c.attribute == HPI_METER_PEAK) { @@ -467,7 +458,7 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, break; case HPI_CONTROL_PAD:{ struct hpi_control_cache_pad *p_pad; - p_pad = (struct hpi_control_cache_pad *)pI; + p_pad = (struct hpi_control_cache_pad *)pC; if (!(p_pad->field_valid_flags & (1 << HPI_CTL_ATTR_INDEX(phm->u.c. @@ -531,7 +522,8 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, HPI_DEBUG_LOG(VERBOSE, "%s Adap %d, Ctl %d, Type %d, Attr %d\n", found ? "Cached" : "Uncached", phm->adapter_index, - pI->control_index, pI->control_type, phm->u.c.attribute); + pC->u.i.control_index, pC->u.i.control_type, + phm->u.c.attribute); if (found) { phr->size = (u16)response_size; @@ -543,34 +535,36 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache, return found; } -/** Updates the cache with Set values. - -Only update if no error. -Volume and Level return the limited values in the response, so use these -Multiplexer does so use sent values -*/ -void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache, +short hpi_check_control_cache(struct hpi_control_cache *p_cache, struct hpi_message *phm, struct hpi_response *phr) { - struct hpi_control_cache_single *pC; struct hpi_control_cache_info *pI; - if (phr->error) - return; - if (!find_control(phm->obj_index, p_cache, &pI)) { HPI_DEBUG_LOG(VERBOSE, "HPICMN find_control() failed for adap %d\n", phm->adapter_index); - return; + return 0; } - /* pC is the default cached control strucure. - May be cast to something else in the following switch statement. - */ - pC = (struct hpi_control_cache_single *)pI; + phr->error = 0; + phr->specific_error = 0; + phr->version = 0; + + return hpi_check_control_cache_single((struct hpi_control_cache_single + *)pI, phm, phr); +} + +/** Updates the cache with Set values. - switch (pI->control_type) { +Only update if no error. +Volume and Level return the limited values in the response, so use these +Multiplexer does so use sent values +*/ +void hpi_cmn_control_cache_sync_to_msg_single(struct hpi_control_cache_single + *pC, struct hpi_message *phm, struct hpi_response *phr) +{ + switch (pC->u.i.control_type) { case HPI_CONTROL_VOLUME: if (phm->u.c.attribute == HPI_VOLUME_GAIN) { pC->u.vol.an_log[0] = phr->u.c.an_log_value[0]; @@ -625,6 +619,30 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache, } } +void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache, + struct hpi_message *phm, struct hpi_response *phr) +{ + struct hpi_control_cache_single *pC; + struct hpi_control_cache_info *pI; + + if (phr->error) + return; + + if (!find_control(phm->obj_index, p_cache, &pI)) { + HPI_DEBUG_LOG(VERBOSE, + "HPICMN find_control() failed for adap %d\n", + phm->adapter_index); + return; + } + + /* pC is the default cached control strucure. + May be cast to something else in the following switch statement. + */ + pC = (struct hpi_control_cache_single *)pI; + + hpi_cmn_control_cache_sync_to_msg_single(pC, phm, phr); +} + /** Allocate control cache. \return Cache pointer, or NULL if allocation fails. @@ -637,12 +655,13 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count, if (!p_cache) return NULL; - p_cache->p_info = kcalloc(control_count, sizeof(*p_cache->p_info), - GFP_KERNEL); + p_cache->p_info = + kcalloc(control_count, sizeof(*p_cache->p_info), GFP_KERNEL); if (!p_cache->p_info) { kfree(p_cache); return NULL; } + p_cache->cache_size_in_bytes = size_in_bytes; p_cache->control_count = control_count; p_cache->p_cache = p_dsp_control_buffer; -- cgit v1.1 From f9a376c3f6d77e59d41350901b2bafbaf8791df0 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:53 +1300 Subject: ALSA: asihpi: Add support for stream interrupt. Some cards have a so-called low-latency mode, in which they present a single multichannel stream with no mixing or samplerate conversion. In this mode the card can generate an interrupt per internal processing block (typically 32 or 64 frames) Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 177 +++++++++++++++++++++++++++++++++------- sound/pci/asihpi/hpi6205.c | 43 ++++++++-- sound/pci/asihpi/hpi_internal.h | 4 +- sound/pci/asihpi/hpicmn.h | 19 ++++- sound/pci/asihpi/hpioctl.c | 124 ++++++++++++++++++++++++++-- sound/pci/asihpi/hpios.h | 4 + 6 files changed, 321 insertions(+), 50 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index c069033..ae29f30 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1,6 +1,6 @@ /* * Asihpi soundcard - * Copyright (c) by AudioScience Inc + * Copyright (c) by AudioScience Inc * * This program is free software; you can redistribute it and/or modify * it under the terms of version 2 of the GNU General Public License as @@ -124,6 +124,16 @@ struct snd_card_asihpi { struct pci_dev *pci; struct hpi_adapter *hpi; + /* In low latency mode there is only one stream, a pointer to its + * private data is stored here on trigger and cleared on stop. + * The interrupt handler uses it as a parameter when calling + * snd_card_asihpi_timer_function(). + */ + struct snd_card_asihpi_pcm *llmode_streampriv; + struct tasklet_struct t; + void (*pcm_start)(struct snd_pcm_substream *substream); + void (*pcm_stop)(struct snd_pcm_substream *substream); + u32 h_mixer; struct clk_cache cc; @@ -544,6 +554,48 @@ static void snd_card_asihpi_pcm_timer_stop(struct snd_pcm_substream *substream) del_timer(&dpcm->timer); } +static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream) +{ + struct snd_card_asihpi_pcm *dpcm; + struct snd_card_asihpi *card; + + BUG_ON(!substream); + + dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; + card = snd_pcm_substream_chip(substream); + + BUG_ON(in_interrupt()); + tasklet_disable(&card->t); + card->llmode_streampriv = dpcm; + tasklet_enable(&card->t); + + hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, + HPI_ADAPTER_PROPERTY_IRQ_RATE, + card->update_interval_frames, 0)); +} + +static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream) +{ + struct snd_card_asihpi_pcm *dpcm; + struct snd_card_asihpi *card; + + BUG_ON(!substream); + + dpcm = (struct snd_card_asihpi_pcm *)substream->runtime->private_data; + card = snd_pcm_substream_chip(substream); + + hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, + HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0)); + + if (in_interrupt()) + card->llmode_streampriv = NULL; + else { + tasklet_disable(&card->t); + card->llmode_streampriv = NULL; + tasklet_enable(&card->t); + } +} + static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -602,7 +654,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; } /* start the master stream */ - snd_card_asihpi_pcm_timer_start(substream); + card->pcm_start(substream); if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || !card->can_dma) hpi_handle_error(hpi_stream_start(dpcm->h_stream)); @@ -610,7 +662,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: snd_printdd("%s trigger stop\n", name); - snd_card_asihpi_pcm_timer_stop(substream); + card->pcm_stop(substream); snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) != card) continue; @@ -641,12 +693,12 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: snd_printdd("%s trigger pause release\n", name); + card->pcm_start(substream); hpi_handle_error(hpi_stream_start(dpcm->h_stream)); - snd_card_asihpi_pcm_timer_start(substream); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printdd("%s trigger pause push\n", name); - snd_card_asihpi_pcm_timer_stop(substream); + card->pcm_stop(substream); hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); break; default: @@ -718,8 +770,8 @@ static void snd_card_asihpi_timer_function(unsigned long data) u32 buffer_size, bytes_avail, samples_played, on_card_bytes; char name[16]; - snd_pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); /* find minimum newdata and buffer pos in group */ snd_pcm_group_for_each_entry(s, substream) { @@ -779,7 +831,8 @@ static void snd_card_asihpi_timer_function(unsigned long data) newdata); } - snd_printddd("timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n", + snd_printddd( + "timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n", name, s->number, state, ds->pcm_buf_elapsed_dma_ofs, ds->pcm_buf_host_rw_ofs, @@ -815,11 +868,13 @@ static void snd_card_asihpi_timer_function(unsigned long data) snd_pcm_group_for_each_entry(s, substream) { struct snd_card_asihpi_pcm *ds = s->runtime->private_data; + runtime = s->runtime; /* don't link Cap and Play */ if (substream->stream != s->stream) continue; + /* Store dma offset for use by pointer callback */ ds->pcm_buf_dma_ofs = pcm_buf_dma_ofs; if (xfercount && @@ -878,16 +933,38 @@ static void snd_card_asihpi_timer_function(unsigned long data) pd, xfer2)); } } + /* ? host_rw_ofs always ahead of elapsed_dma_ofs by preload size? */ ds->pcm_buf_host_rw_ofs += xfercount; ds->pcm_buf_elapsed_dma_ofs += xfercount; snd_pcm_period_elapsed(s); } } - if (dpcm->respawn_timer) + if (!card->hpi->interrupt_mode && dpcm->respawn_timer) add_timer(&dpcm->timer); } +static void snd_card_asihpi_int_task(unsigned long data) +{ + struct hpi_adapter *a = (struct hpi_adapter *)data; + struct snd_card_asihpi *asihpi; + + WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); + asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; + if (asihpi->llmode_streampriv) + snd_card_asihpi_timer_function( + (unsigned long)asihpi->llmode_streampriv); +} + +static void snd_card_asihpi_isr(struct hpi_adapter *a) +{ + struct snd_card_asihpi *asihpi; + + WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); + asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; + tasklet_schedule(&asihpi->t); +} + /***************************** PLAYBACK OPS ****************/ static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) @@ -995,13 +1072,22 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_card_asihpi_runtime_free; memset(&snd_card_asihpi_playback, 0, sizeof(snd_card_asihpi_playback)); - snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX; - snd_card_asihpi_playback.period_bytes_min = PERIOD_BYTES_MIN; - /*?snd_card_asihpi_playback.period_bytes_min = - card->out_max_chans * 4096; */ - snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; - snd_card_asihpi_playback.periods_min = PERIODS_MIN; - snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + if (!card->hpi->interrupt_mode) { + snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_playback.period_bytes_min = PERIOD_BYTES_MIN; + snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_playback.periods_min = PERIODS_MIN; + snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + } else { + size_t pbmin = card->update_interval_frames * + card->out_max_chans; + snd_card_asihpi_playback.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_playback.period_bytes_min = pbmin; + snd_card_asihpi_playback.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_playback.periods_min = PERIODS_MIN; + snd_card_asihpi_playback.periods_max = BUFFER_BYTES_MAX / pbmin; + } + /* snd_card_asihpi_playback.fifo_size = 0; */ snd_card_asihpi_playback.channels_max = card->out_max_chans; snd_card_asihpi_playback.channels_min = card->out_min_chans; @@ -1036,7 +1122,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) card->update_interval_frames); snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - card->update_interval_frames * 2, UINT_MAX); + card->update_interval_frames, UINT_MAX); snd_printdd("playback open\n"); @@ -1102,8 +1188,6 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) return 0; } - - static u64 snd_card_asihpi_capture_formats(struct snd_card_asihpi *asihpi, u32 h_stream) { @@ -1170,11 +1254,21 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) runtime->private_free = snd_card_asihpi_runtime_free; memset(&snd_card_asihpi_capture, 0, sizeof(snd_card_asihpi_capture)); - snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX; - snd_card_asihpi_capture.period_bytes_min = PERIOD_BYTES_MIN; - snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; - snd_card_asihpi_capture.periods_min = PERIODS_MIN; - snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + if (!card->hpi->interrupt_mode) { + snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_capture.period_bytes_min = PERIOD_BYTES_MIN; + snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_capture.periods_min = PERIODS_MIN; + snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN; + } else { + size_t pbmin = card->update_interval_frames * + card->out_max_chans; + snd_card_asihpi_capture.buffer_bytes_max = BUFFER_BYTES_MAX; + snd_card_asihpi_capture.period_bytes_min = pbmin; + snd_card_asihpi_capture.period_bytes_max = BUFFER_BYTES_MAX / PERIODS_MIN; + snd_card_asihpi_capture.periods_min = PERIODS_MIN; + snd_card_asihpi_capture.periods_max = BUFFER_BYTES_MAX / pbmin; + } /* snd_card_asihpi_capture.fifo_size = 0; */ snd_card_asihpi_capture.channels_max = card->in_max_chans; snd_card_asihpi_capture.channels_min = card->in_min_chans; @@ -1199,7 +1293,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames); snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - card->update_interval_frames * 2, UINT_MAX); + card->update_interval_frames, UINT_MAX); snd_pcm_set_sync(substream); @@ -2444,15 +2538,19 @@ static int snd_asihpi_clkrate_get(struct snd_kcontrol *kcontrol, static int snd_asihpi_sampleclock_add(struct snd_card_asihpi *asihpi, struct hpi_control *hpi_ctl) { - struct snd_card *card = asihpi->card; + struct snd_card *card; struct snd_kcontrol_new snd_control; - struct clk_cache *clkcache = &asihpi->cc; + struct clk_cache *clkcache; u32 hSC = hpi_ctl->h_control; int has_aes_in = 0; int i, j; u16 source; + if (snd_BUG_ON(!asihpi)) + return -EINVAL; + card = asihpi->card; + clkcache = &asihpi->cc; snd_control.private_value = hpi_ctl->h_control; clkcache->has_local = 0; @@ -2808,6 +2906,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->card = card; asihpi->pci = pci_dev; asihpi->hpi = hpi; + hpi->snd_card = card; snd_printk(KERN_INFO "adapter ID=%4X index=%d\n", asihpi->hpi->adapter->type, adapter_index); @@ -2830,8 +2929,16 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, if (err) asihpi->update_interval_frames = 512; - if (!asihpi->can_dma) - asihpi->update_interval_frames *= 2; + if (hpi->interrupt_mode) { + asihpi->pcm_start = snd_card_asihpi_pcm_int_start; + asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop; + tasklet_init(&asihpi->t, snd_card_asihpi_int_task, + (unsigned long)hpi); + hpi->interrupt_callback = snd_card_asihpi_isr; + } else { + asihpi->pcm_start = snd_card_asihpi_pcm_timer_start; + asihpi->pcm_stop = snd_card_asihpi_pcm_timer_stop; + } hpi_handle_error(hpi_instream_open(adapter_index, 0, &h_stream)); @@ -2841,6 +2948,9 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, hpi_handle_error(hpi_instream_close(h_stream)); + if (!asihpi->can_dma) + asihpi->update_interval_frames *= 2; + err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_CURCHANNELS, &asihpi->in_max_chans, &asihpi->out_max_chans); @@ -2900,7 +3010,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, err = snd_card_register(card); if (!err) { - hpi->snd_card = card; dev++; return 0; } @@ -2914,6 +3023,16 @@ __nodev: static void snd_asihpi_remove(struct pci_dev *pci_dev) { struct hpi_adapter *hpi = pci_get_drvdata(pci_dev); + struct snd_card_asihpi *asihpi = hpi->snd_card->private_data; + + /* Stop interrupts */ + if (hpi->interrupt_mode) { + hpi->interrupt_callback = NULL; + hpi_handle_error(hpi_adapter_set_property(hpi->adapter->index, + HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0)); + tasklet_kill(&asihpi->t); + } + snd_card_free(hpi->snd_card); hpi->snd_card = NULL; asihpi_adapter_remove(pci_dev); diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 4f28738..8d5abfa 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -163,6 +163,9 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, static void delete_adapter_obj(struct hpi_adapter_obj *pao); +static int adapter_irq_query_and_clear(struct hpi_adapter_obj *pao, + u32 message); + static void outstream_host_buffer_allocate(struct hpi_adapter_obj *pao, struct hpi_message *phm, struct hpi_response *phr); @@ -283,7 +286,6 @@ static void adapter_message(struct hpi_adapter_obj *pao, case HPI_ADAPTER_DELETE: adapter_delete(pao, phm, phr); break; - default: hw_message(pao, phm, phr); break; @@ -673,6 +675,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, HPI_DEBUG_LOG(INFO, "bootload DSP OK\n"); + pao->irq_query_and_clear = adapter_irq_query_and_clear; + pao->instream_host_buffer_status = + phw->p_interface_buffer->instream_host_buffer_status; + pao->outstream_host_buffer_status = + phw->p_interface_buffer->outstream_host_buffer_status; + return hpi_add_adapter(pao); } @@ -713,6 +721,21 @@ static void delete_adapter_obj(struct hpi_adapter_obj *pao) /*****************************************************************************/ /* Adapter functions */ +static int adapter_irq_query_and_clear(struct hpi_adapter_obj *pao, + u32 message) +{ + struct hpi_hw_obj *phw = pao->priv; + u32 hsr = 0; + + hsr = ioread32(phw->prHSR); + if (hsr & C6205_HSR_INTSRC) { + /* reset the interrupt from the DSP */ + iowrite32(C6205_HSR_INTSRC, phw->prHSR); + return HPI_IRQ_MIXER; + } + + return HPI_IRQ_NONE; +} /*****************************************************************************/ /* OutStream Host buffer functions */ @@ -1331,17 +1354,21 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao, if (boot_code_id[1] != 0) { /* DSP 1 is a C6713 */ /* CLKX0 <- '1' release the C6205 bootmode pulldowns */ - boot_loader_write_mem32(pao, 0, (0x018C0024L), 0x00002202); + boot_loader_write_mem32(pao, 0, 0x018C0024, 0x00002202); hpios_delay_micro_seconds(100); /* Reset the 6713 #1 - revB */ boot_loader_write_mem32(pao, 0, C6205_BAR0_TIMER1_CTL, 0); - - /* dummy read every 4 words for 6205 advisory 1.4.4 */ - boot_loader_read_mem32(pao, 0, 0); - + /* value of bit 3 is unknown after DSP reset, other bits shoudl be 0 */ + if (0 != (boot_loader_read_mem32(pao, 0, + (C6205_BAR0_TIMER1_CTL)) & ~8)) + return HPI6205_ERROR_6205_REG; hpios_delay_micro_seconds(100); + /* Release C6713 from reset - revB */ boot_loader_write_mem32(pao, 0, C6205_BAR0_TIMER1_CTL, 4); + if (4 != (boot_loader_read_mem32(pao, 0, + (C6205_BAR0_TIMER1_CTL)) & ~8)) + return HPI6205_ERROR_6205_REG; hpios_delay_micro_seconds(100); } @@ -2089,7 +2116,7 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao, return 0; } - /* Assume buffer of type struct bus_master_interface + /* Assume buffer of type struct bus_master_interface_62 is allocated "noncacheable" */ if (!wait_dsp_ack(phw, H620_HIF_IDLE, HPI6205_TIMEOUT)) { diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index c9bdc28..48380ce 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -686,8 +686,8 @@ union hpi_adapterx_msg { u16 value; } test_assert; struct { - u32 yes; - } irq_query; + u32 message; + } irq; u32 pad[3]; }; diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h index e441212..46629c2 100644 --- a/sound/pci/asihpi/hpicmn.h +++ b/sound/pci/asihpi/hpicmn.h @@ -1,7 +1,7 @@ /** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -21,7 +21,11 @@ struct hpi_adapter_obj; /* a function that takes an adapter obj and returns an int */ -typedef int adapter_int_func(struct hpi_adapter_obj *pao); +typedef int adapter_int_func(struct hpi_adapter_obj *pao, u32 message); + +#define HPI_IRQ_NONE (0) +#define HPI_IRQ_MESSAGE (1) +#define HPI_IRQ_MIXER (2) struct hpi_adapter_obj { struct hpi_pci pci; /* PCI info - bus#,dev#,address etc */ @@ -33,6 +37,9 @@ struct hpi_adapter_obj { u16 dsp_crashed; u16 has_control_cache; void *priv; + adapter_int_func *irq_query_and_clear; + struct hpi_hostbuffer_status *instream_host_buffer_status; + struct hpi_hostbuffer_status *outstream_host_buffer_status; }; struct hpi_control_cache { @@ -55,13 +62,21 @@ void hpi_delete_adapter(struct hpi_adapter_obj *pao); short hpi_check_control_cache(struct hpi_control_cache *pC, struct hpi_message *phm, struct hpi_response *phr); + +short hpi_check_control_cache_single(struct hpi_control_cache_single *pC, + struct hpi_message *phm, struct hpi_response *phr); + struct hpi_control_cache *hpi_alloc_control_cache(const u32 number_of_controls, const u32 size_in_bytes, u8 *pDSP_control_buffer); + void hpi_free_control_cache(struct hpi_control_cache *p_cache); void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *pC, struct hpi_message *phm, struct hpi_response *phr); +void hpi_cmn_control_cache_sync_to_msg_single(struct hpi_control_cache_single + *pC, struct hpi_message *phm, struct hpi_response *phr); + u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr); hpi_handler_func HPI_COMMON; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 7f02720..9454932 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -1,7 +1,8 @@ /******************************************************************************* - AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Common Linux HPI ioctl and module probe/remove functions + + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -12,11 +13,6 @@ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. - You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - -Common Linux HPI ioctl and module probe/remove functions *******************************************************************************/ #define SOURCEFILE_NAME "hpioctl.c" @@ -29,6 +25,7 @@ Common Linux HPI ioctl and module probe/remove functions #include "hpicmn.h" #include +#include #include #include #include @@ -307,10 +304,38 @@ out: return err; } +static int asihpi_irq_count; + +static irqreturn_t asihpi_isr(int irq, void *dev_id) +{ + struct hpi_adapter *a = dev_id; + int handled; + + if (!a->adapter->irq_query_and_clear) { + pr_err("asihpi_isr ASI%04X:%d no handler\n", a->adapter->type, + a->adapter->index); + return IRQ_NONE; + } + + handled = a->adapter->irq_query_and_clear(a->adapter, 0); + + if (!handled) + return IRQ_NONE; + + asihpi_irq_count++; + /* printk(KERN_INFO "asihpi_isr %d ASI%04X:%d irq handled\n", + asihpi_irq_count, a->adapter->type, a->adapter->index); */ + + if (a->interrupt_callback) + a->interrupt_callback(a); + + return IRQ_HANDLED; +} + int asihpi_adapter_probe(struct pci_dev *pci_dev, const struct pci_device_id *pci_id) { - int idx, nm; + int idx, nm, low_latency_mode = 0, irq_supported = 0; int adapter_index; unsigned int memlen; struct hpi_message hm; @@ -388,8 +413,39 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, hm.adapter_index = adapter.adapter->index; hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); - if (hr.error) + if (hr.error) { + HPI_DEBUG_LOG(ERROR, "HPI_ADAPTER_OPEN failed, aborting\n"); + goto err; + } + + /* Check if current mode == Low Latency mode */ + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_GET_MODE); + hm.adapter_index = adapter.adapter->index; + hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); + + if (hr.error) { + HPI_DEBUG_LOG(ERROR, + "HPI_ADAPTER_GET_MODE failed, aborting\n"); goto err; + } + + if (hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY) + low_latency_mode = 1; + + /* Check if IRQs are supported */ + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_GET_PROPERTY); + hm.adapter_index = adapter.adapter->index; + hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_SUPPORTS_IRQ; + hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); + if (hr.error || !hr.u.ax.property_get.parameter1) { + dev_info(&pci_dev->dev, + "IRQs not supported by adapter at index %d\n", + adapter.adapter->index); + } else { + irq_supported = 1; + } /* WARNING can't init mutex in 'adapter' * and then copy it to adapters[] ?!?! @@ -398,6 +454,44 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, mutex_init(&adapters[adapter_index].mutex); pci_set_drvdata(pci_dev, &adapters[adapter_index]); + if (low_latency_mode && irq_supported) { + if (!adapter.adapter->irq_query_and_clear) { + dev_err(&pci_dev->dev, + "no IRQ handler for adapter %d, aborting\n", + adapter.adapter->index); + goto err; + } + + /* Disable IRQ generation on DSP side by setting the rate to 0 */ + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_SET_PROPERTY); + hm.adapter_index = adapter.adapter->index; + hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_IRQ_RATE; + hm.u.ax.property_set.parameter1 = 0; + hm.u.ax.property_set.parameter2 = 0; + hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); + if (hr.error) { + HPI_DEBUG_LOG(ERROR, + "HPI_ADAPTER_GET_MODE failed, aborting\n"); + goto err; + } + + /* Note: request_irq calls asihpi_isr here */ + if (request_irq(pci_dev->irq, asihpi_isr, IRQF_SHARED, + "asihpi", &adapters[adapter_index])) { + dev_err(&pci_dev->dev, "request_irq(%d) failed\n", + pci_dev->irq); + goto err; + } + + adapters[adapter_index].interrupt_mode = 1; + + dev_info(&pci_dev->dev, "using irq %d\n", pci_dev->irq); + adapters[adapter_index].irq = pci_dev->irq; + } else { + dev_info(&pci_dev->dev, "using polled mode\n"); + } + dev_info(&pci_dev->dev, "probe succeeded for ASI%04X HPI index %d\n", adapter.adapter->type, adapter_index); @@ -431,6 +525,15 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) pa = pci_get_drvdata(pci_dev); pci = pa->adapter->pci; + /* Disable IRQ generation on DSP side */ + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_SET_PROPERTY); + hm.adapter_index = pa->adapter->index; + hm.u.ax.property_set.property = HPI_ADAPTER_PROPERTY_IRQ_RATE; + hm.u.ax.property_set.parameter1 = 0; + hm.u.ax.property_set.parameter2 = 0; + hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, HPI_ADAPTER_DELETE); hm.adapter_index = pa->adapter->index; @@ -442,6 +545,9 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) iounmap(pci.ap_mem_base[idx]); } + if (pa->irq) + free_irq(pa->irq, pa); + if (pa->p_buffer) vfree(pa->p_buffer); diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index d17d017..4e38360 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -151,6 +151,10 @@ struct hpi_adapter { struct hpi_adapter_obj *adapter; struct snd_card *snd_card; + int irq; + int interrupt_mode; + void (*interrupt_callback) (struct hpi_adapter *); + /* mutex prevents contention for one card between multiple user programs (via ioctl) */ struct mutex mutex; -- cgit v1.1 From 5bc91f5b3c732bdb3b9e7cc8bd27969d25015bcd Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:54 +1300 Subject: ALSA: asihpi: Turn off msg/resp logging after DSP has crashed. Prevents spewing of useless messages if app keeps trying to access the card. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpimsgx.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index d4790dd..736f453 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -35,6 +35,7 @@ static struct pci_device_id asihpi_pci_tbl[] = { static struct hpios_spinlock msgx_lock; static hpi_handler_func *hpi_entry_points[HPI_MAX_ADAPTERS]; +static int logging_enabled = 1; static hpi_handler_func *hpi_lookup_entry_point_function(const struct hpi_pci *pci_info) @@ -312,7 +313,9 @@ static void instream_message(struct hpi_message *phm, void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr, void *h_owner) { - HPI_DEBUG_MESSAGE(DEBUG, phm); + + if (logging_enabled) + HPI_DEBUG_MESSAGE(DEBUG, phm); if (phm->type != HPI_TYPE_REQUEST) { hpi_init_response(phr, phm->object, phm->function, @@ -352,8 +355,14 @@ void hpi_send_recv_ex(struct hpi_message *phm, struct hpi_response *phr, hw_entry_point(phm, phr); break; } - HPI_DEBUG_RESPONSE(phr); + if (logging_enabled) + HPI_DEBUG_RESPONSE(phr); + + if (phr->error >= HPI_ERROR_DSP_COMMUNICATION) { + hpi_debug_level_set(HPI_DEBUG_LEVEL_ERROR); + logging_enabled = 0; + } } static void adapter_open(struct hpi_message *phm, struct hpi_response *phr) -- cgit v1.1 From 12eb0898741870882ca474708e811983d5a5d768 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:55 +1300 Subject: ALSA: asihpi: Use standard printk helpers Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index ae29f30..e9273fb 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -380,7 +380,7 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, HPI_SOURCENODE_CLOCK_SOURCE, 0, 0, 0, HPI_CONTROL_SAMPLECLOCK, &h_control); if (err) { - snd_printk(KERN_ERR + dev_err(&asihpi->pci->dev, "No local sampleclock, err %d\n", err); } @@ -1438,7 +1438,7 @@ static inline int ctl_add(struct snd_card *card, struct snd_kcontrol_new *ctl, if (err < 0) return err; else if (mixer_dump) - snd_printk(KERN_INFO "added %s(%d)\n", ctl->name, ctl->index); + dev_info(&asihpi->pci->dev, "added %s(%d)\n", ctl->name, ctl->index); return 0; } @@ -2652,7 +2652,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (err) { if (err == HPI_ERROR_CONTROL_DISABLED) { if (mixer_dump) - snd_printk(KERN_INFO + dev_info(&asihpi->pci->dev, "Disabled HPI Control(%d)\n", idx); continue; @@ -2717,9 +2717,8 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) case HPI_CONTROL_COMPANDER: default: if (mixer_dump) - snd_printk(KERN_INFO - "Untranslated HPI Control" - "(%d) %d %d %d %d %d\n", + dev_info(&asihpi->pci->dev, + "Untranslated HPI Control (%d) %d %d %d %d %d\n", idx, hpi_ctl.control_type, hpi_ctl.src_node_type, @@ -2734,7 +2733,7 @@ static int snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi) if (HPI_ERROR_INVALID_OBJ_INDEX != err) hpi_handle_error(err); - snd_printk(KERN_INFO "%d mixer controls found\n", idx); + dev_info(&asihpi->pci->dev, "%d mixer controls found\n", idx); return 0; } @@ -2897,8 +2896,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, &card); if (err < 0) return err; - snd_printk(KERN_WARNING - "**** WARNING **** Adapter index %d->ALSA index %d\n", + dev_warn(&pci_dev->dev, "Adapter index %d->ALSA index %d\n", adapter_index, card->number); } @@ -2908,9 +2906,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->hpi = hpi; hpi->snd_card = card; - snd_printk(KERN_INFO "adapter ID=%4X index=%d\n", - asihpi->hpi->adapter->type, adapter_index); - err = hpi_adapter_get_property(adapter_index, HPI_ADAPTER_PROPERTY_CAPS1, NULL, &asihpi->support_grouping); @@ -2968,22 +2963,21 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->in_min_chans = 1; } - snd_printk(KERN_INFO "update_interval_frames: %d", - asihpi->update_interval_frames); - snd_printk(KERN_INFO "Has dma:%d, grouping:%d, mrx:%d\n", + dev_info(&pci_dev->dev, "Has dma:%d, grouping:%d, mrx:%d, uif:%d\n", asihpi->can_dma, asihpi->support_grouping, - asihpi->support_mrx + asihpi->support_mrx, + asihpi->update_interval_frames ); err = snd_card_asihpi_pcm_new(asihpi, 0); if (err < 0) { - snd_printk(KERN_ERR "pcm_new failed\n"); + dev_err(&pci_dev->dev, "pcm_new failed\n"); goto __nodev; } err = snd_card_asihpi_mixer_new(asihpi); if (err < 0) { - snd_printk(KERN_ERR "mixer_new failed\n"); + dev_err(&pci_dev->dev, "mixer_new failed\n"); goto __nodev; } @@ -3015,7 +3009,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, } __nodev: snd_card_free(card); - snd_printk(KERN_ERR "snd_asihpi_probe error %d\n", err); + dev_err(&pci_dev->dev, "snd_asihpi_probe error %d\n", err); return err; } -- cgit v1.1 From dc612838eac746b11bb4e5d923dafeea0ba7e81b Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:56 +1300 Subject: ALSA: asihpi: don't fail probe if adapter mode read fails Only determining if low latency mode is enabled. Failure indicates adapter has no modes Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpioctl.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 9454932..e457eb8 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -424,14 +424,13 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, hm.adapter_index = adapter.adapter->index; hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); - if (hr.error) { - HPI_DEBUG_LOG(ERROR, - "HPI_ADAPTER_GET_MODE failed, aborting\n"); - goto err; - } - - if (hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY) + if (!hr.error + && hr.u.ax.mode.adapter_mode == HPI_ADAPTER_MODE_LOW_LATENCY) low_latency_mode = 1; + else + dev_info(&pci_dev->dev, + "Adapter at index %d is not in low latency mode\n", + adapter.adapter->index); /* Check if IRQs are supported */ hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, -- cgit v1.1 From 51e6f47dd2e3463dac6f37128fd7b7cb40c500de Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Thu, 20 Nov 2014 16:22:57 +1300 Subject: ALSA: asihpi: used parts of message/response are zeroed before use Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpimsginit.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/pci/asihpi/hpimsginit.c b/sound/pci/asihpi/hpimsginit.c index 032d563..7eb6171 100644 --- a/sound/pci/asihpi/hpimsginit.c +++ b/sound/pci/asihpi/hpimsginit.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2014 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -37,11 +37,15 @@ static u16 gwSSX2_bypass; static void hpi_init_message(struct hpi_message *phm, u16 object, u16 function) { - memset(phm, 0, sizeof(*phm)); + u16 size; + if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) - phm->size = msg_size[object]; + size = msg_size[object]; else - phm->size = sizeof(*phm); + size = sizeof(*phm); + + memset(phm, 0, size); + phm->size = size; if (gwSSX2_bypass) phm->type = HPI_TYPE_SSX2BYPASS_MESSAGE; @@ -60,12 +64,16 @@ static void hpi_init_message(struct hpi_message *phm, u16 object, void hpi_init_response(struct hpi_response *phr, u16 object, u16 function, u16 error) { - memset(phr, 0, sizeof(*phr)); - phr->type = HPI_TYPE_RESPONSE; + u16 size; + if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) - phr->size = res_size[object]; + size = res_size[object]; else - phr->size = sizeof(*phr); + size = sizeof(*phr); + + memset(phr, 0, sizeof(*phr)); + phr->size = size; + phr->type = HPI_TYPE_RESPONSE; phr->object = object; phr->function = function; phr->error = error; @@ -86,7 +94,7 @@ void hpi_init_message_response(struct hpi_message *phm, static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size, u16 object, u16 function) { - memset(phm, 0, sizeof(*phm)); + memset(phm, 0, size); if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) { phm->size = size; phm->type = HPI_TYPE_REQUEST; @@ -100,7 +108,9 @@ static void hpi_init_messageV1(struct hpi_message_header *phm, u16 size, void hpi_init_responseV1(struct hpi_response_header *phr, u16 size, u16 object, u16 function) { - memset(phr, 0, sizeof(*phr)); + (void)object; + (void)function; + memset(phr, 0, size); phr->size = size; phr->version = 1; phr->type = HPI_TYPE_RESPONSE; -- cgit v1.1 From 4cf703a7bca4c29d06028821db60f253390a84a7 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Nov 2014 15:32:35 +0200 Subject: ASoC: max98090: Fix digital microphone Commit e409dfbfccf9 ("ASoC: dapm: Add a few supply widget sanity checks") broke digital microphone support in max98090.c: max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICL_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICL_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICL_ENA -> DMIC -> DMIC Mux max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICR_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICR_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICR_ENA -> DMIC -> DMIC Mux Problem is partially caused by commit f69e3caa9e18 ("ASoC: max98090: Enable both DMIC channels also when using mono configuration") which connects "DMICL_ENA" and "DMICR_ENA" supply widgets to "DMIC Mux". Fix the breakage by reverting f69e3caa9e18 and then by adding additional "DMICR_ENA" to "DMICL" and "DMICL_ENA" to "DMICR" cross-connections. This disconnects these supply widgets from the mux and makes sure that both DMIC data channels are still enabled together. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1229554..994d02c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,6 +1311,10 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICL", NULL, "DMICR_ENA"}, + {"DMICR", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1368,8 +1372,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, - {"DMIC Mux", "DMIC", "DMICL_ENA"}, - {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- cgit v1.1 From 48826ee590da03e9882922edf96d8d27bdfe9552 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Nov 2014 15:32:36 +0200 Subject: ASoC: max98090: Fix ill-defined sidetone route Commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") revealed ill-defined control in a route between "STENL Mux" and DACs in max98090.c: max98090 i2c-193C9890:00: Control not supported for path STENL Mux -> [NULL] -> DACL max98090 i2c-193C9890:00: ASoC: no dapm match for STENL Mux --> NULL --> DACL max98090 i2c-193C9890:00: ASoC: Failed to add route STENL Mux -> NULL -> DACL max98090 i2c-193C9890:00: Control not supported for path STENL Mux -> [NULL] -> DACR max98090 i2c-193C9890:00: ASoC: no dapm match for STENL Mux --> NULL --> DACR max98090 i2c-193C9890:00: ASoC: Failed to add route STENL Mux -> NULL -> DACR Since there is no control between "STENL Mux" and DACs the control name must be NULL not "NULL". Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/max98090.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 994d02c..20b50e4 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1397,8 +1397,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"STENL Mux", "Sidetone Left", "DMICL"}, {"STENR Mux", "Sidetone Right", "ADCR"}, {"STENR Mux", "Sidetone Right", "DMICR"}, - {"DACL", "NULL", "STENL Mux"}, - {"DACR", "NULL", "STENL Mux"}, + {"DACL", NULL, "STENL Mux"}, + {"DACR", NULL, "STENL Mux"}, {"AIFINL", NULL, "SHDN"}, {"AIFINR", NULL, "SHDN"}, -- cgit v1.1 From 418382f29d99f1faffdd6636f378da41b44815db Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 24 Nov 2014 15:32:37 +0200 Subject: ASoC: max98090: Fix right sidetone connection It is right not left sidetone which goes to "DACR". Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 20b50e4..34ed9a9 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1398,7 +1398,7 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"STENR Mux", "Sidetone Right", "ADCR"}, {"STENR Mux", "Sidetone Right", "DMICR"}, {"DACL", NULL, "STENL Mux"}, - {"DACR", NULL, "STENL Mux"}, + {"DACR", NULL, "STENR Mux"}, {"AIFINL", NULL, "SHDN"}, {"AIFINR", NULL, "SHDN"}, -- cgit v1.1 From a6e4599f8d232b5911c46bb16f5a79b86f3dfb75 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 15:04:12 +0100 Subject: ASoC: uda134x: Remove is_powered_on_standby from platform data According to its documentation the is_powered_on_standby field of the uda134x platform data is supposed to prevent the the driver from shutting down the ADC and DAC in standby mode. This behavior was broken in commit commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") almost 5 years ago and all the flag does now is cause the driver to go to SND_SOC_BIAS_ON in probe, just for the ASoC core to put it back into SND_SOC_BIAS_STANDBY right after probe. Apparently the intended behavior has not been missed, so just remove is_powered_on_standby from the platform data struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/uda134x.h | 12 ------------ sound/soc/codecs/uda134x.c | 5 +---- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index e475659b..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,18 +18,6 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; - /* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you unset is_powered_on_standby the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - set is_powered_on_standby, you will have slightly higher power - consumption. Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 32b2f78..54240f1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,10 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) uda134x_reset(codec); - if (pd->is_powered_on_standby) - uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); - else - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (pd->model == UDA134X_UDA1341) { widgets = uda1341_dapm_widgets; -- cgit v1.1 From e03b975506545d21b1daa5c8310b59d66e74919c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 15:04:13 +0100 Subject: ASoC: uda134x: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 29 ++--------------------------- 1 file changed, 2 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 54240f1..4056260 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,8 +518,6 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) uda134x_reset(codec); - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (pd->model == UDA134X_UDA1341) { widgets = uda1341_dapm_widgets; num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); @@ -571,44 +569,21 @@ static int uda134x_soc_remove(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); - kfree(uda134x); return 0; } -#if defined(CONFIG_PM) -static int uda134x_soc_suspend(struct snd_soc_codec *codec) -{ - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int uda134x_soc_resume(struct snd_soc_codec *codec) -{ - uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); - return 0; -} -#else -#define uda134x_soc_suspend NULL -#define uda134x_soc_resume NULL -#endif /* CONFIG_PM */ - static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .probe = uda134x_soc_probe, .remove = uda134x_soc_remove, - .suspend = uda134x_soc_suspend, - .resume = uda134x_soc_resume, .reg_cache_size = sizeof(uda134x_reg), .reg_word_size = sizeof(u8), .reg_cache_default = uda134x_reg, .reg_cache_step = 1, .read = uda134x_read_reg_cache, - .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = uda134x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), .dapm_routes = uda134x_dapm_routes, -- cgit v1.1 From e8125f04421f7757df0017a59cd9b756148ee769 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 15:04:14 +0100 Subject: ASoC: uda1380: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 20 ++------------------ 1 file changed, 2 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e62e707..dc7778b 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -693,18 +693,6 @@ static struct snd_soc_dai_driver uda1380_dai[] = { }, }; -static int uda1380_suspend(struct snd_soc_codec *codec) -{ - uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int uda1380_resume(struct snd_soc_codec *codec) -{ - uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int uda1380_probe(struct snd_soc_codec *codec) { struct uda1380_platform_data *pdata =codec->dev->platform_data; @@ -739,8 +727,6 @@ static int uda1380_probe(struct snd_soc_codec *codec) INIT_WORK(&uda1380->work, uda1380_flush_work); - /* power on device */ - uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set clock input */ switch (pdata->dac_clk) { case UDA1380_DAC_CLK_SYSCLK: @@ -766,8 +752,6 @@ static int uda1380_remove(struct snd_soc_codec *codec) { struct uda1380_platform_data *pdata =codec->dev->platform_data; - uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - gpio_free(pdata->gpio_reset); gpio_free(pdata->gpio_power); @@ -777,11 +761,11 @@ static int uda1380_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_uda1380 = { .probe = uda1380_probe, .remove = uda1380_remove, - .suspend = uda1380_suspend, - .resume = uda1380_resume, .read = uda1380_read_reg_cache, .write = uda1380_write, .set_bias_level = uda1380_set_bias_level, + .suspend_bias_off = true, + .reg_cache_size = ARRAY_SIZE(uda1380_reg), .reg_word_size = sizeof(u16), .reg_cache_default = uda1380_reg, -- cgit v1.1 From 2849bde56aac38645c5ed2af3971358b89a929f6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:47:50 +0100 Subject: ASoC: alc5623: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the manual sequencing back to SND_SOC_BIAS_ON in resume as this is already handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { -- cgit v1.1 From 5c9dc0898f343473efa3056fff9d5a9fbd577272 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:47:51 +0100 Subject: ASoC: alc5632: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 22 ++-------------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 85942ca..d1fdbc2 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1038,23 +1038,15 @@ static struct snd_soc_dai_driver alc5632_dai = { }; #ifdef CONFIG_PM -static int alc5632_suspend(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int alc5632_resume(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); regcache_sync(alc5632->regmap); - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } #else -#define alc5632_suspend NULL #define alc5632_resume NULL #endif @@ -1062,9 +1054,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - /* power on device */ - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - switch (alc5632->id) { case 0x5c: snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls, @@ -1077,19 +1066,12 @@ static int alc5632_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5632_remove(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, - .remove = alc5632_remove, - .suspend = alc5632_suspend, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, + .suspend_bias_off = true, + .controls = alc5632_snd_controls, .num_controls = ARRAY_SIZE(alc5632_snd_controls), .dapm_widgets = alc5632_dapm_widgets, -- cgit v1.1 From e2dce944cc2bf22d0295330cbdcbd2ad7bd47cb4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:47:52 +0100 Subject: ASoC: rt5631: Cleanup bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 27 +-------------------------- 1 file changed, 1 insertion(+), 26 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 9425545..6d7b7ca 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1612,29 +1612,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) return 0; } -static int rt5631_remove(struct snd_soc_codec *codec) -{ - rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -#ifdef CONFIG_PM -static int rt5631_suspend(struct snd_soc_codec *codec) -{ - rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int rt5631_resume(struct snd_soc_codec *codec) -{ - rt5631_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define rt5631_suspend NULL -#define rt5631_resume NULL -#endif - #define RT5631_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5631_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ @@ -1672,10 +1649,8 @@ static struct snd_soc_dai_driver rt5631_dai[] = { static struct snd_soc_codec_driver soc_codec_dev_rt5631 = { .probe = rt5631_probe, - .remove = rt5631_remove, - .suspend = rt5631_suspend, - .resume = rt5631_resume, .set_bias_level = rt5631_set_bias_level, + .suspend_bias_off = true, .controls = rt5631_snd_controls, .num_controls = ARRAY_SIZE(rt5631_snd_controls), .dapm_widgets = rt5631_dapm_widgets, -- cgit v1.1 From 21a942fdd85efde65512f1458bcb952fda88886e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:17 +0100 Subject: ASoC: wm8350: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 628ec77..87f664b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1242,19 +1242,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8350_resume(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) @@ -1565,9 +1552,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); - - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -1596,8 +1580,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) * wait for its completion */ flush_delayed_work(&codec->dapm.delayed_work); - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); return 0; @@ -1613,10 +1595,9 @@ static struct regmap *wm8350_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .probe = wm8350_codec_probe, .remove = wm8350_codec_remove, - .suspend = wm8350_suspend, - .resume = wm8350_resume, .get_regmap = wm8350_get_regmap, .set_bias_level = wm8350_set_bias_level, + .suspend_bias_off = true, .controls = wm8350_snd_controls, .num_controls = ARRAY_SIZE(wm8350_snd_controls), -- cgit v1.1 From 098f6f17c3f1beeccdce78f9722ccaa7925b8041 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:18 +0100 Subject: ASoC: wm8400: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual asynchronous transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Also running this asynchronously has the problem of potential race conditions with the core. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 34 +--------------------------------- 1 file changed, 1 insertion(+), 33 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 72471be..385894f 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -58,12 +58,10 @@ static struct regulator_bulk_data power[] = { /* codec private data */ struct wm8400_priv { - struct snd_soc_codec *codec; struct wm8400 *wm8400; u16 fake_register; unsigned int sysclk; unsigned int pcmclk; - struct work_struct work; int fll_in, fll_out; }; @@ -1278,30 +1276,6 @@ static struct snd_soc_dai_driver wm8400_dai = { .ops = &wm8400_dai_ops, }; -static int wm8400_suspend(struct snd_soc_codec *codec) -{ - wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8400_resume(struct snd_soc_codec *codec) -{ - wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static void wm8400_probe_deferred(struct work_struct *work) -{ - struct wm8400_priv *priv = container_of(work, struct wm8400_priv, - work); - struct snd_soc_codec *codec = priv->codec; - - /* charge output caps */ - wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - static int wm8400_codec_probe(struct snd_soc_codec *codec) { struct wm8400 *wm8400 = dev_get_platdata(codec->dev); @@ -1316,7 +1290,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->wm8400 = wm8400; - priv->codec = codec; ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); @@ -1325,8 +1298,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) return ret; } - INIT_WORK(&priv->work, wm8400_probe_deferred); - wm8400_codec_reset(codec); reg = snd_soc_read(codec, WM8400_POWER_MANAGEMENT_1); @@ -1343,8 +1314,6 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - if (!schedule_work(&priv->work)) - return -EINVAL; return 0; } @@ -1369,10 +1338,9 @@ static struct regmap *wm8400_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_wm8400 = { .probe = wm8400_codec_probe, .remove = wm8400_codec_remove, - .suspend = wm8400_suspend, - .resume = wm8400_resume, .get_regmap = wm8400_get_regmap, .set_bias_level = wm8400_set_bias_level, + .suspend_bias_off = true, .controls = wm8400_snd_controls, .num_controls = ARRAY_SIZE(wm8400_snd_controls), -- cgit v1.1 From 99b108c73f3876d71ac6631e85e0f093e53b7e66 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:19 +0100 Subject: ASoC: wm8510: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 26 +------------------------- 1 file changed, 1 insertion(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index e11127f..8736ad0 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -575,41 +575,17 @@ static struct snd_soc_dai_driver wm8510_dai = { .symmetric_rates = 1, }; -static int wm8510_suspend(struct snd_soc_codec *codec) -{ - wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8510_resume(struct snd_soc_codec *codec) -{ - wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8510_probe(struct snd_soc_codec *codec) { wm8510_reset(codec); - /* power on device */ - wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* power down chip */ -static int wm8510_remove(struct snd_soc_codec *codec) -{ - wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .probe = wm8510_probe, - .remove = wm8510_remove, - .suspend = wm8510_suspend, - .resume = wm8510_resume, .set_bias_level = wm8510_set_bias_level, + .suspend_bias_off = true, .controls = wm8510_snd_controls, .num_controls = ARRAY_SIZE(wm8510_snd_controls), -- cgit v1.1 From ca5e7c6afff94b4e103d79db835bc2990d3d340e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:20 +0100 Subject: ASoC: wm8523: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 29 +---------------------------- 1 file changed, 1 insertion(+), 28 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index ec1f574..b1cc94f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -372,23 +372,6 @@ static struct snd_soc_dai_driver wm8523_dai = { .ops = &wm8523_dai_ops, }; -#ifdef CONFIG_PM -static int wm8523_suspend(struct snd_soc_codec *codec) -{ - wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8523_resume(struct snd_soc_codec *codec) -{ - wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8523_suspend NULL -#define wm8523_resume NULL -#endif - static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); @@ -402,23 +385,13 @@ static int wm8523_probe(struct snd_soc_codec *codec) WM8523_DACR_VU, WM8523_DACR_VU); snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); - wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int wm8523_remove(struct snd_soc_codec *codec) -{ - wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .probe = wm8523_probe, - .remove = wm8523_remove, - .suspend = wm8523_suspend, - .resume = wm8523_resume, .set_bias_level = wm8523_set_bias_level, + .suspend_bias_off = true, .controls = wm8523_controls, .num_controls = ARRAY_SIZE(wm8523_controls), -- cgit v1.1 From 4d0a4c3c6dd2359c3d5facac7a306d513d79bff2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:21 +0100 Subject: ASoC: wm8580: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 911605e..0a887c5 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -882,8 +882,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) goto err_regulator_enable; } - wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; err_regulator_enable: @@ -897,8 +895,6 @@ static int wm8580_remove(struct snd_soc_codec *codec) { struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); - wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); return 0; -- cgit v1.1 From 0bd324b1ad5c0922ac3f157763123d1550bdffd7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:22 +0100 Subject: ASoC: wm8711: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the extra write that sets the WM8711_ACTIVE register to 0x00 in the suspend handler since this write is already done when transitioning to SND_SOC_BIAS_OFF. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 27 ++------------------------- 1 file changed, 2 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 32187e7..121e46d 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -350,19 +350,6 @@ static struct snd_soc_dai_driver wm8711_dai = { .ops = &wm8711_ops, }; -static int wm8711_suspend(struct snd_soc_codec *codec) -{ - snd_soc_write(codec, WM8711_ACTIVE, 0x0); - wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8711_resume(struct snd_soc_codec *codec) -{ - wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8711_probe(struct snd_soc_codec *codec) { int ret; @@ -373,8 +360,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) return ret; } - wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch the update bits */ snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100); @@ -383,19 +368,11 @@ static int wm8711_probe(struct snd_soc_codec *codec) } -/* power down chip */ -static int wm8711_remove(struct snd_soc_codec *codec) -{ - wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .probe = wm8711_probe, - .remove = wm8711_remove, - .suspend = wm8711_suspend, - .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, + .suspend_bias_off = true, + .controls = wm8711_snd_controls, .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, -- cgit v1.1 From d4d41436ff3b1fddf2f8feafa6772647eac6b61d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:23 +0100 Subject: ASoC: wm8728: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 34 ++-------------------------------- 1 file changed, 2 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 38ff826..55c7fb4 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -212,40 +212,10 @@ static struct snd_soc_dai_driver wm8728_dai = { .ops = &wm8728_dai_ops, }; -static int wm8728_suspend(struct snd_soc_codec *codec) -{ - wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8728_resume(struct snd_soc_codec *codec) -{ - wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int wm8728_probe(struct snd_soc_codec *codec) -{ - /* power on device */ - wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int wm8728_remove(struct snd_soc_codec *codec) -{ - wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { - .probe = wm8728_probe, - .remove = wm8728_remove, - .suspend = wm8728_suspend, - .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, + .suspend_bias_off = true, + .controls = wm8728_snd_controls, .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, -- cgit v1.1 From 2081b2cf05d022ac6245334e8baa25f589e5635a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:24 +0100 Subject: ASoC: wm8731: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 25 ++----------------------- 1 file changed, 2 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2c9f2a7..3b3786e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -559,25 +559,6 @@ static struct snd_soc_dai_driver wm8731_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int wm8731_suspend(struct snd_soc_codec *codec) -{ - wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8731_resume(struct snd_soc_codec *codec) -{ - wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm8731_suspend NULL -#define wm8731_resume NULL -#endif - static int wm8731_probe(struct snd_soc_codec *codec) { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); @@ -633,8 +614,6 @@ static int wm8731_remove(struct snd_soc_codec *codec) { struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); return 0; @@ -643,9 +622,9 @@ static int wm8731_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .probe = wm8731_probe, .remove = wm8731_remove, - .suspend = wm8731_suspend, - .resume = wm8731_resume, .set_bias_level = wm8731_set_bias_level, + .suspend_bias_off = true, + .dapm_widgets = wm8731_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = wm8731_intercon, -- cgit v1.1 From 67cac3a351b9d411f8736a180767f0e898b50423 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:25 +0100 Subject: ASoC: wm8737: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 27 +-------------------------- 1 file changed, 1 insertion(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index fe41dd2..ada9ac1 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -537,23 +537,6 @@ static struct snd_soc_dai_driver wm8737_dai = { .ops = &wm8737_dai_ops, }; -#ifdef CONFIG_PM -static int wm8737_suspend(struct snd_soc_codec *codec) -{ - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8737_resume(struct snd_soc_codec *codec) -{ - wm8737_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8737_suspend NULL -#define wm8737_resume NULL -#endif - static int wm8737_probe(struct snd_soc_codec *codec) { struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); @@ -590,18 +573,10 @@ err_get: return ret; } -static int wm8737_remove(struct snd_soc_codec *codec) -{ - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .probe = wm8737_probe, - .remove = wm8737_remove, - .suspend = wm8737_suspend, - .resume = wm8737_resume, .set_bias_level = wm8737_set_bias_level, + .suspend_bias_off = true, .controls = wm8737_snd_controls, .num_controls = ARRAY_SIZE(wm8737_snd_controls), -- cgit v1.1 From d12cbf956f428229bb29fb58dee8729e16873ca7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:26 +0100 Subject: ASoC: wm8750: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8750.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 67653a2..f6847fd 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -686,18 +686,6 @@ static struct snd_soc_dai_driver wm8750_dai = { .ops = &wm8750_dai_ops, }; -static int wm8750_suspend(struct snd_soc_codec *codec) -{ - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8750_resume(struct snd_soc_codec *codec) -{ - wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8750_probe(struct snd_soc_codec *codec) { int ret; @@ -708,9 +696,6 @@ static int wm8750_probe(struct snd_soc_codec *codec) return ret; } - /* charge output caps */ - wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* set the update bits */ snd_soc_update_bits(codec, WM8750_LDAC, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8750_RDAC, 0x0100, 0x0100); @@ -724,18 +709,10 @@ static int wm8750_probe(struct snd_soc_codec *codec) return ret; } -static int wm8750_remove(struct snd_soc_codec *codec) -{ - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8750 = { .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, .set_bias_level = wm8750_set_bias_level, + .suspend_bias_off = true, .controls = wm8750_snd_controls, .num_controls = ARRAY_SIZE(wm8750_snd_controls), -- cgit v1.1 From 6c286afb01cc641e2a78e485467e4a90aedfbd75 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:27 +0100 Subject: ASoC: wm8776: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 31 +------------------------------ 1 file changed, 1 insertion(+), 30 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 70952ce..c13050b 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -408,24 +408,6 @@ static struct snd_soc_dai_driver wm8776_dai[] = { }, }; -#ifdef CONFIG_PM -static int wm8776_suspend(struct snd_soc_codec *codec) -{ - wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8776_resume(struct snd_soc_codec *codec) -{ - wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8776_suspend NULL -#define wm8776_resume NULL -#endif - static int wm8776_probe(struct snd_soc_codec *codec) { int ret = 0; @@ -436,8 +418,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) return ret; } - wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch the update bits; right channel only since we always * update both. */ snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); @@ -446,19 +426,10 @@ static int wm8776_probe(struct snd_soc_codec *codec) return ret; } -/* power down chip */ -static int wm8776_remove(struct snd_soc_codec *codec) -{ - wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .probe = wm8776_probe, - .remove = wm8776_remove, - .suspend = wm8776_suspend, - .resume = wm8776_resume, .set_bias_level = wm8776_set_bias_level, + .suspend_bias_off = true, .controls = wm8776_snd_controls, .num_controls = ARRAY_SIZE(wm8776_snd_controls), -- cgit v1.1 From a4235a14bef979752fb2ddb4dafdb696f622beb0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:28 +0100 Subject: ASoC: wm8804: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 3addc5f..1315f76 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -524,7 +524,6 @@ static int wm8804_remove(struct snd_soc_codec *codec) int i; wm8804 = snd_soc_codec_get_drvdata(codec); - wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF); for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) regulator_unregister_notifier(wm8804->supplies[i].consumer, @@ -606,8 +605,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) goto err_reg_enable; } - wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; err_reg_enable: -- cgit v1.1 From d2a9bc68512696a896abf7a94852c5fb5e6733a1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:29 +0100 Subject: ASoC: wm8900: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 44a5f15..3a0d4b7 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1209,16 +1209,8 @@ static int wm8900_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int wm8900_remove(struct snd_soc_codec *codec) -{ - wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .probe = wm8900_probe, - .remove = wm8900_remove, .suspend = wm8900_suspend, .resume = wm8900_resume, .set_bias_level = wm8900_set_bias_level, -- cgit v1.1 From b0d55b1a63ea3c3d694c58694d93f74bea61215f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:30 +0100 Subject: ASoC: wm8903: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Also remove the unused codec field from the wm8903_priv struct so we can remove the whole probe callback. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 35 ++--------------------------------- 1 file changed, 2 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c038b3e..9758d2e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -117,7 +117,6 @@ static const struct reg_default wm8903_reg_defaults[] = { struct wm8903_priv { struct wm8903_platform_data *pdata; struct device *dev; - struct snd_soc_codec *codec; struct regmap *regmap; int sysclk; @@ -1757,21 +1756,12 @@ static struct snd_soc_dai_driver wm8903_dai = { .symmetric_rates = 1, }; -static int wm8903_suspend(struct snd_soc_codec *codec) -{ - wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static int wm8903_resume(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); regcache_sync(wm8903->regmap); - wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -1889,33 +1879,12 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) } #endif -static int wm8903_probe(struct snd_soc_codec *codec) -{ - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - wm8903->codec = codec; - - /* power on device */ - wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* power down chip */ -static int wm8903_remove(struct snd_soc_codec *codec) -{ - wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { - .probe = wm8903_probe, - .remove = wm8903_remove, - .suspend = wm8903_suspend, .resume = wm8903_resume, .set_bias_level = wm8903_set_bias_level, .seq_notifier = wm8903_seq_notifier, + .suspend_bias_off = true, + .controls = wm8903_snd_controls, .num_controls = ARRAY_SIZE(wm8903_snd_controls), .dapm_widgets = wm8903_dapm_widgets, -- cgit v1.1 From 5fdf082b43995ae31d746d3d9e3b616afa24c542 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:31 +0100 Subject: ASoC: wm8940: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 22 ++-------------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 5201104..e4142b4 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -695,17 +695,6 @@ static struct snd_soc_dai_driver wm8940_dai = { .symmetric_rates = 1, }; -static int wm8940_suspend(struct snd_soc_codec *codec) -{ - return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); -} - -static int wm8940_resume(struct snd_soc_codec *codec) -{ - wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8940_probe(struct snd_soc_codec *codec) { struct wm8940_setup_data *pdata = codec->dev->platform_data; @@ -736,18 +725,11 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } -static int wm8940_remove(struct snd_soc_codec *codec) -{ - wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .probe = wm8940_probe, - .remove = wm8940_remove, - .suspend = wm8940_suspend, - .resume = wm8940_resume, .set_bias_level = wm8940_set_bias_level, + .suspend_bias_off = true, + .controls = wm8940_snd_controls, .num_controls = ARRAY_SIZE(wm8940_snd_controls), .dapm_widgets = wm8940_dapm_widgets, -- cgit v1.1 From bf68a0470876b5bf43758c50a7585eb5f6e177ea Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:32 +0100 Subject: ASoC: wm8955: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the regcache_mark_dirty() from the suspend handler since this is already done by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 33 +-------------------------------- 1 file changed, 1 insertion(+), 32 deletions(-) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 09d91d9..1173f7f 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -866,29 +866,6 @@ static struct snd_soc_dai_driver wm8955_dai = { .ops = &wm8955_dai_ops, }; -#ifdef CONFIG_PM -static int wm8955_suspend(struct snd_soc_codec *codec) -{ - struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - - wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); - - regcache_mark_dirty(wm8955->regmap); - - return 0; -} - -static int wm8955_resume(struct snd_soc_codec *codec) -{ - wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm8955_suspend NULL -#define wm8955_resume NULL -#endif - static int wm8955_probe(struct snd_soc_codec *codec) { struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); @@ -964,18 +941,10 @@ err_enable: return ret; } -static int wm8955_remove(struct snd_soc_codec *codec) -{ - wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8955 = { .probe = wm8955_probe, - .remove = wm8955_remove, - .suspend = wm8955_suspend, - .resume = wm8955_resume, .set_bias_level = wm8955_set_bias_level, + .suspend_bias_off = true, .controls = wm8955_snd_controls, .num_controls = ARRAY_SIZE(wm8955_snd_controls), -- cgit v1.1 From 0a87a6e1c09c3b93d91bf65809e79cf6cf358785 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:33 +0100 Subject: ASoC: wm8960: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 31 +------------------------------ 1 file changed, 1 insertion(+), 30 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 99d6457..bc8793cd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -943,22 +943,6 @@ static struct snd_soc_dai_driver wm8960_dai = { .symmetric_rates = 1, }; -static int wm8960_suspend(struct snd_soc_codec *codec) -{ - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - - wm8960->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8960_resume(struct snd_soc_codec *codec) -{ - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - - wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); @@ -976,8 +960,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) return ret; } - wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -997,21 +979,10 @@ static int wm8960_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int wm8960_remove(struct snd_soc_codec *codec) -{ - struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - - wm8960->set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .probe = wm8960_probe, - .remove = wm8960_remove, - .suspend = wm8960_suspend, - .resume = wm8960_resume, .set_bias_level = wm8960_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config wm8960_regmap = { -- cgit v1.1 From 7bea32c5b2493044d31a2116328c71c7048de0e3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:34 +0100 Subject: ASoC: wm8961: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index e077bb2..eeffd05 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -870,44 +870,26 @@ static int wm8961_probe(struct snd_soc_codec *codec) reg &= ~WM8961_MANUAL_MODE; snd_soc_write(codec, WM8961_CLOCKING_3, reg); - wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int wm8961_remove(struct snd_soc_codec *codec) -{ - wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } #ifdef CONFIG_PM -static int wm8961_suspend(struct snd_soc_codec *codec) -{ - wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} static int wm8961_resume(struct snd_soc_codec *codec) { snd_soc_cache_sync(codec); - wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } #else -#define wm8961_suspend NULL #define wm8961_resume NULL #endif static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .probe = wm8961_probe, - .remove = wm8961_remove, - .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, + .suspend_bias_off = true, .controls = wm8961_snd_controls, .num_controls = ARRAY_SIZE(wm8961_snd_controls), -- cgit v1.1 From 387fe80fb13ed9f5f3741a661f96e409a2c959b5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:37 +0100 Subject: ASoC: wm8983: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 27 +-------------------------- 1 file changed, 1 insertion(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index ac5defd..5d1cf08 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -967,29 +967,6 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8983_suspend(struct snd_soc_codec *codec) -{ - wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8983_resume(struct snd_soc_codec *codec) -{ - wm8983_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8983_suspend NULL -#define wm8983_resume NULL -#endif - -static int wm8983_remove(struct snd_soc_codec *codec) -{ - wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int wm8983_probe(struct snd_soc_codec *codec) { int ret; @@ -1055,10 +1032,8 @@ static struct snd_soc_dai_driver wm8983_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8983 = { .probe = wm8983_probe, - .remove = wm8983_remove, - .suspend = wm8983_suspend, - .resume = wm8983_resume, .set_bias_level = wm8983_set_bias_level, + .suspend_bias_off = true, .controls = wm8983_snd_controls, .num_controls = ARRAY_SIZE(wm8983_snd_controls), .dapm_widgets = wm8983_dapm_widgets, -- cgit v1.1 From d02486fd42a3295edbec4db8f7f81c1432fa60a4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:36 +0100 Subject: ASoC: wm8978: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index ee2ba57..cf70329 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -991,21 +991,11 @@ static int wm8978_probe(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(update_reg); i++) snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); - wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -/* power down chip */ -static int wm8978_remove(struct snd_soc_codec *codec) -{ - wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .probe = wm8978_probe, - .remove = wm8978_remove, .suspend = wm8978_suspend, .resume = wm8978_resume, .set_bias_level = wm8978_set_bias_level, -- cgit v1.1 From ed1358f508e1ebcb01e1e545c5330599098b7687 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:35 +0100 Subject: ASoC: wm8974: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 682e9ed..ff0e464 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -568,18 +568,6 @@ static struct snd_soc_dai_driver wm8974_dai = { .symmetric_rates = 1, }; -static int wm8974_suspend(struct snd_soc_codec *codec) -{ - wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8974_resume(struct snd_soc_codec *codec) -{ - wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static const struct regmap_config wm8974_regmap = { .reg_bits = 7, .val_bits = 9, @@ -599,24 +587,13 @@ static int wm8974_probe(struct snd_soc_codec *codec) return ret; } - wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return ret; -} - -/* power down chip */ -static int wm8974_remove(struct snd_soc_codec *codec) -{ - wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .probe = wm8974_probe, - .remove = wm8974_remove, - .suspend = wm8974_suspend, - .resume = wm8974_resume, .set_bias_level = wm8974_set_bias_level, + .suspend_bias_off = true, .controls = wm8974_snd_controls, .num_controls = ARRAY_SIZE(wm8974_snd_controls), -- cgit v1.1 From a5dde8c42ef6b3cb47c69905ee51520e18ac6515 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:38 +0100 Subject: ASoC: wm8985: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 28 +--------------------------- 1 file changed, 1 insertion(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ee38019..0b3b54c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -961,29 +961,6 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, return 0; } -#ifdef CONFIG_PM -static int wm8985_suspend(struct snd_soc_codec *codec) -{ - wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8985_resume(struct snd_soc_codec *codec) -{ - wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} -#else -#define wm8985_suspend NULL -#define wm8985_resume NULL -#endif - -static int wm8985_remove(struct snd_soc_codec *codec) -{ - wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int wm8985_probe(struct snd_soc_codec *codec) { size_t i; @@ -1023,7 +1000,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8985_BIAS_CTRL, WM8985_BIASCUT, WM8985_BIASCUT); - wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; err_reg_enable: @@ -1064,10 +1040,8 @@ static struct snd_soc_dai_driver wm8985_dai = { static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .probe = wm8985_probe, - .remove = wm8985_remove, - .suspend = wm8985_suspend, - .resume = wm8985_resume, .set_bias_level = wm8985_set_bias_level, + .suspend_bias_off = true, .controls = wm8985_snd_controls, .num_controls = ARRAY_SIZE(wm8985_snd_controls), -- cgit v1.1 From 1f07b8de451f5f4f6a268a95a34183e528cda711 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:39 +0100 Subject: ASoC: wm8988: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the regcache_mark_dirty() from the suspend handler since it is already called by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 27 +-------------------------- 1 file changed, 1 insertion(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index a5130d9..e418199 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -793,21 +793,6 @@ static struct snd_soc_dai_driver wm8988_dai = { .symmetric_rates = 1, }; -static int wm8988_suspend(struct snd_soc_codec *codec) -{ - struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); - - wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); - regcache_mark_dirty(wm8988->regmap); - return 0; -} - -static int wm8988_resume(struct snd_soc_codec *codec) -{ - wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int wm8988_probe(struct snd_soc_codec *codec) { int ret = 0; @@ -825,23 +810,13 @@ static int wm8988_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8988_ROUT2V, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8988_RINVOL, 0x0100, 0x0100); - wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int wm8988_remove(struct snd_soc_codec *codec) -{ - wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .probe = wm8988_probe, - .remove = wm8988_remove, - .suspend = wm8988_suspend, - .resume = wm8988_resume, .set_bias_level = wm8988_set_bias_level, + .suspend_bias_off = true, .controls = wm8988_snd_controls, .num_controls = ARRAY_SIZE(wm8988_snd_controls), -- cgit v1.1 From 955efc8f50eb11d1c85daca6db7943c63dc5c2e7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:40 +0100 Subject: ASoC: wm8990: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 03e43e3..8a58422 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1271,18 +1271,6 @@ static struct snd_soc_dai_driver wm8990_dai = { .ops = &wm8990_dai_ops, }; -static int wm8990_suspend(struct snd_soc_codec *codec) -{ - wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8990_resume(struct snd_soc_codec *codec) -{ - wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - /* * initialise the WM8990 driver * register the mixer and dsp interfaces with the kernel @@ -1309,19 +1297,11 @@ static int wm8990_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int wm8990_remove(struct snd_soc_codec *codec) -{ - wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8990 = { .probe = wm8990_probe, - .remove = wm8990_remove, - .suspend = wm8990_suspend, - .resume = wm8990_resume, .set_bias_level = wm8990_set_bias_level, + .suspend_bias_off = true, + .controls = wm8990_snd_controls, .num_controls = ARRAY_SIZE(wm8990_snd_controls), .dapm_widgets = wm8990_dapm_widgets, -- cgit v1.1 From 497b900f83c56d513794ccf56b7a87c50a34a454 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:41 +0100 Subject: ASoC: wm8991: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 32 ++------------------------------ 1 file changed, 2 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index d0be897..b0ac2c3 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1227,32 +1227,6 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8991_suspend(struct snd_soc_codec *codec) -{ - wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8991_resume(struct snd_soc_codec *codec) -{ - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - -/* power down chip */ -static int wm8991_remove(struct snd_soc_codec *codec) -{ - wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8991_probe(struct snd_soc_codec *codec) -{ - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - #define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1293,11 +1267,9 @@ static struct snd_soc_dai_driver wm8991_dai = { }; static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { - .probe = wm8991_probe, - .remove = wm8991_remove, - .suspend = wm8991_suspend, - .resume = wm8991_resume, .set_bias_level = wm8991_set_bias_level, + .suspend_bias_off = true, + .controls = wm8991_snd_controls, .num_controls = ARRAY_SIZE(wm8991_snd_controls), .dapm_widgets = wm8991_dapm_widgets, -- cgit v1.1 From 77d05e7f81da95eb2b6c7ae24ae0fb3272c49282 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:42 +0100 Subject: ASoC: wm8993: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 93b14ed..53c6fe3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1486,7 +1486,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; wm8993->hubs_data.hp_startup_mode = 1; wm8993->hubs_data.dcs_codes_l = -2; @@ -1518,10 +1517,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); - ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (ret != 0) - return ret; - snd_soc_add_codec_controls(codec, wm8993_snd_controls, ARRAY_SIZE(wm8993_snd_controls)); if (wm8993->pdata.num_retune_configs != 0) { @@ -1550,12 +1545,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) } -static int wm8993_remove(struct snd_soc_codec *codec) -{ - wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - #ifdef CONFIG_PM static int wm8993_suspend(struct snd_soc_codec *codec) { @@ -1629,7 +1618,6 @@ static const struct regmap_config wm8993_regmap = { static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .probe = wm8993_probe, - .remove = wm8993_remove, .suspend = wm8993_suspend, .resume = wm8993_resume, .set_bias_level = wm8993_set_bias_level, -- cgit v1.1 From 49d9ac383cddc3e8d4cae8bc7a8f4da9dc071121 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:43 +0100 Subject: ASoC: wm8994: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1fcb9f3..c3a2e75 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4391,8 +4391,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) struct wm8994 *control = wm8994->wm8994; int i; - wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); -- cgit v1.1 From aee9ffabec81d96d68d8537ccc6fedfbb0e6c468 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:44 +0100 Subject: ASoC: wm8995: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index e40c8a6..c280f0a 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2004,7 +2004,6 @@ static int wm8995_remove(struct snd_soc_codec *codec) int i; wm8995 = snd_soc_codec_get_drvdata(codec); - wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF); for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i) regulator_unregister_notifier(wm8995->supplies[i].consumer, @@ -2078,8 +2077,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) goto err_reg_enable; } - wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch volume updates (right only; we always do left then right). */ snd_soc_update_bits(codec, WM8995_AIF1_DAC1_RIGHT_VOLUME, WM8995_AIF1DAC1_VU_MASK, WM8995_AIF1DAC1_VU); -- cgit v1.1 From 68201d6998a0d7701f7c3009130c5d8cd6ad7562 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:45 +0100 Subject: ASoC: wm9081: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 0cdc9e2..b1d946f 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1277,15 +1277,8 @@ static int wm9081_probe(struct snd_soc_codec *codec) return 0; } -static int wm9081_remove(struct snd_soc_codec *codec) -{ - wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .probe = wm9081_probe, - .remove = wm9081_remove, .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, -- cgit v1.1 From a70cf928ca396447416422c2ec1697a530534ac9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:46 +0100 Subject: ASoC: wm9090: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 32 +------------------------------- 1 file changed, 1 insertion(+), 31 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a13f072..6ffe8dc 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -550,45 +550,15 @@ static int wm9090_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM9090_CLOCKING_1, WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); - wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm9090_add_controls(codec); return 0; } -#ifdef CONFIG_PM -static int wm9090_suspend(struct snd_soc_codec *codec) -{ - wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm9090_resume(struct snd_soc_codec *codec) -{ - wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm9090_suspend NULL -#define wm9090_resume NULL -#endif - -static int wm9090_remove(struct snd_soc_codec *codec) -{ - wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm9090 = { .probe = wm9090_probe, - .remove = wm9090_remove, - .suspend = wm9090_suspend, - .resume = wm9090_resume, .set_bias_level = wm9090_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config wm9090_regmap = { -- cgit v1.1 From ab492b86b89be6c98bdca71cfc97b411ca42e140 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:47 +0100 Subject: ASoC: wm9712: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend. This makes the code a bit shorter and cleaner. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f3aab6e..30f4b17 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -599,12 +599,6 @@ err: return -EIO; } -static int wm9712_soc_suspend(struct snd_soc_codec *codec) -{ - wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int wm9712_soc_resume(struct snd_soc_codec *codec) { int i, ret; @@ -646,8 +640,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) /* set alc mux to none */ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); - wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; reset_err: @@ -664,11 +656,11 @@ static int wm9712_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .probe = wm9712_soc_probe, .remove = wm9712_soc_remove, - .suspend = wm9712_soc_suspend, .resume = wm9712_soc_resume, .read = ac97_read, .write = ac97_write, .set_bias_level = wm9712_set_bias_level, + .suspend_bias_off = true, .reg_cache_size = ARRAY_SIZE(wm9712_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, -- cgit v1.1 From 0cb6b1419ec864996835991a62788c588693f27d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 23 Nov 2014 13:37:48 +0100 Subject: ASoC: wm9713: Cleanup manual bias level transitions The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ac13fc8..e977b13 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1203,8 +1203,6 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (ret < 0) goto reset_err; - wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* unmute the adc - move to kcontrol */ reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); -- cgit v1.1 From bbc686b34650b0f54affe9d9a637ccbe02b03760 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Mon, 24 Nov 2014 20:37:12 +0200 Subject: ASoC: tlv320aic31xx: Fix off by one error in the loop stucture. Fix off by one read beyond the end of a table. Reported-by: David Binderman Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic31xx.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 145fe5b..93de5dd 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -911,12 +911,13 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, } aic31xx->p_div = i; - for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) { - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", - __func__, freq); - return -EINVAL; - } + for (i = 0; i < ARRAY_SIZE(aic31xx_divs) && + aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) + ; + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; } /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ -- cgit v1.1 From 8d213de7ffff614e3aa751a31a8980e1be3036e1 Mon Sep 17 00:00:00 2001 From: Andreas Ruprecht Date: Fri, 21 Nov 2014 20:50:46 +0100 Subject: ASoC: rockchip: i2s: Fix Kconfig for I2S device driver Currently, CONFIG_SND_SOC_ROCKCHIP_I2S could also be selected without having CONFIG_SND_SOC_ROCKCHIP enabled. As this makes no sense, a Kconfig dependency is added to CONFIG_SND_SOC_ROCKCHIP_I2S. This will make the item visible only if CONFIG_SND_SOC_ROCKCHIP is enabled. Additionally, as the code connected to CONFIG_SND_SOC_ROCKCHIP_I2S depends on CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM, the dependency is moved to reflect this more clearly. Signed-off-by: Andreas Ruprecht Reported-by: Jim Davis Signed-off-by: Mark Brown --- sound/soc/rockchip/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index b1fc0ca..e181826 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -1,7 +1,6 @@ config SND_SOC_ROCKCHIP tristate "ASoC support for Rockchip" depends on COMPILE_TEST || ARCH_ROCKCHIP - select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Rockchip SoCs' Audio interfaces. You will also need to @@ -9,7 +8,8 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate "Rockchip I2S Device Driver" - depends on CLKDEV_LOOKUP + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of -- cgit v1.1 From 141f87d4d6ab36bfcd4c5683cf90abf83b306d90 Mon Sep 17 00:00:00 2001 From: kbuild test robot Date: Tue, 25 Nov 2014 04:49:44 +0800 Subject: ASoC: sigmadsp: fix simple_return.cocci warnings sound/soc/codecs/sigmadsp.c:656:1-4: WARNING: end returns can be simpified and declaration on line 636 can be dropped Simplify a trivial if-return sequence. Possibly combine with a preceding function call. Generated by: scripts/coccinelle/misc/simple_return.cocci Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 6abefd2..34fdc40 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -633,7 +633,6 @@ static int sigmadsp_alloc_control(struct sigmadsp *sigmadsp, { struct snd_kcontrol_new template; struct snd_kcontrol *kcontrol; - int ret; memset(&template, 0, sizeof(template)); template.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -653,11 +652,7 @@ static int sigmadsp_alloc_control(struct sigmadsp *sigmadsp, kcontrol->private_free = sigmadsp_control_free; ctrl->kcontrol = kcontrol; - ret = snd_ctl_add(sigmadsp->component->card->snd_card, kcontrol); - if (ret) - return ret; - - return 0; + return snd_ctl_add(sigmadsp->component->card->snd_card, kcontrol); } static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, -- cgit v1.1 From 2d4e2d020516632288e8c8d1f8be2f3042d6b8de Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 18 Nov 2014 16:50:18 +0800 Subject: ASoC: rt5645: multiple JD mode support There are 3 JD modes in RT5645. This patch configure register values according to platform data. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 7 +++++++ 3 files changed, 42 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 937f421..120d961 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -26,6 +26,7 @@ struct rt5645_platform_data { /* true if codec's jd function is used */ bool en_jd_func; + unsigned int jd_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ef88b50..6e9cd8e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2239,7 +2239,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); } @@ -2543,6 +2544,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); } + if (rt5645->pdata.jd_mode) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_IRQ_JD_1_1_EN, RT5645_IRQ_JD_1_1_EN); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_JD_PSV_MODE, RT5645_JD_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_HPO_MIXER, + RT5645_IRQ_PSV_MODE, RT5645_IRQ_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_MIC2_OVCD_EN, RT5645_MIC2_OVCD_EN); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + switch (rt5645->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_2); + break; + default: + break; + } + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index c72220a..a815e36 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -594,6 +594,7 @@ #define RT5645_M_DAC1_HM_SFT 14 #define RT5645_M_HPVOL_HM (0x1 << 13) #define RT5645_M_HPVOL_HM_SFT 13 +#define RT5645_IRQ_PSV_MODE (0x1 << 12) /* SPK Left Mixer Control (0x46) */ #define RT5645_G_RM_L_SM_L_MASK (0x3 << 14) @@ -1350,6 +1351,10 @@ #define RT5645_PWR_CLK25M_PU (0x1 << 4) #define RT5645_IRQ_CLK_MCLK (0x0 << 3) #define RT5645_IRQ_CLK_INT (0x1 << 3) +#define RT5645_JD1_MODE_MASK (0x3 << 0) +#define RT5645_JD1_MODE_0 (0x0 << 0) +#define RT5645_JD1_MODE_1 (0x1 << 0) +#define RT5645_JD1_MODE_2 (0x2 << 0) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -1638,6 +1643,7 @@ #define RT5645_OT_P_SFT 10 #define RT5645_OT_P_NOR (0x0 << 10) #define RT5645_OT_P_INV (0x1 << 10) +#define RT5645_IRQ_JD_1_1_EN (0x1 << 9) /* IRQ Control 2 (0xbe) */ #define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) @@ -2120,6 +2126,7 @@ enum { #define RT5645_RXDP2_SEL_SFT (3) /* General Control3 (0xfc) */ +#define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) -- cgit v1.1 From e50334d4e1c3bacfeb3bb1530f73a419d4ec6832 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 17 Nov 2014 15:27:21 +0800 Subject: ASoC: rt5670: check if asrc is useable To use ASRC, the sysclk should be faster than 384 times sample rate of I2S1. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 5e54ac9..3ddb34e 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -576,6 +576,18 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int can_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->sysclk > rt5670->lrck[RT5670_AIF1] * 384) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, @@ -1639,8 +1651,8 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, - { "I2S1", NULL, "I2S1 ASRC" }, - { "I2S2", NULL, "I2S2 ASRC" }, + { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, + { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, { "DMIC1", NULL, "DMIC L1" }, { "DMIC1", NULL, "DMIC R1" }, -- cgit v1.1 From ff4541c3f48781f84e1cc162d73013aa32a09a41 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 17 Nov 2014 15:27:22 +0800 Subject: ASoC: rt5670: add DMIC ASRC support This patch will enable ASRC for DMIC if ASRC is useable. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 3ddb34e..8bf3a56 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1294,6 +1294,14 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5670_ASRC_1, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5670_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO2 ASRC", 1, RT5670_ASRC_1, + 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5670_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5670_ASRC_1, + 4, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5670_ASRC_1, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("ADC STO2 ASRC", 1, RT5670_ASRC_1, @@ -1650,6 +1658,10 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "DAC Mono Left Filter", NULL, "DAC MONO L ASRC", is_using_asrc }, { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC", can_use_asrc }, + { "Stereo2 DMIC Mux", NULL, "DMIC STO2 ASRC", can_use_asrc }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC", can_use_asrc }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC", can_use_asrc }, { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, -- cgit v1.1 From 6fe17da00ba7046db2d3a952a930e127dcd7f06e Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 25 Nov 2014 09:51:41 +0800 Subject: ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed After the patch "ASoC: rt5677: Use specific r/w function for DSP mode", the regmap_range "rt5677_ranges" was not registered in rt5677_regmap_physical, and it caused that the regmap_range "rt5677_ranges" cannot be accessed by the specific r/w function. The patch fixes this issue. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- cgit v1.1 From 86ea522b369abbbe92b4d66a238e79319ca46ba5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 24 Oct 2014 16:48:12 -0700 Subject: ASoC: fsl_esai: Use dynamic slot width as default The driver previously used 32-bit fixed slot width as default. In result, ESAI might use 32-bit length to capture 16-bit width audio slot from CODEC side when ESAI is running as DAI slave. So this patch just removes the default slot_width so as to use dynamic slot width. If there comes a specific situation that needs a fixed width, the machine driver shall set slot_width via set_tdm_slot() so as to let the ESAI driver replace the dynamic width policy with the fixed value. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_esai.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a645e29..ca319d5 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -513,10 +513,15 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 slot_width = width; u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; + /* Override slot_width if being specifially set */ + if (esai_priv->slot_width) + slot_width = esai_priv->slot_width; + + bclk = params_rate(params) * slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -538,7 +543,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); @@ -780,9 +785,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - /* Set a default slot size */ - esai_priv->slot_width = 32; - /* Set a default slot number */ esai_priv->slots = 2; -- cgit v1.1 From cb3fc1ff46667ee4ab23ef22c9e391aa7d014cdf Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 24 Oct 2014 16:48:13 -0700 Subject: ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai ESAI may need to use fixed slot width to comply with external CODEC. So this set_tdm_slot() call will allow the ESAI driver to override its default dynamic slot width policy. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 14572e6..3f6959c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -51,6 +51,7 @@ struct codec_priv { * @sysclk_freq[2]: SYSCLK rates for set_sysclk() * @sysclk_dir[2]: SYSCLK directions for set_sysclk() * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * @slot_width: Slot width of each frame * * Note: [1] for tx and [0] for rx */ @@ -58,6 +59,7 @@ struct cpu_priv { unsigned long sysclk_freq[2]; u32 sysclk_dir[2]; u32 sysclk_id[2]; + u32 slot_width; }; /** @@ -142,6 +144,15 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return ret; } + if (cpu_priv->slot_width) { + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, + cpu_priv->slot_width); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + } + return 0; } @@ -453,6 +464,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; -- cgit v1.1 From f1e5982546edf96e4537ba689cfad83b90b70143 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 25 Nov 2014 21:00:53 +0800 Subject: ASoC: Intel: Fix stream volume set no effect issue on Broadwell The volume setting control for capture stream doesn't take effect on intel Broadwell platform. Root cause it at 2 points: 1. set stream volume with channel=2 is wrongly bapassed; 2. the saved stream volume should be restored after stream is commit. Here correct these 2 items to fix the stream volume set issue. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-ipc.c | 35 ++++++++++++++++++++++++++--------- sound/soc/intel/sst-haswell-ipc.h | 1 + sound/soc/intel/sst-haswell-pcm.c | 21 ++++++++++++--------- 3 files changed, 39 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ffd5728..3f8c482 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1042,14 +1042,9 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, trace_ipc_request("set stream volume", stream->reply.stream_hw_id); - if (channel > 1) + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) return -EINVAL; - if (stream->mute[channel]) { - stream->mute_volume[channel] = volume; - return 0; - } - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); @@ -1057,9 +1052,28 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, header |= (stage_id << IPC_STG_ID_SHIFT); req = &stream->vol_req; - req->channel = channel; req->target_volume = volume; + /* set both at same time ? */ + if (channel == SST_HSW_CHANNELS_ALL) { + if (hsw->mute[0] && hsw->mute[1]) { + hsw->mute_volume[0] = hsw->mute_volume[1] = volume; + return 0; + } else if (hsw->mute[0]) + req->channel = 1; + else if (hsw->mute[1]) + req->channel = 0; + else + req->channel = SST_HSW_CHANNELS_ALL; + } else { + /* set only 1 channel */ + if (hsw->mute[channel]) { + hsw->mute_volume[channel] = volume; + return 0; + } + req->channel = channel; + } + ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: set stream volume failed\n"); @@ -1138,8 +1152,11 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, trace_ipc_request("set mixer volume", volume); + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) + return -EINVAL; + /* set both at same time ? */ - if (channel == 2) { + if (channel == SST_HSW_CHANNELS_ALL) { if (hsw->mute[0] && hsw->mute[1]) { hsw->mute_volume[0] = hsw->mute_volume[1] = volume; return 0; @@ -1148,7 +1165,7 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, else if (hsw->mute[1]) req.channel = 0; else - req.channel = 0xffffffff; + req.channel = SST_HSW_CHANNELS_ALL; } else { /* set only 1 channel */ if (hsw->mute[channel]) { diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 387511f..138e894 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -24,6 +24,7 @@ #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) +#define SST_HSW_CHANNELS_ALL 0xffffffff #define SST_HSW_FW_LOG_CONFIG_DWORDS 12 #define SST_HSW_GLOBAL_LOG 15 diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index e7a3b6a..13c0100 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -189,7 +189,8 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume); + /* apply volume value to all channels */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume); @@ -255,7 +256,7 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_mixer_set_volume(hsw, 0, 2, volume); + sst_hsw_mixer_set_volume(hsw, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); @@ -525,7 +526,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, dev_err(rtd->dev, "error: failed to commit stream %d\n", ret); return ret; } - pcm_data->allocated = true; + + if (!pcm_data->allocated) { + /* Set previous saved volume */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 0, pcm_data->volume[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 1, pcm_data->volume[1]); + pcm_data->allocated = true; + } ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1); if (ret < 0) @@ -632,12 +641,6 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) return -EINVAL; } - /* Set previous saved volume */ - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 0, pcm_data->volume[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 1, pcm_data->volume[1]); - mutex_unlock(&pcm_data->mutex); return 0; } -- cgit v1.1 From 93b0f3eeebdce6f32417187b5d24ea218a3e7fb4 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 25 Nov 2014 13:16:12 +0100 Subject: ASoC: core: add multi-codec support in DT This patch exports a core function which handles the DT description of multi-codec links (as: "sound-dai = <&hdmi 0>, <&spdif_codec>;") and creates a CODEC component array in the DAI link. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- include/sound/soc.h | 3 ++ sound/soc/soc-core.c | 110 ++++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 98 insertions(+), 15 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..2750e6a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1451,6 +1451,9 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, struct device_node **framemaster); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); +int snd_soc_of_get_dai_link_codecs(struct device *dev, + struct device_node *of_node, + struct snd_soc_dai_link *dai_link); #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a2..72a3ad0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4750,36 +4750,30 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); -int snd_soc_of_get_dai_name(struct device_node *of_node, - const char **dai_name) +static int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name) { struct snd_soc_component *pos; - struct of_phandle_args args; - int ret; - - ret = of_parse_phandle_with_args(of_node, "sound-dai", - "#sound-dai-cells", 0, &args); - if (ret) - return ret; - - ret = -EPROBE_DEFER; + int ret = -EPROBE_DEFER; mutex_lock(&client_mutex); list_for_each_entry(pos, &component_list, list) { - if (pos->dev->of_node != args.np) + if (pos->dev->of_node != args->np) continue; if (pos->driver->of_xlate_dai_name) { - ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); + ret = pos->driver->of_xlate_dai_name(pos, + args, + dai_name); } else { int id = -1; - switch (args.args_count) { + switch (args->args_count) { case 0: id = 0; /* same as dai_drv[0] */ break; case 1: - id = args.args[0]; + id = args->args[0]; break; default: /* not supported */ @@ -4801,6 +4795,21 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, break; } mutex_unlock(&client_mutex); + return ret; +} + +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name) +{ + struct of_phandle_args args; + int ret; + + ret = of_parse_phandle_with_args(of_node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + ret = snd_soc_get_dai_name(&args, dai_name); of_node_put(args.np); @@ -4808,6 +4817,77 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, } EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); +/* + * snd_soc_of_get_dai_link_codecs - Parse a list of CODECs in the devicetree + * @dev: Card device + * @of_node: Device node + * @dai_link: DAI link + * + * Builds an array of CODEC DAI components from the DAI link property + * 'sound-dai'. + * The array is set in the DAI link and the number of DAIs is set accordingly. + * The device nodes in the array (of_node) must be dereferenced by the caller. + * + * Returns 0 for success + */ +int snd_soc_of_get_dai_link_codecs(struct device *dev, + struct device_node *of_node, + struct snd_soc_dai_link *dai_link) +{ + struct of_phandle_args args; + struct snd_soc_dai_link_component *component; + char *name; + int index, num_codecs, ret; + + /* Count the number of CODECs */ + name = "sound-dai"; + num_codecs = of_count_phandle_with_args(of_node, name, + "#sound-dai-cells"); + if (num_codecs <= 0) { + if (num_codecs == -ENOENT) + dev_err(dev, "No 'sound-dai' property\n"); + else + dev_err(dev, "Bad phandle in 'sound-dai'\n"); + return num_codecs; + } + component = devm_kzalloc(dev, + sizeof *component * num_codecs, + GFP_KERNEL); + if (!component) + return -ENOMEM; + dai_link->codecs = component; + dai_link->num_codecs = num_codecs; + + /* Parse the list */ + for (index = 0, component = dai_link->codecs; + index < dai_link->num_codecs; + index++, component++) { + ret = of_parse_phandle_with_args(of_node, name, + "#sound-dai-cells", + index, &args); + if (ret) + goto err; + component->of_node = args.np; + ret = snd_soc_get_dai_name(&args, &component->dai_name); + if (ret < 0) + goto err; + } + return 0; +err: + for (index = 0, component = dai_link->codecs; + index < dai_link->num_codecs; + index++, component++) { + if (!component->of_node) + break; + of_node_put(component->of_node); + component->of_node = NULL; + } + dai_link->codecs = NULL; + dai_link->num_codecs = 0; + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_link_codecs); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS -- cgit v1.1 From 7195d920bd6094f6810e20a903027cd438bc6d8e Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 25 Nov 2014 13:22:45 +0100 Subject: ASoC: simple-card: Remove useless function argument The device node pointer 'cpu' is not used in the function asoc_simple_card_parse_daifmt(). Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3e3fec4..ece22d5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -232,7 +232,6 @@ asoc_simple_card_sub_parse_of(struct device_node *np, static int asoc_simple_card_parse_daifmt(struct device_node *node, struct simple_card_data *priv, - struct device_node *cpu, struct device_node *codec, char *prefix, int idx) { @@ -309,7 +308,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, } ret = asoc_simple_card_parse_daifmt(node, priv, - cpu, codec, prefix, idx); + codec, prefix, idx); if (ret < 0) goto dai_link_of_err; -- cgit v1.1 From 69b93607a9862b1db2f0f2e078e1396f8e20fa9b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 25 Nov 2014 20:29:40 +0100 Subject: ASoC: qi_lb60: Pass flags to gpiod_get() Pass flags to gpiod_get() to automatically configure the GPIOs. This is shorter and not passing any flags to gpiod_get() will eventually no longer be supported. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 5cb91f9..0fb7d2a 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -77,25 +77,18 @@ static int qi_lb60_probe(struct platform_device *pdev) { struct qi_lb60 *qi_lb60; struct snd_soc_card *card = &qi_lb60_card; - int ret; qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL); if (!qi_lb60) return -ENOMEM; - qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd"); + qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->snd_gpio)) return PTR_ERR(qi_lb60->snd_gpio); - ret = gpiod_direction_output(qi_lb60->snd_gpio, 0); - if (ret) - return ret; - qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp"); + qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->amp_gpio)) return PTR_ERR(qi_lb60->amp_gpio); - ret = gpiod_direction_output(qi_lb60->amp_gpio, 0); - if (ret) - return ret; card->dev = &pdev->dev; -- cgit v1.1 From e874bf5f7647a9fdf14d72dbb376ec95327e3a81 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 25 Nov 2014 21:41:03 +0100 Subject: ASoC: Disable regmap helpers if regmap is disabled If regmap is disabled there will be no users of the ASoC regmap helpers. Furthermore regmap_exit() will no be defined causing the following compile error: sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap': sound/soc/soc-core.c:2645:2: error: implicit declaration of function 'regmap_exit' [-Werror=implicit-function-declaration] So disable the helpers if regmap is disabled. Reported-by: kbuild test robot Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup") Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/soc-core.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 80ca937..b532348 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1286,6 +1286,8 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +#ifdef CONFIG_REGMAP + void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); void snd_soc_component_exit_regmap(struct snd_soc_component *component); @@ -1321,6 +1323,8 @@ static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) snd_soc_component_exit_regmap(&codec->component); } +#endif + /* device driver data */ static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index db74c06..1edc519 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3661,6 +3661,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component) component->val_bytes = val_bytes; } +#ifdef CONFIG_REGMAP + /** * snd_soc_component_init_regmap() - Initialize regmap instance for the component * @component: The component for which to initialize the regmap instance @@ -3696,6 +3698,8 @@ void snd_soc_component_exit_regmap(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); +#endif + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { if (!component->write && !component->read) { -- cgit v1.1 From c362effe5cda4df02aa7670d58636ea73979e304 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 25 Nov 2014 12:14:48 +0100 Subject: ASoC: Remove 'const' from the device_node pointers As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." [This isn't quite the application here, we're not geting or putting the pointer though we did add some other users who call non-const OF functions -- broonie] Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- include/sound/soc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index edbb07b..d1f3dac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -881,7 +881,7 @@ struct snd_soc_platform_driver { struct snd_soc_dai_link_component { const char *name; - const struct device_node *of_node; + struct device_node *of_node; const char *dai_name; }; @@ -983,7 +983,7 @@ struct snd_soc_codec_conf { * DT/OF node, but not both. */ const char *dev_name; - const struct device_node *of_node; + struct device_node *of_node; /* * optional map of kcontrol, widget and path name prefixes that are @@ -1000,7 +1000,7 @@ struct snd_soc_aux_dev { * DT/OF node, but not both. */ const char *codec_name; - const struct device_node *codec_of_node; + struct device_node *codec_of_node; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_component *component); -- cgit v1.1 From d683d0b690c13437d752ccce47963ac64119b07a Mon Sep 17 00:00:00 2001 From: Krishna Mohan Dani Date: Wed, 26 Nov 2014 14:53:04 +0530 Subject: ASoC: Samsung: Add arndale_rt5631 machine driver and binding Adding machine driver to instantiate I2S based realtek's ALC5631 sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec hence support for ALC5631 based machine driver is being added. This patch also documents the device tree binding for the Arndale board based machine driver. Signed-off-by: Claude Youn Signed-off-by: Krishna Mohan Dani Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/arndale.txt | 24 ++++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 150 +++++++++++++++++++++ 4 files changed, 182 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e0e737f..fc67f97 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -239,3 +239,9 @@ config SND_SOC_ODROIDX2 select SND_SAMSUNG_I2S help Say Y here to enable audio support for the Odroid-X2/U3. + +config SND_SOC_ARNDALE_RT5631_ALC5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on SND_SOC_SAMSUNG + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 91505dd..31e3dba 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -45,6 +45,7 @@ snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o +snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o +obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c new file mode 100644 index 0000000..1e2b61c --- /dev/null +++ b/sound/soc/samsung/arndale_rt5631.c @@ -0,0 +1,150 @@ +/* + * arndale_rt5631.c + * + * Copyright (c) 2014, Insignal Co., Ltd. + * + * Author: Claude + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include "i2s.h" + +static int arndale_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int rfs, ret; + unsigned long rclk; + + rfs = 256; + + rclk = params_rate(params) * rfs; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, + 0, SND_SOC_CLOCK_OUT); + + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops arndale_ops = { + .hw_params = arndale_hw_params, +}; + +static struct snd_soc_dai_link arndale_rt5631_dai[] = { + { + .name = "RT5631 HiFi", + .stream_name = "Primary", + .codec_dai_name = "rt5631-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ops = &arndale_ops, + }, +}; + +static struct snd_soc_card arndale_rt5631 = { + .name = "Arndale RT5631", + .dai_link = arndale_rt5631_dai, + .num_links = ARRAY_SIZE(arndale_rt5631_dai), +}; + +static int arndale_audio_probe(struct platform_device *pdev) +{ + int n, ret; + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &arndale_rt5631; + + card->dev = &pdev->dev; + + for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { + if (!arndale_rt5631_dai[n].cpu_dai_name) { + arndale_rt5631_dai[n].cpu_of_node = of_parse_phandle(np, + "samsung,audio-cpu", n); + + if (!arndale_rt5631_dai[n].cpu_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } + } + if (!arndale_rt5631_dai[n].platform_name) + arndale_rt5631_dai[n].platform_of_node = + arndale_rt5631_dai[n].cpu_of_node; + + arndale_rt5631_dai[n].codec_name = NULL; + arndale_rt5631_dai[n].codec_of_node = of_parse_phandle(np, + "samsung,audio-codec", n); + if (!arndale_rt5631_dai[0].codec_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-codec' missing or invalid\n"); + return -EINVAL; + } + } + + ret = devm_snd_soc_register_card(card->dev, card); + + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); + + return ret; +} + +static int arndale_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { + { .compatible = "samsung,arndale-rt5631", }, + { .compatible = "samsung,arndale-alc5631", }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match); + +static struct platform_driver arndale_audio_driver = { + .driver = { + .name = "arndale-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), + }, + .probe = arndale_audio_probe, + .remove = arndale_audio_remove, +}; + +module_platform_driver(arndale_audio_driver); + +MODULE_AUTHOR("Claude "); +MODULE_DESCRIPTION("ALSA SoC Driver for Arndale Board"); +MODULE_LICENSE("GPL"); -- cgit v1.1 From 69eba10e606a80665f8573221fec589430d9d1cb Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 27 Nov 2014 01:34:43 +0300 Subject: ALSA: hda - using uninitialized data In olden times the snd_hda_param_read() function always set "*start_id" but in 2007 we introduced a new return and it causes uninitialized data bugs in a couple of the callers: print_codec_info() and hdmi_parse_codec(). Fixes: e8a7f136f5ed ('[ALSA] hda-intel - Improve HD-audio codec probing robustness') Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b2d5899..2fe86d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -346,8 +346,10 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); - if (parm == -1) + if (parm == -1) { + *start_id = 0; return 0; + } *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); } -- cgit v1.1 From 3ad5e861a715cbe932cd145d4612c11e5912a72f Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Thu, 27 Nov 2014 16:53:08 +0800 Subject: ASoC: wm8960: Move register initialisation to I2C driver probe() We must ensure that the clocking configuration is valid as rapidly as possible. And do software reset before the others registers updates, or the registers will be reset to the default state. Signed-off-by: Zidan Wang Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 41 ++++++++++++++++++++--------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index bc8793cd..031a1ae 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -128,7 +128,7 @@ struct wm8960_priv { struct wm8960_data pdata; }; -#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", @@ -947,31 +947,12 @@ static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = &wm8960->pdata; - int ret; if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; else wm8960->set_bias_level = wm8960_set_bias_level_out3; - ret = wm8960_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - - /* Latch the update bits */ - snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); - snd_soc_add_codec_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); @@ -1030,7 +1011,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); - if (pdata && pdata->shared_lrclk) { + ret = wm8960_reset(wm8960->regmap); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + return ret; + } + + if (wm8960->pdata.shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); if (ret != 0) { @@ -1040,6 +1027,18 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, } } + /* Latch the update bits */ + regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT2, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT2, 0x100, 0x100); + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.1 From d98123a76be53d570d72e04aac3e195a560ef149 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 27 Nov 2014 01:35:16 +0300 Subject: ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl() The "changed" variable should be set to false at the start. Fixes: a35daac77a03 ('ASoC: sigmadsp: Add support for fw v2') Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/sigmadsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 34fdc40..d53680a 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -661,7 +661,8 @@ static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, struct snd_card *card = sigmadsp->component->card->snd_card; struct snd_kcontrol_volatile *vd; struct snd_ctl_elem_id id; - bool active, changed; + bool active; + bool changed = false; active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask); -- cgit v1.1 From 525b8634d8dee0e3a8409a73ef5f22ac8676a8c4 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Thu, 27 Nov 2014 14:04:23 +0800 Subject: ASoC: Intel: Remove useless loopback volume control for Broadwell On Broadwell, the ADSP FW don't support loopback record volume tuning, so here remove this control. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 13c0100..7aa348d 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -308,10 +308,6 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, ARRAY_SIZE(volume_map), 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), - /* Loopback volume */ - SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8, - ARRAY_SIZE(volume_map), 0, - hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map), 0, -- cgit v1.1 From 002fe7c831404d179266cfe0dad00a67333256f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:50 +0100 Subject: ASoC: cq93vc: Remove unused state struct While two of the fields in the cq93vc driver state struct are initialized none of them are ever acutally read again. So remove the whole struct. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/linux/mfd/davinci_voicecodec.h | 7 ------- sound/soc/codecs/cq93vc.c | 8 -------- 2 files changed, 15 deletions(-) diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h index cb01496..8e1cdbe 100644 --- a/include/linux/mfd/davinci_voicecodec.h +++ b/include/linux/mfd/davinci_voicecodec.h @@ -99,12 +99,6 @@ struct davinci_vcif { dma_addr_t dma_rx_addr; }; -struct cq93vc { - struct platform_device *pdev; - struct snd_soc_codec *codec; - u32 sysclk; -}; - struct davinci_vc; struct davinci_vc { @@ -122,7 +116,6 @@ struct davinci_vc { /* Client devices */ struct davinci_vcif davinci_vcif; - struct cq93vc cq93vc; }; #endif diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 537327c..036a877 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -62,14 +62,10 @@ static int cq93vc_mute(struct snd_soc_dai *dai, int mute) static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct davinci_vc *davinci_vc = codec->dev->platform_data; - switch (freq) { case 22579200: case 27000000: case 33868800: - davinci_vc->cq93vc.sysclk = freq; return 0; } @@ -135,10 +131,6 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; - - davinci_vc->cq93vc.codec = codec; - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.1 From 6200b75a8bbc16e434bf3d8ca54538ea678ccbd7 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:51 +0100 Subject: ASoC: cq93vc: Cleanup manual bias level transitions Remove the manual transition back to SND_SOC_BIAS_STANDBY in resume. This is already be automatically handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. While we are at it also remove the unused codec field from the cq93vc struct so the whole probe function can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 25 ------------------------- 1 file changed, 25 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 036a877..8d638e8 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -122,28 +122,6 @@ static struct snd_soc_dai_driver cq93vc_dai = { .ops = &cq93vc_dai_ops, }; -static int cq93vc_resume(struct snd_soc_codec *codec) -{ - cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int cq93vc_probe(struct snd_soc_codec *codec) -{ - /* Off, with power on */ - cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - -static int cq93vc_remove(struct snd_soc_codec *codec) -{ - cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct regmap *cq93vc_get_regmap(struct device *dev) { struct davinci_vc *davinci_vc = dev->platform_data; @@ -153,9 +131,6 @@ static struct regmap *cq93vc_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .set_bias_level = cq93vc_set_bias_level, - .probe = cq93vc_probe, - .remove = cq93vc_remove, - .resume = cq93vc_resume, .get_regmap = cq93vc_get_regmap, .controls = cq93vc_snd_controls, .num_controls = ARRAY_SIZE(cq93vc_snd_controls), -- cgit v1.1 From 68d27bc63c4f331c912dfb92168f5fe4753c61c9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:52 +0100 Subject: ASoC: lm49453: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c1ae576..c4dfde9 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1395,15 +1395,7 @@ static struct snd_soc_dai_driver lm49453_dai[] = { }, }; -/* power down chip */ -static int lm49453_remove(struct snd_soc_codec *codec) -{ - lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { - .remove = lm49453_remove, .set_bias_level = lm49453_set_bias_level, .controls = lm49453_snd_controls, .num_controls = ARRAY_SIZE(lm49453_snd_controls), -- cgit v1.1 From 0eef4ed5970a736bf2449b389fb44f6fe3635765 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:53 +0100 Subject: ASoC: sn95031: Cleanup bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 6167c59..31d97cd 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -870,17 +870,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) return 0; } -static int sn95031_codec_remove(struct snd_soc_codec *codec) -{ - pr_debug("codec_remove called\n"); - sn95031_set_vaud_bias(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, - .remove = sn95031_codec_remove, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, -- cgit v1.1 From aabb87f00304764dffe097e3b65f6a1862c2c2b5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:54 +0100 Subject: ASoC: tlv320aic23: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d671679..cc17e7e 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -540,19 +540,11 @@ static struct snd_soc_dai_driver tlv320aic23_dai = { .ops = &tlv320aic23_dai_ops, }; -static int tlv320aic23_suspend(struct snd_soc_codec *codec) -{ - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - static int tlv320aic23_resume(struct snd_soc_codec *codec) { struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); regcache_mark_dirty(aic23->regmap); regcache_sync(aic23->regmap); - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } @@ -562,9 +554,6 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); - /* power on device */ - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); /* Unmute input */ @@ -589,18 +578,12 @@ static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) return 0; } -static int tlv320aic23_remove(struct snd_soc_codec *codec) -{ - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .probe = tlv320aic23_codec_probe, - .remove = tlv320aic23_remove, - .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, + .suspend_bias_off = true, + .controls = tlv320aic23_snd_controls, .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls), .dapm_widgets = tlv320aic23_dapm_widgets, -- cgit v1.1 From a43a262901363ea412c288e5ebc3a3c0a8ff6591 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:55 +0100 Subject: ASoC: tlv320aix31xx: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Acked-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 145fe5b..6cd5f50 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1056,18 +1056,6 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int aic31xx_suspend(struct snd_soc_codec *codec) -{ - aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int aic31xx_resume(struct snd_soc_codec *codec) -{ - aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int aic31xx_codec_probe(struct snd_soc_codec *codec) { int ret = 0; @@ -1110,8 +1098,6 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); int i; - /* power down chip */ - aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) regulator_unregister_notifier(aic31xx->supplies[i].consumer, @@ -1123,9 +1109,9 @@ static int aic31xx_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, .remove = aic31xx_codec_remove, - .suspend = aic31xx_suspend, - .resume = aic31xx_resume, .set_bias_level = aic31xx_set_bias_level, + .suspend_bias_off = true, + .controls = aic31xx_snd_controls, .num_controls = ARRAY_SIZE(aic31xx_snd_controls), .dapm_widgets = aic31xx_dapm_widgets, -- cgit v1.1 From f10c0a71e6efc7c8cbc3bfcfd0ecf822607f0b3d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:56 +0100 Subject: ASoC: tlv320aic32x4: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 6ea662d..015467e 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -597,18 +597,6 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_suspend(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int aic32x4_resume(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); @@ -654,8 +642,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K); - aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* * Workaround: for an unknown reason, the ADC needs to be powered up * and down for the first capture to work properly. It seems related to @@ -669,18 +655,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) return 0; } -static int aic32x4_remove(struct snd_soc_codec *codec) -{ - aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .probe = aic32x4_probe, - .remove = aic32x4_remove, - .suspend = aic32x4_suspend, - .resume = aic32x4_resume, .set_bias_level = aic32x4_set_bias_level, + .suspend_bias_off = true, .controls = aic32x4_snd_controls, .num_controls = ARRAY_SIZE(aic32x4_snd_controls), -- cgit v1.1 From 68f438378cde79e29f71c7e043b10d76001d8892 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:57 +0100 Subject: ASoC: tlv320aic3x: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f0a8281..b7ebce0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1491,7 +1491,6 @@ static int aic3x_remove(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int i; - aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, -- cgit v1.1 From 90db15e17e46e2841843bf20f258fed963228bed Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:58 +0100 Subject: ASoC: tlv320dac33: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index e21ed93..0fe2ced 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1436,8 +1436,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (dac33->irq >= 0) { free_irq(dac33->irq, dac33->codec); destroy_workqueue(dac33->dac33_wq); -- cgit v1.1 From 3ec8d2036464d961a6314281ae68d80a5c071d07 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:57:59 +0100 Subject: ASoC: twl4030: Cleanup manual bias level transitions Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b6b0cb3..27f3b21 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2177,8 +2177,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct twl4030_codec_data *pdata = twl4030->pdata; - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (pdata && pdata->hs_extmute && gpio_is_valid(pdata->hs_extmute_gpio)) gpio_free(pdata->hs_extmute_gpio); -- cgit v1.1 From c4ee42a050e82855aa06d7217937b1549c95bef3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 26 Nov 2014 20:58:00 +0100 Subject: ASoC: twl6040: Cleanup manual bias level transitions Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 23 +---------------------- 1 file changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0f6067f..5ff2b1e 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1095,25 +1095,6 @@ static struct snd_soc_dai_driver twl6040_dai[] = { }, }; -#ifdef CONFIG_PM -static int twl6040_suspend(struct snd_soc_codec *codec) -{ - twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int twl6040_resume(struct snd_soc_codec *codec) -{ - twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define twl6040_suspend NULL -#define twl6040_resume NULL -#endif - static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; @@ -1160,7 +1141,6 @@ static int twl6040_remove(struct snd_soc_codec *codec) struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); free_irq(priv->plug_irq, codec); - twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -1168,11 +1148,10 @@ static int twl6040_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .probe = twl6040_probe, .remove = twl6040_remove, - .suspend = twl6040_suspend, - .resume = twl6040_resume, .read = twl6040_read, .write = twl6040_write, .set_bias_level = twl6040_set_bias_level, + .suspend_bias_off = true, .ignore_pmdown_time = true, .controls = twl6040_snd_controls, -- cgit v1.1 From d819ce965d451aac08e46c9f8e2119fe3a845786 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 27 Nov 2014 13:02:00 -0200 Subject: ASoC: sgtl5000: Remove MCLK restriction According to the sgtl5000 datasheet the MCLK frequency range restriction of 8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode. When running the codec as slave, the master should generate MCLK in the range of 256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK input mode. In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the current check would return -EINVAL, which is not correct. Remove the 8 to 27MHz frequency check, since this only applies to the synchronous SYS_MCLK input case. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 490404c..4735791 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1435,7 +1435,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, { struct sgtl5000_priv *sgtl5000; int ret, reg, rev; - unsigned int mclk; struct device_node *np = client->dev.of_node; u32 value; @@ -1460,14 +1459,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } - /* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */ - mclk = clk_get_rate(sgtl5000->mclk); - if (mclk < 8000000 || mclk > 27000000) { - dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n", - mclk / 1000000, mclk / 1000 % 1000); - return -EINVAL; - } - ret = clk_prepare_enable(sgtl5000->mclk); if (ret) return ret; -- cgit v1.1 From 2a4cfd10229dc93507aa5ddbc1ba0162140f4951 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 27 Nov 2014 13:02:01 -0200 Subject: ASoC: sgtl5000: Allow 8kHz playback in codec slave mode When trying to play a 8kHz file with codec in slave mode we get the following error on a mx28evk: $ aplay -Dhw:0,0 stereo_8k.wav Playing WAVE 'stereo_8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo [ 21.218647] sgtl5000 0-000a: PLL not supported in slave mode [ 21.224559] sgtl5000 0-000a: 128 ratio is not supported. SYS_MCLK needs to be 256, 384 or 512 * fs [ 21.233687] sgtl5000 0-000a: ASoC: can't set sgtl5000 hw params: -22 aplay: set_params:1123: Unable to install hw params: This error happens because we are using 'sys_fs' instead of 'frame_rate' in the valid ratio check. Use the real'frame_rate' so that the ratio is correctly calculated and the playback can run. sgtl5000 codec manual states that in 'Synchronous SYS_MCLK input' mode that the following SYS_CLK frequencies are allowed: 256*fs, 384*fs, 512*fs. , where fs is the sampling frequency, which can be in the range of: 8, 11.025, 16, 22.5, 32, 44.1, 48, 96 kHz. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 4735791..47d6ca0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -618,7 +618,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) * factor of freq = 96 kHz can only be 256, since mclk is in the range * of 8 MHz - 27 MHz */ - switch (sgtl5000->sysclk / sys_fs) { + switch (sgtl5000->sysclk / frame_rate) { case 256: clk_ctl |= SGTL5000_MCLK_FREQ_256FS << SGTL5000_MCLK_FREQ_SHIFT; @@ -641,7 +641,7 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) "PLL not supported in slave mode\n"); dev_err(codec->dev, "%d ratio is not supported. " "SYS_MCLK needs to be 256, 384 or 512 * fs\n", - sgtl5000->sysclk / sys_fs); + sgtl5000->sysclk / frame_rate); return -EINVAL; } } -- cgit v1.1 From d206f66177ab7cd69d79c7e01b43f45d935f43dd Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 27 Nov 2014 13:01:59 -0200 Subject: ASoC: mxs-sgtl5000: Remove MCLK restriction According to the sgtl5000 datasheet the MCLK frequency range restriction of 8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode. mxs-sgtl5000 machine sets the codec as slave, and mx28 generates MCLK in the range of 256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK input. In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the current check would return -EINVAL, which is not correct. Remove the 8 to 27MHz frequency check, since this only applies to the synchronous SYS_MCLK input case. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-sgtl5000.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 61822cc..3bba6cf 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -49,13 +49,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, break; } - /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ - if (mclk < 8000000 || mclk > 27000000) { - dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", - mclk / 1000000, mclk / 1000 % 1000); - return -EINVAL; - } - /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); if (ret) { -- cgit v1.1 From 37e661ee10c6d0d1310c62b3d29ae9a63073ac5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Nov 2014 11:28:07 +0100 Subject: ALSA: hda - Add AZX_DCAPS_SNOOP_OFF (and refactor snoop setup) Add a new driver_caps bit, AZX_DCAPS_SNOOP_OFF, to set the snoop off as default. This new bit is used for the checks in azx_check_snoop_available(). Most of case-switches are replaced with the new dcaps in each entry. While working on it, for avoiding to spend more bits, combine three bits AZX_DCAPS_SNOOP_SCH, AZX_DCAPS_SNOOP_ATI and AZX_DCAPS_SNOOP_NVIDIA bits into a flat type of two bits. This reduces the bits usages, and assign AZX_DCAPS_OFF to this empty bit now. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 71 +++++++++++++++++++++++++---------------------- sound/pci/hda/hda_priv.h | 12 ++++++-- 2 files changed, 47 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 966e6f9..633020d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -272,42 +272,51 @@ enum { AZX_NUM_DRIVERS, /* keep this as last entry */ }; +#define azx_get_snoop_type(chip) \ + (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) +#define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_REVERSE_ASSIGN) + (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) #define AZX_DCAPS_INTEL_HASWELL \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME | \ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ - AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_PM_RUNTIME | \ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \ - AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) +/* quirks for ATI HDMI with snoop off */ +#define AZX_DCAPS_PRESET_ATI_HDMI_NS \ + (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\ - AZX_DCAPS_CORBRP_SELF_CLEAR) + (AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\ + AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\ + AZX_DCAPS_SNOOP_TYPE(NVIDIA)) #define AZX_DCAPS_PRESET_CTHDA \ - (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) + (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) /* * VGA-switcher support @@ -436,6 +445,8 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, static void azx_init_pci(struct azx *chip) { + int snoop_type = azx_get_snoop_type(chip); + /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS * Ensuring these bits are 0 clears playback static on some HD Audio @@ -450,7 +461,7 @@ static void azx_init_pci(struct azx *chip) /* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio, * we need to enable snoop. */ - if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_ATI) { dev_dbg(chip->card->dev, "Setting ATI snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -459,7 +470,7 @@ static void azx_init_pci(struct azx *chip) } /* For NVIDIA HDA, enable snoop */ - if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_NVIDIA) { dev_dbg(chip->card->dev, "Setting Nvidia snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -474,7 +485,7 @@ static void azx_init_pci(struct azx *chip) } /* Enable SCH/PCH snoop if needed */ - if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_SCH) { unsigned short snoop; pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) || @@ -1361,8 +1372,8 @@ static void azx_check_snoop_available(struct azx *chip) { bool snoop = chip->snoop; - switch (chip->driver_type) { - case AZX_DRIVER_VIA: + if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE && + chip->driver_type == AZX_DRIVER_VIA) { /* force to non-snoop mode for a new VIA controller * when BIOS is set */ @@ -1372,17 +1383,11 @@ static void azx_check_snoop_available(struct azx *chip) if (!(val & 0x80) && chip->pci->revision == 0x30) snoop = false; } - break; - case AZX_DRIVER_ATIHDMI_NS: - /* new ATI HDMI requires non-snoop */ - snoop = false; - break; - case AZX_DRIVER_CTHDA: - case AZX_DRIVER_CMEDIA: - snoop = false; - break; } + if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF) + snoop = false; + if (snoop != chip->snoop) { dev_info(chip->card->dev, "Force to %s mode\n", snoop ? "snoop" : "non-snoop"); @@ -2116,13 +2121,13 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaa98), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x9902), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, @@ -2169,7 +2174,7 @@ static const struct pci_device_id azx_ids[] = { /* CM8888 */ { PCI_DEVICE(0x13f6, 0x5011), .driver_data = AZX_DRIVER_CMEDIA | - AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_SNOOP_OFF }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index 949cd43..602536c 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -152,9 +152,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* bits 0-7 are used for indicating driver type */ #define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ #define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_ATI_SNOOP (1 << 10) /* ATI snoop enable */ -#define AZX_DCAPS_NVIDIA_SNOOP (1 << 11) /* Nvidia snoop enable */ -#define AZX_DCAPS_SCH_SNOOP (1 << 12) /* SCH/PCH snoop enable */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ #define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ #define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ @@ -172,6 +171,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +enum { + AZX_SNOOP_TYPE_NONE , + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + /* HD Audio class code */ #define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 -- cgit v1.1 From 7c7320157a37ed459b59e2f6b53b73780b12ad80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 25 Nov 2014 12:54:16 +0100 Subject: ALSA: hda - Allow forcibly enabling/disabling snoop User can pass snoop option to enable/disable the snoop behavior, but currently azx_check_snoop_available() always turns it off for some devices. For better debuggability, change the parameter as bint, and allow user to enable/disable forcibly the snoop when specified via the module option. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++--------------- 1 file changed, 18 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 633020d..728663d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -196,8 +196,8 @@ MODULE_PARM_DESC(align_buffer_size, "Force buffer and period sizes to be multiple of 128 bytes."); #ifdef CONFIG_X86 -static bool hda_snoop = true; -module_param_named(snoop, hda_snoop, bool, 0444); +static int hda_snoop = -1; +module_param_named(snoop, hda_snoop, bint, 0444); MODULE_PARM_DESC(snoop, "Enable/disable snooping"); #else #define hda_snoop true @@ -1370,29 +1370,33 @@ static void check_msi(struct azx *chip) /* check the snoop mode availability */ static void azx_check_snoop_available(struct azx *chip) { - bool snoop = chip->snoop; + int snoop = hda_snoop; + if (snoop >= 0) { + dev_info(chip->card->dev, "Force to %s mode by module option\n", + snoop ? "snoop" : "non-snoop"); + chip->snoop = snoop; + return; + } + + snoop = true; if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE && chip->driver_type == AZX_DRIVER_VIA) { /* force to non-snoop mode for a new VIA controller * when BIOS is set */ - if (snoop) { - u8 val; - pci_read_config_byte(chip->pci, 0x42, &val); - if (!(val & 0x80) && chip->pci->revision == 0x30) - snoop = false; - } + u8 val; + pci_read_config_byte(chip->pci, 0x42, &val); + if (!(val & 0x80) && chip->pci->revision == 0x30) + snoop = false; } if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF) snoop = false; - if (snoop != chip->snoop) { - dev_info(chip->card->dev, "Force to %s mode\n", - snoop ? "snoop" : "non-snoop"); - chip->snoop = snoop; - } + chip->snoop = snoop; + if (!snoop) + dev_info(chip->card->dev, "Force to non-snoop mode\n"); } static void azx_probe_work(struct work_struct *work) @@ -1452,7 +1456,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, check_probe_mask(chip, dev); chip->single_cmd = single_cmd; - chip->snoop = hda_snoop; azx_check_snoop_available(chip); if (bdl_pos_adj[dev] < 0) { -- cgit v1.1 From c5f0406bdce6eebc9147a54f4be0250618d3a423 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 28 Nov 2014 10:41:28 +0800 Subject: ASoC: Intel: Correct the xmax volume The xmax volume should be corrected to ARRAY_SIZE(volume_map)-1, otherwise, the xmax value will be mapped to 0 wrongly. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-haswell-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 7aa348d..f993079 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -298,19 +298,19 @@ static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); static const struct snd_kcontrol_new hsw_volume_controls[] = { /* Global DSP volume */ SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8, - ARRAY_SIZE(volume_map) -1, 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_volume_get, hsw_volume_put, hsw_vol_tlv), /* Offload 0 volume */ SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Offload 1 volume */ SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; -- cgit v1.1 From 7bb73cbd073c495b5aed9ec446faa17a262f18be Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 28 Nov 2014 21:52:11 +0800 Subject: ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell Move capture PCM pin from PCM4 to PCM0 for Broadwell/Haswell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/haswell.c | 14 ++------------ sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 3 files changed, 20 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 3981982..cb8a482 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { /* Front End DAI links */ { .name = "System", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .init = haswell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, /* Back End DAI links */ { diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index f993079..f6a9acf 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -781,6 +783,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -817,17 +826,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- cgit v1.1 From 7a2e9ddc903225d8fb3a510a842144a239017ee4 Mon Sep 17 00:00:00 2001 From: Jurgen Kramer Date: Fri, 28 Nov 2014 17:32:53 +0100 Subject: ALSA: usb-audio: Add native DSD support for Denon/Marantz DACs This patch adds native DSD support for the following devices: - Marantz SA-14S1 - Marants HD-DAC1 Signed-off-by: Jurgen Kramer Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c1018e..a9d4add 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1213,5 +1213,16 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; } + /* Denon/Marantz devices with USB DAC functionality */ + switch (chip->usb_id) { + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + if (fp->altsetting == 2) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; + default: + break; + } + return 0; } -- cgit v1.1 From 6874daad4b0fbed5b2f9bef7f4d3f2b895463a95 Mon Sep 17 00:00:00 2001 From: Jurgen Kramer Date: Fri, 28 Nov 2014 17:32:54 +0100 Subject: ALSA: usb-audio: Add mode select quirk for Denon/Marantz DACs Denon/Marantz USB DACs need a specific vendor command to switch between PCM and DSD mode. This patch adds a new quirk function to switch between the two modes using the specific USB vendor command. This patch applies to the following devices: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 5 +++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 46 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c62a165..0d8aba5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -482,6 +482,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* set interface */ if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { + + err = snd_usb_select_mode_quirk(subs, fmt); + if (err < 0) + return -EIO; + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); if (err < 0) { dev_err(&dev->dev, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a9d4add..e0cde74 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1111,6 +1111,44 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } + +/* Marantz/Denon USB DACs need a vendor cmd to switch + * between PCM and native DSD mode + */ +int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + struct usb_device *dev = subs->dev; + int err; + + switch (subs->stream->chip->usb_id) { + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + + /* First switch to alt set 0, otherwise the mode switch cmd + * will not be accepted by the DAC + */ + err = usb_set_interface(dev, fmt->iface, 0); + if (err < 0) + return err; + + mdelay(20); /* Delay needed after setting the interface */ + + switch (fmt->altsetting) { + case 2: /* DSD mode requested */ + case 1: /* PCM mode requested */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, + USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, + fmt->altsetting - 1, 1, NULL, 0); + if (err < 0) + return err; + break; + } + mdelay(20); + } + return 0; +} + void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) { /* diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 665e972..1b86238 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -31,6 +31,9 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); +int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt); + u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int sample_bytes); -- cgit v1.1 From 81fc5ad51539e765cd470f1d0d9d0477d1c4c30c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:10 +0900 Subject: ALSA: dice: suppress checkpatch.pl warnings The checkpatch.pl generates some warnings due to: - C99 comment - a line over 80 characters - min() for parameters with different types Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index e3a04d6..72af110 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -133,7 +133,7 @@ static inline u64 global_address(struct dice *dice, unsigned int offset) return DICE_PRIVATE_SPACE + dice->global_offset + offset; } -// TODO: rx index +/* TODO: rx index */ static inline u64 rx_address(struct dice *dice, unsigned int offset) { return DICE_PRIVATE_SPACE + dice->rx_offset + offset; @@ -721,13 +721,14 @@ static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, event.lock_status.status = dice->dev_lock_count > 0; dice->dev_lock_changed = false; - count = min(count, (long)sizeof(event.lock_status)); + count = min_t(long, count, sizeof(event.lock_status)); } else { - event.dice_notification.type = SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; + event.dice_notification.type = + SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; event.dice_notification.notification = dice->notification_bits; dice->notification_bits = 0; - count = min(count, (long)sizeof(event.dice_notification)); + count = min_t(long, count, sizeof(event.dice_notification)); } spin_unlock_irq(&dice->lock); -- cgit v1.1 From 732d153fbeac1774393508b3cec4b1d09e567585 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:11 +0900 Subject: ALSA: dice: Rename structure and its members Currently, dice driver supports AMDTP out-stream. In followed commits, AMDTP in-stream will be supported but current name of members in dice structure are not propper. This commit renames these members to proper name. Additionally, for easily distinguishing local symbols from structure tag, rename dice tag into snd_dice. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice.c | 162 +++++++++++++++++++++++++------------------------- 1 file changed, 81 insertions(+), 81 deletions(-) diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 72af110..eafd74b 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -32,7 +32,7 @@ #include "dice-interface.h" -struct dice { +struct snd_dice { struct snd_card *card; struct fw_unit *unit; spinlock_t lock; @@ -50,8 +50,8 @@ struct dice { struct completion clock_accepted; wait_queue_head_t hwdep_wait; u32 notification_bits; - struct fw_iso_resources resources; - struct amdtp_stream stream; + struct fw_iso_resources rx_resources; + struct amdtp_stream rx_stream; }; MODULE_DESCRIPTION("DICE driver"); @@ -87,13 +87,13 @@ static unsigned int rate_index_to_mode(unsigned int rate_index) return ((int)rate_index - 1) / 2; } -static void dice_lock_changed(struct dice *dice) +static void dice_lock_changed(struct snd_dice *dice) { dice->dev_lock_changed = true; wake_up(&dice->hwdep_wait); } -static int dice_try_lock(struct dice *dice) +static int dice_try_lock(struct snd_dice *dice) { int err; @@ -114,7 +114,7 @@ out: return err; } -static void dice_unlock(struct dice *dice) +static void dice_unlock(struct snd_dice *dice) { spin_lock_irq(&dice->lock); @@ -128,18 +128,18 @@ out: spin_unlock_irq(&dice->lock); } -static inline u64 global_address(struct dice *dice, unsigned int offset) +static inline u64 global_address(struct snd_dice *dice, unsigned int offset) { return DICE_PRIVATE_SPACE + dice->global_offset + offset; } /* TODO: rx index */ -static inline u64 rx_address(struct dice *dice, unsigned int offset) +static inline u64 rx_address(struct snd_dice *dice, unsigned int offset) { return DICE_PRIVATE_SPACE + dice->rx_offset + offset; } -static int dice_owner_set(struct dice *dice) +static int dice_owner_set(struct snd_dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; @@ -183,7 +183,7 @@ static int dice_owner_set(struct dice *dice) return err; } -static int dice_owner_update(struct dice *dice) +static int dice_owner_update(struct snd_dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; @@ -226,7 +226,7 @@ static int dice_owner_update(struct dice *dice) return err; } -static void dice_owner_clear(struct dice *dice) +static void dice_owner_clear(struct snd_dice *dice) { struct fw_device *device = fw_parent_device(dice->unit); __be64 *buffer; @@ -249,7 +249,7 @@ static void dice_owner_clear(struct dice *dice) dice->owner_generation = -1; } -static int dice_enable_set(struct dice *dice) +static int dice_enable_set(struct snd_dice *dice) { __be32 value; int err; @@ -267,7 +267,7 @@ static int dice_enable_set(struct dice *dice) return 0; } -static void dice_enable_clear(struct dice *dice) +static void dice_enable_clear(struct snd_dice *dice) { __be32 value; @@ -288,7 +288,7 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, int generation, unsigned long long offset, void *data, size_t length, void *callback_data) { - struct dice *dice = callback_data; + struct snd_dice *dice = callback_data; u32 bits; unsigned long flags; @@ -317,7 +317,7 @@ static void dice_notification(struct fw_card *card, struct fw_request *request, static int dice_rate_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct dice *dice = rule->private; + struct snd_dice *dice = rule->private; const struct snd_interval *channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *rate = @@ -344,7 +344,7 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, static int dice_channels_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct dice *dice = rule->private; + struct snd_dice *dice = rule->private; const struct snd_interval *rate = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = @@ -384,7 +384,7 @@ static int dice_open(struct snd_pcm_substream *substream) .periods_min = 1, .periods_max = UINT_MAX, }; - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; unsigned int i; int err; @@ -420,7 +420,7 @@ static int dice_open(struct snd_pcm_substream *substream) if (err < 0) goto err_lock; - err = amdtp_stream_add_pcm_hw_constraints(&dice->stream, runtime); + err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); if (err < 0) goto err_lock; @@ -434,47 +434,47 @@ error: static int dice_close(struct snd_pcm_substream *substream) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; dice_unlock(dice); return 0; } -static int dice_stream_start_packets(struct dice *dice) +static int dice_stream_start_packets(struct snd_dice *dice) { int err; - if (amdtp_stream_running(&dice->stream)) + if (amdtp_stream_running(&dice->rx_stream)) return 0; - err = amdtp_stream_start(&dice->stream, dice->resources.channel, + err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, fw_parent_device(dice->unit)->max_speed); if (err < 0) return err; err = dice_enable_set(dice); if (err < 0) { - amdtp_stream_stop(&dice->stream); + amdtp_stream_stop(&dice->rx_stream); return err; } return 0; } -static int dice_stream_start(struct dice *dice) +static int dice_stream_start(struct snd_dice *dice) { __be32 channel; int err; - if (!dice->resources.allocated) { - err = fw_iso_resources_allocate(&dice->resources, - amdtp_stream_get_max_payload(&dice->stream), + if (!dice->rx_resources.allocated) { + err = fw_iso_resources_allocate(&dice->rx_resources, + amdtp_stream_get_max_payload(&dice->rx_stream), fw_parent_device(dice->unit)->max_speed); if (err < 0) goto error; - channel = cpu_to_be32(dice->resources.channel); + channel = cpu_to_be32(dice->rx_resources.channel); err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, rx_address(dice, RX_ISOCHRONOUS), @@ -494,36 +494,36 @@ err_rx_channel: snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); err_resources: - fw_iso_resources_free(&dice->resources); + fw_iso_resources_free(&dice->rx_resources); error: return err; } -static void dice_stream_stop_packets(struct dice *dice) +static void dice_stream_stop_packets(struct snd_dice *dice) { - if (amdtp_stream_running(&dice->stream)) { + if (amdtp_stream_running(&dice->rx_stream)) { dice_enable_clear(dice); - amdtp_stream_stop(&dice->stream); + amdtp_stream_stop(&dice->rx_stream); } } -static void dice_stream_stop(struct dice *dice) +static void dice_stream_stop(struct snd_dice *dice) { __be32 channel; dice_stream_stop_packets(dice); - if (!dice->resources.allocated) + if (!dice->rx_resources.allocated) return; channel = cpu_to_be32((u32)-1); snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); - fw_iso_resources_free(&dice->resources); + fw_iso_resources_free(&dice->rx_resources); } -static int dice_change_rate(struct dice *dice, unsigned int clock_rate) +static int dice_change_rate(struct snd_dice *dice, unsigned int clock_rate) { __be32 value; int err; @@ -547,7 +547,7 @@ static int dice_change_rate(struct dice *dice, unsigned int clock_rate) static int dice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; unsigned int rate_index, mode, rate, channels, i; int err; @@ -585,24 +585,24 @@ static int dice_hw_params(struct snd_pcm_substream *substream, rate /= 2; channels *= 2; - dice->stream.double_pcm_frames = true; + dice->rx_stream.double_pcm_frames = true; } else { - dice->stream.double_pcm_frames = false; + dice->rx_stream.double_pcm_frames = false; } mode = rate_index_to_mode(rate_index); - amdtp_stream_set_parameters(&dice->stream, rate, channels, + amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, dice->rx_midi_ports[mode]); if (rate_index > 4) { channels /= 2; for (i = 0; i < channels; i++) { - dice->stream.pcm_positions[i] = i * 2; - dice->stream.pcm_positions[i + channels] = i * 2 + 1; + dice->rx_stream.pcm_positions[i] = i * 2; + dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; } } - amdtp_stream_set_pcm_format(&dice->stream, + amdtp_stream_set_pcm_format(&dice->rx_stream, params_format(hw_params)); return 0; @@ -610,7 +610,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, static int dice_hw_free(struct snd_pcm_substream *substream) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; mutex_lock(&dice->mutex); dice_stream_stop(dice); @@ -621,12 +621,12 @@ static int dice_hw_free(struct snd_pcm_substream *substream) static int dice_prepare(struct snd_pcm_substream *substream) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; int err; mutex_lock(&dice->mutex); - if (amdtp_streaming_error(&dice->stream)) + if (amdtp_streaming_error(&dice->rx_stream)) dice_stream_stop_packets(dice); err = dice_stream_start(dice); @@ -637,14 +637,14 @@ static int dice_prepare(struct snd_pcm_substream *substream) mutex_unlock(&dice->mutex); - amdtp_stream_pcm_prepare(&dice->stream); + amdtp_stream_pcm_prepare(&dice->rx_stream); return 0; } static int dice_trigger(struct snd_pcm_substream *substream, int cmd) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; struct snd_pcm_substream *pcm; switch (cmd) { @@ -657,19 +657,19 @@ static int dice_trigger(struct snd_pcm_substream *substream, int cmd) default: return -EINVAL; } - amdtp_stream_pcm_trigger(&dice->stream, pcm); + amdtp_stream_pcm_trigger(&dice->rx_stream, pcm); return 0; } static snd_pcm_uframes_t dice_pointer(struct snd_pcm_substream *substream) { - struct dice *dice = substream->private_data; + struct snd_dice *dice = substream->private_data; - return amdtp_stream_pcm_pointer(&dice->stream); + return amdtp_stream_pcm_pointer(&dice->rx_stream); } -static int dice_create_pcm(struct dice *dice) +static int dice_create_pcm(struct snd_dice *dice) { static struct snd_pcm_ops ops = { .open = dice_open, @@ -699,7 +699,7 @@ static int dice_create_pcm(struct dice *dice) static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, loff_t *offset) { - struct dice *dice = hwdep->private_data; + struct snd_dice *dice = hwdep->private_data; DEFINE_WAIT(wait); union snd_firewire_event event; @@ -742,7 +742,7 @@ static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, poll_table *wait) { - struct dice *dice = hwdep->private_data; + struct snd_dice *dice = hwdep->private_data; unsigned int events; poll_wait(file, &dice->hwdep_wait, wait); @@ -757,7 +757,7 @@ static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, return events; } -static int dice_hwdep_get_info(struct dice *dice, void __user *arg) +static int dice_hwdep_get_info(struct snd_dice *dice, void __user *arg) { struct fw_device *dev = fw_parent_device(dice->unit); struct snd_firewire_get_info info; @@ -776,7 +776,7 @@ static int dice_hwdep_get_info(struct dice *dice, void __user *arg) return 0; } -static int dice_hwdep_lock(struct dice *dice) +static int dice_hwdep_lock(struct snd_dice *dice) { int err; @@ -794,7 +794,7 @@ static int dice_hwdep_lock(struct dice *dice) return err; } -static int dice_hwdep_unlock(struct dice *dice) +static int dice_hwdep_unlock(struct snd_dice *dice) { int err; @@ -814,7 +814,7 @@ static int dice_hwdep_unlock(struct dice *dice) static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) { - struct dice *dice = hwdep->private_data; + struct snd_dice *dice = hwdep->private_data; spin_lock_irq(&dice->lock); if (dice->dev_lock_count == -1) @@ -827,7 +827,7 @@ static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, unsigned int cmd, unsigned long arg) { - struct dice *dice = hwdep->private_data; + struct snd_dice *dice = hwdep->private_data; switch (cmd) { case SNDRV_FIREWIRE_IOCTL_GET_INFO: @@ -852,7 +852,7 @@ static int dice_hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, #define dice_hwdep_compat_ioctl NULL #endif -static int dice_create_hwdep(struct dice *dice) +static int dice_create_hwdep(struct snd_dice *dice) { static const struct snd_hwdep_ops ops = { .read = dice_hwdep_read, @@ -876,7 +876,7 @@ static int dice_create_hwdep(struct dice *dice) return 0; } -static int dice_proc_read_mem(struct dice *dice, void *buffer, +static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, unsigned int offset_q, unsigned int quadlets) { unsigned int i; @@ -899,8 +899,8 @@ static const char *str_from_array(const char *const strs[], unsigned int count, { if (i < count) return strs[i]; - else - return "(unknown)"; + + return "(unknown)"; } static void dice_proc_fixup_string(char *s, unsigned int size) @@ -935,7 +935,7 @@ static void dice_proc_read(struct snd_info_entry *entry, "32000", "44100", "48000", "88200", "96000", "176400", "192000", "any low", "any mid", "any high", "none" }; - struct dice *dice = entry->private_data; + struct snd_dice *dice = entry->private_data; u32 sections[ARRAY_SIZE(section_names) * 2]; struct { u32 number; @@ -1110,7 +1110,7 @@ static void dice_proc_read(struct snd_info_entry *entry, } } -static void dice_create_proc(struct dice *dice) +static void dice_create_proc(struct snd_dice *dice) { struct snd_info_entry *entry; @@ -1120,9 +1120,9 @@ static void dice_create_proc(struct dice *dice) static void dice_card_free(struct snd_card *card) { - struct dice *dice = card->private_data; + struct snd_dice *dice = card->private_data; - amdtp_stream_destroy(&dice->stream); + amdtp_stream_destroy(&dice->rx_stream); fw_core_remove_address_handler(&dice->notification_handler); mutex_destroy(&dice->mutex); } @@ -1217,7 +1217,7 @@ static int dice_interface_check(struct fw_unit *unit) return 0; } -static int highest_supported_mode_rate(struct dice *dice, unsigned int mode) +static int highest_supported_mode_rate(struct snd_dice *dice, unsigned int mode) { int i; @@ -1229,7 +1229,7 @@ static int highest_supported_mode_rate(struct dice *dice, unsigned int mode) return -1; } -static int dice_read_mode_params(struct dice *dice, unsigned int mode) +static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) { __be32 values[2]; int rate_index, err; @@ -1257,7 +1257,7 @@ static int dice_read_mode_params(struct dice *dice, unsigned int mode) return 0; } -static int dice_read_params(struct dice *dice) +static int dice_read_params(struct snd_dice *dice) { __be32 pointers[6]; __be32 value; @@ -1298,7 +1298,7 @@ static int dice_read_params(struct dice *dice) return 0; } -static void dice_card_strings(struct dice *dice) +static void dice_card_strings(struct snd_dice *dice) { struct snd_card *card = dice->card; struct fw_device *dev = fw_parent_device(dice->unit); @@ -1336,7 +1336,7 @@ static void dice_card_strings(struct dice *dice) static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_card *card; - struct dice *dice; + struct snd_dice *dice; __be32 clock_sel; int err; @@ -1373,12 +1373,12 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto err_owner; - err = fw_iso_resources_init(&dice->resources, unit); + err = fw_iso_resources_init(&dice->rx_resources, unit); if (err < 0) goto err_owner; - dice->resources.channels_mask = 0x00000000ffffffffuLL; + dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; - err = amdtp_stream_init(&dice->stream, unit, AMDTP_OUT_STREAM, + err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, CIP_BLOCKING); if (err < 0) goto err_resources; @@ -1419,7 +1419,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) return 0; err_resources: - fw_iso_resources_destroy(&dice->resources); + fw_iso_resources_destroy(&dice->rx_resources); err_owner: dice_owner_clear(dice); err_notification_handler: @@ -1433,9 +1433,9 @@ error: static void dice_remove(struct fw_unit *unit) { - struct dice *dice = dev_get_drvdata(&unit->device); + struct snd_dice *dice = dev_get_drvdata(&unit->device); - amdtp_stream_pcm_abort(&dice->stream); + amdtp_stream_pcm_abort(&dice->rx_stream); snd_card_disconnect(dice->card); @@ -1451,7 +1451,7 @@ static void dice_remove(struct fw_unit *unit) static void dice_bus_reset(struct fw_unit *unit) { - struct dice *dice = dev_get_drvdata(&unit->device); + struct snd_dice *dice = dev_get_drvdata(&unit->device); /* * On a bus reset, the DICE firmware disables streaming and then goes @@ -1461,7 +1461,7 @@ static void dice_bus_reset(struct fw_unit *unit) * to stop so that the application can restart them in an orderly * manner. */ - amdtp_stream_pcm_abort(&dice->stream); + amdtp_stream_pcm_abort(&dice->rx_stream); mutex_lock(&dice->mutex); @@ -1470,7 +1470,7 @@ static void dice_bus_reset(struct fw_unit *unit) dice_owner_update(dice); - fw_iso_resources_update(&dice->resources); + fw_iso_resources_update(&dice->rx_resources); mutex_unlock(&dice->mutex); } -- cgit v1.1 From 14ff6a094815988b018ea4d698c2e2cc3ceee27c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:12 +0900 Subject: ALSA: dice: Move file to its own directory In followed commits, dice driver is split into several files. For easily managing these files, this commit adds subdirectory and move file into the directory. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/Makefile | 3 +- sound/firewire/dice-interface.h | 371 --------- sound/firewire/dice.c | 1512 ---------------------------------- sound/firewire/dice/Makefile | 2 + sound/firewire/dice/dice-interface.h | 371 +++++++++ sound/firewire/dice/dice.c | 1512 ++++++++++++++++++++++++++++++++++ 6 files changed, 1886 insertions(+), 1885 deletions(-) delete mode 100644 sound/firewire/dice-interface.h delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-interface.h create mode 100644 sound/firewire/dice/dice.c diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index fad8d49..c50761c 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,12 +1,11 @@ snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp.o -snd-dice-objs := dice.o snd-firewire-speakers-objs := speakers.o snd-isight-objs := isight.o snd-scs1x-objs := scs1x.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o -obj-$(CONFIG_SND_DICE) += snd-dice.o +obj-$(CONFIG_SND_DICE) += dice/ obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o obj-$(CONFIG_SND_ISIGHT) += snd-isight.o obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o diff --git a/sound/firewire/dice-interface.h b/sound/firewire/dice-interface.h deleted file mode 100644 index 27b044f..0000000 --- a/sound/firewire/dice-interface.h +++ /dev/null @@ -1,371 +0,0 @@ -#ifndef SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED -#define SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED - -/* - * DICE device interface definitions - */ - -/* - * Generally, all registers can be read like memory, i.e., with quadlet read or - * block read transactions with at least quadlet-aligned offset and length. - * Writes are not allowed except where noted; quadlet-sized registers must be - * written with a quadlet write transaction. - * - * All values are in big endian. The DICE firmware runs on a little-endian CPU - * and just byte-swaps _all_ quadlets on the bus, so values without endianness - * (e.g. strings) get scrambled and must be byte-swapped again by the driver. - */ - -/* - * Streaming is handled by the "DICE driver" interface. Its registers are - * located in this private address space. - */ -#define DICE_PRIVATE_SPACE 0xffffe0000000uLL - -/* - * The registers are organized in several sections, which are organized - * separately to allow them to be extended individually. Whether a register is - * supported can be detected by checking its offset against its section's size. - * - * The section offset values are relative to DICE_PRIVATE_SPACE; the offset/ - * size values are measured in quadlets. Read-only. - */ -#define DICE_GLOBAL_OFFSET 0x00 -#define DICE_GLOBAL_SIZE 0x04 -#define DICE_TX_OFFSET 0x08 -#define DICE_TX_SIZE 0x0c -#define DICE_RX_OFFSET 0x10 -#define DICE_RX_SIZE 0x14 -#define DICE_EXT_SYNC_OFFSET 0x18 -#define DICE_EXT_SYNC_SIZE 0x1c -#define DICE_UNUSED2_OFFSET 0x20 -#define DICE_UNUSED2_SIZE 0x24 - -/* - * Global settings. - */ - -/* - * Stores the full 64-bit address (node ID and offset in the node's address - * space) where the device will send notifications. Must be changed with - * a compare/swap transaction by the owner. This register is automatically - * cleared on a bus reset. - */ -#define GLOBAL_OWNER 0x000 -#define OWNER_NO_OWNER 0xffff000000000000uLL -#define OWNER_NODE_SHIFT 48 - -/* - * A bitmask with asynchronous events; read-only. When any event(s) happen, - * the bits of previous events are cleared, and the value of this register is - * also written to the address stored in the owner register. - */ -#define GLOBAL_NOTIFICATION 0x008 -/* Some registers in the Rx/Tx sections may have changed. */ -#define NOTIFY_RX_CFG_CHG 0x00000001 -#define NOTIFY_TX_CFG_CHG 0x00000002 -/* Lock status of the current clock source may have changed. */ -#define NOTIFY_LOCK_CHG 0x00000010 -/* Write to the clock select register has been finished. */ -#define NOTIFY_CLOCK_ACCEPTED 0x00000020 -/* Lock status of some clock source has changed. */ -#define NOTIFY_EXT_STATUS 0x00000040 -/* Other bits may be used for device-specific events. */ - -/* - * A name that can be customized for each device; read/write. Padded with zero - * bytes. Quadlets are byte-swapped. The encoding is whatever the host driver - * happens to be using. - */ -#define GLOBAL_NICK_NAME 0x00c -#define NICK_NAME_SIZE 64 - -/* - * The current sample rate and clock source; read/write. Whether a clock - * source or sample rate is supported is device-specific; the internal clock - * source is always available. Low/mid/high = up to 48/96/192 kHz. This - * register can be changed even while streams are running. - */ -#define GLOBAL_CLOCK_SELECT 0x04c -#define CLOCK_SOURCE_MASK 0x000000ff -#define CLOCK_SOURCE_AES1 0x00000000 -#define CLOCK_SOURCE_AES2 0x00000001 -#define CLOCK_SOURCE_AES3 0x00000002 -#define CLOCK_SOURCE_AES4 0x00000003 -#define CLOCK_SOURCE_AES_ANY 0x00000004 -#define CLOCK_SOURCE_ADAT 0x00000005 -#define CLOCK_SOURCE_TDIF 0x00000006 -#define CLOCK_SOURCE_WC 0x00000007 -#define CLOCK_SOURCE_ARX1 0x00000008 -#define CLOCK_SOURCE_ARX2 0x00000009 -#define CLOCK_SOURCE_ARX3 0x0000000a -#define CLOCK_SOURCE_ARX4 0x0000000b -#define CLOCK_SOURCE_INTERNAL 0x0000000c -#define CLOCK_RATE_MASK 0x0000ff00 -#define CLOCK_RATE_32000 0x00000000 -#define CLOCK_RATE_44100 0x00000100 -#define CLOCK_RATE_48000 0x00000200 -#define CLOCK_RATE_88200 0x00000300 -#define CLOCK_RATE_96000 0x00000400 -#define CLOCK_RATE_176400 0x00000500 -#define CLOCK_RATE_192000 0x00000600 -#define CLOCK_RATE_ANY_LOW 0x00000700 -#define CLOCK_RATE_ANY_MID 0x00000800 -#define CLOCK_RATE_ANY_HIGH 0x00000900 -#define CLOCK_RATE_NONE 0x00000a00 -#define CLOCK_RATE_SHIFT 8 - -/* - * Enable streaming; read/write. Writing a non-zero value (re)starts all - * streams that have a valid iso channel set; zero stops all streams. The - * streams' parameters must be configured before starting. This register is - * automatically cleared on a bus reset. - */ -#define GLOBAL_ENABLE 0x050 - -/* - * Status of the sample clock; read-only. - */ -#define GLOBAL_STATUS 0x054 -/* The current clock source is locked. */ -#define STATUS_SOURCE_LOCKED 0x00000001 -/* The actual sample rate; CLOCK_RATE_32000-_192000 or _NONE. */ -#define STATUS_NOMINAL_RATE_MASK 0x0000ff00 - -/* - * Status of all clock sources; read-only. - */ -#define GLOBAL_EXTENDED_STATUS 0x058 -/* - * The _LOCKED bits always show the current status; any change generates - * a notification. - */ -#define EXT_STATUS_AES1_LOCKED 0x00000001 -#define EXT_STATUS_AES2_LOCKED 0x00000002 -#define EXT_STATUS_AES3_LOCKED 0x00000004 -#define EXT_STATUS_AES4_LOCKED 0x00000008 -#define EXT_STATUS_ADAT_LOCKED 0x00000010 -#define EXT_STATUS_TDIF_LOCKED 0x00000020 -#define EXT_STATUS_ARX1_LOCKED 0x00000040 -#define EXT_STATUS_ARX2_LOCKED 0x00000080 -#define EXT_STATUS_ARX3_LOCKED 0x00000100 -#define EXT_STATUS_ARX4_LOCKED 0x00000200 -#define EXT_STATUS_WC_LOCKED 0x00000400 -/* - * The _SLIP bits do not generate notifications; a set bit indicates that an - * error occurred since the last time when this register was read with - * a quadlet read transaction. - */ -#define EXT_STATUS_AES1_SLIP 0x00010000 -#define EXT_STATUS_AES2_SLIP 0x00020000 -#define EXT_STATUS_AES3_SLIP 0x00040000 -#define EXT_STATUS_AES4_SLIP 0x00080000 -#define EXT_STATUS_ADAT_SLIP 0x00100000 -#define EXT_STATUS_TDIF_SLIP 0x00200000 -#define EXT_STATUS_ARX1_SLIP 0x00400000 -#define EXT_STATUS_ARX2_SLIP 0x00800000 -#define EXT_STATUS_ARX3_SLIP 0x01000000 -#define EXT_STATUS_ARX4_SLIP 0x02000000 -#define EXT_STATUS_WC_SLIP 0x04000000 - -/* - * The measured rate of the current clock source, in Hz; read-only. - */ -#define GLOBAL_SAMPLE_RATE 0x05c - -/* - * The version of the DICE driver specification that this device conforms to; - * read-only. - */ -#define GLOBAL_VERSION 0x060 - -/* Some old firmware versions do not have the following global registers: */ - -/* - * Supported sample rates and clock sources; read-only. - */ -#define GLOBAL_CLOCK_CAPABILITIES 0x064 -#define CLOCK_CAP_RATE_32000 0x00000001 -#define CLOCK_CAP_RATE_44100 0x00000002 -#define CLOCK_CAP_RATE_48000 0x00000004 -#define CLOCK_CAP_RATE_88200 0x00000008 -#define CLOCK_CAP_RATE_96000 0x00000010 -#define CLOCK_CAP_RATE_176400 0x00000020 -#define CLOCK_CAP_RATE_192000 0x00000040 -#define CLOCK_CAP_SOURCE_AES1 0x00010000 -#define CLOCK_CAP_SOURCE_AES2 0x00020000 -#define CLOCK_CAP_SOURCE_AES3 0x00040000 -#define CLOCK_CAP_SOURCE_AES4 0x00080000 -#define CLOCK_CAP_SOURCE_AES_ANY 0x00100000 -#define CLOCK_CAP_SOURCE_ADAT 0x00200000 -#define CLOCK_CAP_SOURCE_TDIF 0x00400000 -#define CLOCK_CAP_SOURCE_WC 0x00800000 -#define CLOCK_CAP_SOURCE_ARX1 0x01000000 -#define CLOCK_CAP_SOURCE_ARX2 0x02000000 -#define CLOCK_CAP_SOURCE_ARX3 0x04000000 -#define CLOCK_CAP_SOURCE_ARX4 0x08000000 -#define CLOCK_CAP_SOURCE_INTERNAL 0x10000000 - -/* - * Names of all clock sources; read-only. Quadlets are byte-swapped. Names - * are separated with one backslash, the list is terminated with two - * backslashes. Unused clock sources are included. - */ -#define GLOBAL_CLOCK_SOURCE_NAMES 0x068 -#define CLOCK_SOURCE_NAMES_SIZE 256 - -/* - * Capture stream settings. This section includes the number/size registers - * and the registers of all streams. - */ - -/* - * The number of supported capture streams; read-only. - */ -#define TX_NUMBER 0x000 - -/* - * The size of one stream's register block, in quadlets; read-only. The - * registers of the first stream follow immediately afterwards; the registers - * of the following streams are offset by this register's value. - */ -#define TX_SIZE 0x004 - -/* - * The isochronous channel number on which packets are sent, or -1 if the - * stream is not to be used; read/write. - */ -#define TX_ISOCHRONOUS 0x008 - -/* - * The number of audio channels; read-only. There will be one quadlet per - * channel; the first channel is the first quadlet in a data block. - */ -#define TX_NUMBER_AUDIO 0x00c - -/* - * The number of MIDI ports, 0-8; read-only. If > 0, there will be one - * additional quadlet in each data block, following the audio quadlets. - */ -#define TX_NUMBER_MIDI 0x010 - -/* - * The speed at which the packets are sent, SCODE_100-_400; read/write. - */ -#define TX_SPEED 0x014 - -/* - * Names of all audio channels; read-only. Quadlets are byte-swapped. Names - * are separated with one backslash, the list is terminated with two - * backslashes. - */ -#define TX_NAMES 0x018 -#define TX_NAMES_SIZE 256 - -/* - * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio - * channel. - */ -#define TX_AC3_CAPABILITIES 0x118 - -/* - * Send audio data with IEC60958 label; read/write. Bitmask with one bit per - * audio channel. This register can be changed even while the stream is - * running. - */ -#define TX_AC3_ENABLE 0x11c - -/* - * Playback stream settings. This section includes the number/size registers - * and the registers of all streams. - */ - -/* - * The number of supported playback streams; read-only. - */ -#define RX_NUMBER 0x000 - -/* - * The size of one stream's register block, in quadlets; read-only. The - * registers of the first stream follow immediately afterwards; the registers - * of the following streams are offset by this register's value. - */ -#define RX_SIZE 0x004 - -/* - * The isochronous channel number on which packets are received, or -1 if the - * stream is not to be used; read/write. - */ -#define RX_ISOCHRONOUS 0x008 - -/* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x00c - -/* - * The number of audio channels; read-only. There will be one quadlet per - * channel. - */ -#define RX_NUMBER_AUDIO 0x010 - -/* - * The number of MIDI ports, 0-8; read-only. If > 0, there will be one - * additional quadlet in each data block, following the audio quadlets. - */ -#define RX_NUMBER_MIDI 0x014 - -/* - * Names of all audio channels; read-only. Quadlets are byte-swapped. Names - * are separated with one backslash, the list is terminated with two - * backslashes. - */ -#define RX_NAMES 0x018 -#define RX_NAMES_SIZE 256 - -/* - * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio - * channel. - */ -#define RX_AC3_CAPABILITIES 0x118 - -/* - * Receive audio data with IEC60958 label; read/write. Bitmask with one bit - * per audio channel. This register can be changed even while the stream is - * running. - */ -#define RX_AC3_ENABLE 0x11c - -/* - * Extended synchronization information. - * This section can be read completely with a block read request. - */ - -/* - * Current clock source; read-only. - */ -#define EXT_SYNC_CLOCK_SOURCE 0x000 - -/* - * Clock source is locked (boolean); read-only. - */ -#define EXT_SYNC_LOCKED 0x004 - -/* - * Current sample rate (CLOCK_RATE_* >> CLOCK_RATE_SHIFT), _32000-_192000 or - * _NONE; read-only. - */ -#define EXT_SYNC_RATE 0x008 - -/* - * ADAT user data bits; read-only. - */ -#define EXT_SYNC_ADAT_USER_DATA 0x00c -/* The data bits, if available. */ -#define ADAT_USER_DATA_MASK 0x0f -/* The data bits are not available. */ -#define ADAT_USER_DATA_NO_DATA 0x10 - -#endif diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c deleted file mode 100644 index eafd74b..0000000 --- a/sound/firewire/dice.c +++ /dev/null @@ -1,1512 +0,0 @@ -/* - * TC Applied Technologies Digital Interface Communications Engine driver - * - * Copyright (c) Clemens Ladisch - * Licensed under the terms of the GNU General Public License, version 2. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "amdtp.h" -#include "iso-resources.h" -#include "lib.h" -#include "dice-interface.h" - - -struct snd_dice { - struct snd_card *card; - struct fw_unit *unit; - spinlock_t lock; - struct mutex mutex; - unsigned int global_offset; - unsigned int rx_offset; - unsigned int clock_caps; - unsigned int rx_channels[3]; - unsigned int rx_midi_ports[3]; - struct fw_address_handler notification_handler; - int owner_generation; - int dev_lock_count; /* > 0 driver, < 0 userspace */ - bool dev_lock_changed; - bool global_enabled; - struct completion clock_accepted; - wait_queue_head_t hwdep_wait; - u32 notification_bits; - struct fw_iso_resources rx_resources; - struct amdtp_stream rx_stream; -}; - -MODULE_DESCRIPTION("DICE driver"); -MODULE_AUTHOR("Clemens Ladisch "); -MODULE_LICENSE("GPL v2"); - -static const unsigned int dice_rates[] = { - /* mode 0 */ - [0] = 32000, - [1] = 44100, - [2] = 48000, - /* mode 1 */ - [3] = 88200, - [4] = 96000, - /* mode 2 */ - [5] = 176400, - [6] = 192000, -}; - -static unsigned int rate_to_index(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) - if (dice_rates[i] == rate) - return i; - - return 0; -} - -static unsigned int rate_index_to_mode(unsigned int rate_index) -{ - return ((int)rate_index - 1) / 2; -} - -static void dice_lock_changed(struct snd_dice *dice) -{ - dice->dev_lock_changed = true; - wake_up(&dice->hwdep_wait); -} - -static int dice_try_lock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count < 0) { - err = -EBUSY; - goto out; - } - - if (dice->dev_lock_count++ == 0) - dice_lock_changed(dice); - err = 0; - -out: - spin_unlock_irq(&dice->lock); - - return err; -} - -static void dice_unlock(struct snd_dice *dice) -{ - spin_lock_irq(&dice->lock); - - if (WARN_ON(dice->dev_lock_count <= 0)) - goto out; - - if (--dice->dev_lock_count == 0) - dice_lock_changed(dice); - -out: - spin_unlock_irq(&dice->lock); -} - -static inline u64 global_address(struct snd_dice *dice, unsigned int offset) -{ - return DICE_PRIVATE_SPACE + dice->global_offset + offset; -} - -/* TODO: rx index */ -static inline u64 rx_address(struct snd_dice *dice, unsigned int offset) -{ - return DICE_PRIVATE_SPACE + dice->rx_offset + offset; -} - -static int dice_owner_set(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - int err, errors = 0; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return -ENOMEM; - - for (;;) { - buffer[0] = cpu_to_be64(OWNER_NO_OWNER); - buffer[1] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - - dice->owner_generation = device->generation; - smp_rmb(); /* node_id vs. generation */ - err = snd_fw_transaction(dice->unit, - TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, - FW_FIXED_GENERATION | - dice->owner_generation); - - if (err == 0) { - if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { - dev_err(&dice->unit->device, - "device is already in use\n"); - err = -EBUSY; - } - break; - } - if (err != -EAGAIN || ++errors >= 3) - break; - - msleep(20); - } - - kfree(buffer); - - return err; -} - -static int dice_owner_update(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - int err; - - if (dice->owner_generation == -1) - return 0; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return -ENOMEM; - - buffer[0] = cpu_to_be64(OWNER_NO_OWNER); - buffer[1] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - - dice->owner_generation = device->generation; - smp_rmb(); /* node_id vs. generation */ - err = snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, - FW_FIXED_GENERATION | dice->owner_generation); - - if (err == 0) { - if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { - dev_err(&dice->unit->device, - "device is already in use\n"); - err = -EBUSY; - } - } else if (err == -EAGAIN) { - err = 0; /* try again later */ - } - - kfree(buffer); - - if (err < 0) - dice->owner_generation = -1; - - return err; -} - -static void dice_owner_clear(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return; - - buffer[0] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - buffer[1] = cpu_to_be64(OWNER_NO_OWNER); - snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, FW_QUIET | - FW_FIXED_GENERATION | dice->owner_generation); - - kfree(buffer); - - dice->owner_generation = -1; -} - -static int dice_enable_set(struct snd_dice *dice) -{ - __be32 value; - int err; - - value = cpu_to_be32(1); - err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_ENABLE), - &value, 4, - FW_FIXED_GENERATION | dice->owner_generation); - if (err < 0) - return err; - - dice->global_enabled = true; - - return 0; -} - -static void dice_enable_clear(struct snd_dice *dice) -{ - __be32 value; - - if (!dice->global_enabled) - return; - - value = 0; - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_ENABLE), - &value, 4, FW_QUIET | - FW_FIXED_GENERATION | dice->owner_generation); - - dice->global_enabled = false; -} - -static void dice_notification(struct fw_card *card, struct fw_request *request, - int tcode, int destination, int source, - int generation, unsigned long long offset, - void *data, size_t length, void *callback_data) -{ - struct snd_dice *dice = callback_data; - u32 bits; - unsigned long flags; - - if (tcode != TCODE_WRITE_QUADLET_REQUEST) { - fw_send_response(card, request, RCODE_TYPE_ERROR); - return; - } - if ((offset & 3) != 0) { - fw_send_response(card, request, RCODE_ADDRESS_ERROR); - return; - } - - bits = be32_to_cpup(data); - - spin_lock_irqsave(&dice->lock, flags); - dice->notification_bits |= bits; - spin_unlock_irqrestore(&dice->lock, flags); - - fw_send_response(card, request, RCODE_COMPLETE); - - if (bits & NOTIFY_CLOCK_ACCEPTED) - complete(&dice->clock_accepted); - wake_up(&dice->hwdep_wait); -} - -static int dice_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_dice *dice = rule->private; - const struct snd_interval *channels = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval allowed_rates = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, mode; - - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) { - mode = rate_index_to_mode(i); - if ((dice->clock_caps & (1 << i)) && - snd_interval_test(channels, dice->rx_channels[mode])) { - allowed_rates.min = min(allowed_rates.min, - dice_rates[i]); - allowed_rates.max = max(allowed_rates.max, - dice_rates[i]); - } - } - - return snd_interval_refine(rate, &allowed_rates); -} - -static int dice_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_dice *dice = rule->private; - const struct snd_interval *rate = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval allowed_channels = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, mode; - - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) - if ((dice->clock_caps & (1 << i)) && - snd_interval_test(rate, dice_rates[i])) { - mode = rate_index_to_mode(i); - allowed_channels.min = min(allowed_channels.min, - dice->rx_channels[mode]); - allowed_channels.max = max(allowed_channels.max, - dice->rx_channels[mode]); - } - - return snd_interval_refine(channels, &allowed_channels); -} - -static int dice_open(struct snd_pcm_substream *substream) -{ - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = UINT_MAX, - .channels_max = 0, - .buffer_bytes_max = 16 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct snd_dice *dice = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int i; - int err; - - err = dice_try_lock(dice); - if (err < 0) - goto error; - - runtime->hw = hardware; - - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) - if (dice->clock_caps & (1 << i)) - runtime->hw.rates |= - snd_pcm_rate_to_rate_bit(dice_rates[i]); - snd_pcm_limit_hw_rates(runtime); - - for (i = 0; i < 3; ++i) - if (dice->rx_channels[i]) { - runtime->hw.channels_min = min(runtime->hw.channels_min, - dice->rx_channels[i]); - runtime->hw.channels_max = max(runtime->hw.channels_max, - dice->rx_channels[i]); - } - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - dice_rate_constraint, dice, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - goto err_lock; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dice_channels_constraint, dice, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - goto err_lock; - - err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); - if (err < 0) - goto err_lock; - - return 0; - -err_lock: - dice_unlock(dice); -error: - return err; -} - -static int dice_close(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - dice_unlock(dice); - - return 0; -} - -static int dice_stream_start_packets(struct snd_dice *dice) -{ - int err; - - if (amdtp_stream_running(&dice->rx_stream)) - return 0; - - err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, - fw_parent_device(dice->unit)->max_speed); - if (err < 0) - return err; - - err = dice_enable_set(dice); - if (err < 0) { - amdtp_stream_stop(&dice->rx_stream); - return err; - } - - return 0; -} - -static int dice_stream_start(struct snd_dice *dice) -{ - __be32 channel; - int err; - - if (!dice->rx_resources.allocated) { - err = fw_iso_resources_allocate(&dice->rx_resources, - amdtp_stream_get_max_payload(&dice->rx_stream), - fw_parent_device(dice->unit)->max_speed); - if (err < 0) - goto error; - - channel = cpu_to_be32(dice->rx_resources.channel); - err = snd_fw_transaction(dice->unit, - TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), - &channel, 4, 0); - if (err < 0) - goto err_resources; - } - - err = dice_stream_start_packets(dice); - if (err < 0) - goto err_rx_channel; - - return 0; - -err_rx_channel: - channel = cpu_to_be32((u32)-1); - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); -err_resources: - fw_iso_resources_free(&dice->rx_resources); -error: - return err; -} - -static void dice_stream_stop_packets(struct snd_dice *dice) -{ - if (amdtp_stream_running(&dice->rx_stream)) { - dice_enable_clear(dice); - amdtp_stream_stop(&dice->rx_stream); - } -} - -static void dice_stream_stop(struct snd_dice *dice) -{ - __be32 channel; - - dice_stream_stop_packets(dice); - - if (!dice->rx_resources.allocated) - return; - - channel = cpu_to_be32((u32)-1); - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); - - fw_iso_resources_free(&dice->rx_resources); -} - -static int dice_change_rate(struct snd_dice *dice, unsigned int clock_rate) -{ - __be32 value; - int err; - - reinit_completion(&dice->clock_accepted); - - value = cpu_to_be32(clock_rate | CLOCK_SOURCE_ARX1); - err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &value, 4, 0); - if (err < 0) - return err; - - if (!wait_for_completion_timeout(&dice->clock_accepted, - msecs_to_jiffies(100))) - dev_warn(&dice->unit->device, "clock change timed out\n"); - - return 0; -} - -static int dice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_dice *dice = substream->private_data; - unsigned int rate_index, mode, rate, channels, i; - int err; - - mutex_lock(&dice->mutex); - dice_stream_stop(dice); - mutex_unlock(&dice->mutex); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - - rate = params_rate(hw_params); - rate_index = rate_to_index(rate); - err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); - if (err < 0) - return err; - - /* - * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in - * one data block of AMDTP packet. Thus sampling transfer frequency is - * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are - * transferred on AMDTP packets at 96 kHz. Two successive samples of a - * channel are stored consecutively in the packet. This quirk is called - * as 'Dual Wire'. - * For this quirk, blocking mode is required and PCM buffer size should - * be aligned to SYT_INTERVAL. - */ - channels = params_channels(hw_params); - if (rate_index > 4) { - if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { - err = -ENOSYS; - return err; - } - - rate /= 2; - channels *= 2; - dice->rx_stream.double_pcm_frames = true; - } else { - dice->rx_stream.double_pcm_frames = false; - } - - mode = rate_index_to_mode(rate_index); - amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, - dice->rx_midi_ports[mode]); - if (rate_index > 4) { - channels /= 2; - - for (i = 0; i < channels; i++) { - dice->rx_stream.pcm_positions[i] = i * 2; - dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; - } - } - - amdtp_stream_set_pcm_format(&dice->rx_stream, - params_format(hw_params)); - - return 0; -} - -static int dice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - mutex_lock(&dice->mutex); - dice_stream_stop(dice); - mutex_unlock(&dice->mutex); - - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -static int dice_prepare(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - int err; - - mutex_lock(&dice->mutex); - - if (amdtp_streaming_error(&dice->rx_stream)) - dice_stream_stop_packets(dice); - - err = dice_stream_start(dice); - if (err < 0) { - mutex_unlock(&dice->mutex); - return err; - } - - mutex_unlock(&dice->mutex); - - amdtp_stream_pcm_prepare(&dice->rx_stream); - - return 0; -} - -static int dice_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_dice *dice = substream->private_data; - struct snd_pcm_substream *pcm; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pcm = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pcm = NULL; - break; - default: - return -EINVAL; - } - amdtp_stream_pcm_trigger(&dice->rx_stream, pcm); - - return 0; -} - -static snd_pcm_uframes_t dice_pointer(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - return amdtp_stream_pcm_pointer(&dice->rx_stream); -} - -static int dice_create_pcm(struct snd_dice *dice) -{ - static struct snd_pcm_ops ops = { - .open = dice_open, - .close = dice_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dice_hw_params, - .hw_free = dice_hw_free, - .prepare = dice_prepare, - .trigger = dice_trigger, - .pointer = dice_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, - }; - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); - if (err < 0) - return err; - pcm->private_data = dice; - strcpy(pcm->name, dice->card->shortname); - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->ops = &ops; - - return 0; -} - -static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, - long count, loff_t *offset) -{ - struct snd_dice *dice = hwdep->private_data; - DEFINE_WAIT(wait); - union snd_firewire_event event; - - spin_lock_irq(&dice->lock); - - while (!dice->dev_lock_changed && dice->notification_bits == 0) { - prepare_to_wait(&dice->hwdep_wait, &wait, TASK_INTERRUPTIBLE); - spin_unlock_irq(&dice->lock); - schedule(); - finish_wait(&dice->hwdep_wait, &wait); - if (signal_pending(current)) - return -ERESTARTSYS; - spin_lock_irq(&dice->lock); - } - - memset(&event, 0, sizeof(event)); - if (dice->dev_lock_changed) { - event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; - event.lock_status.status = dice->dev_lock_count > 0; - dice->dev_lock_changed = false; - - count = min_t(long, count, sizeof(event.lock_status)); - } else { - event.dice_notification.type = - SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; - event.dice_notification.notification = dice->notification_bits; - dice->notification_bits = 0; - - count = min_t(long, count, sizeof(event.dice_notification)); - } - - spin_unlock_irq(&dice->lock); - - if (copy_to_user(buf, &event, count)) - return -EFAULT; - - return count; -} - -static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, - poll_table *wait) -{ - struct snd_dice *dice = hwdep->private_data; - unsigned int events; - - poll_wait(file, &dice->hwdep_wait, wait); - - spin_lock_irq(&dice->lock); - if (dice->dev_lock_changed || dice->notification_bits != 0) - events = POLLIN | POLLRDNORM; - else - events = 0; - spin_unlock_irq(&dice->lock); - - return events; -} - -static int dice_hwdep_get_info(struct snd_dice *dice, void __user *arg) -{ - struct fw_device *dev = fw_parent_device(dice->unit); - struct snd_firewire_get_info info; - - memset(&info, 0, sizeof(info)); - info.type = SNDRV_FIREWIRE_TYPE_DICE; - info.card = dev->card->index; - *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); - *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), - sizeof(info.device_name)); - - if (copy_to_user(arg, &info, sizeof(info))) - return -EFAULT; - - return 0; -} - -static int dice_hwdep_lock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count == 0) { - dice->dev_lock_count = -1; - err = 0; - } else { - err = -EBUSY; - } - - spin_unlock_irq(&dice->lock); - - return err; -} - -static int dice_hwdep_unlock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count == -1) { - dice->dev_lock_count = 0; - err = 0; - } else { - err = -EBADFD; - } - - spin_unlock_irq(&dice->lock); - - return err; -} - -static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) -{ - struct snd_dice *dice = hwdep->private_data; - - spin_lock_irq(&dice->lock); - if (dice->dev_lock_count == -1) - dice->dev_lock_count = 0; - spin_unlock_irq(&dice->lock); - - return 0; -} - -static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct snd_dice *dice = hwdep->private_data; - - switch (cmd) { - case SNDRV_FIREWIRE_IOCTL_GET_INFO: - return dice_hwdep_get_info(dice, (void __user *)arg); - case SNDRV_FIREWIRE_IOCTL_LOCK: - return dice_hwdep_lock(dice); - case SNDRV_FIREWIRE_IOCTL_UNLOCK: - return dice_hwdep_unlock(dice); - default: - return -ENOIOCTLCMD; - } -} - -#ifdef CONFIG_COMPAT -static int dice_hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, - unsigned int cmd, unsigned long arg) -{ - return dice_hwdep_ioctl(hwdep, file, cmd, - (unsigned long)compat_ptr(arg)); -} -#else -#define dice_hwdep_compat_ioctl NULL -#endif - -static int dice_create_hwdep(struct snd_dice *dice) -{ - static const struct snd_hwdep_ops ops = { - .read = dice_hwdep_read, - .release = dice_hwdep_release, - .poll = dice_hwdep_poll, - .ioctl = dice_hwdep_ioctl, - .ioctl_compat = dice_hwdep_compat_ioctl, - }; - struct snd_hwdep *hwdep; - int err; - - err = snd_hwdep_new(dice->card, "DICE", 0, &hwdep); - if (err < 0) - return err; - strcpy(hwdep->name, "DICE"); - hwdep->iface = SNDRV_HWDEP_IFACE_FW_DICE; - hwdep->ops = ops; - hwdep->private_data = dice; - hwdep->exclusive = true; - - return 0; -} - -static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, - unsigned int offset_q, unsigned int quadlets) -{ - unsigned int i; - int err; - - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE + 4 * offset_q, - buffer, 4 * quadlets, 0); - if (err < 0) - return err; - - for (i = 0; i < quadlets; ++i) - be32_to_cpus(&((u32 *)buffer)[i]); - - return 0; -} - -static const char *str_from_array(const char *const strs[], unsigned int count, - unsigned int i) -{ - if (i < count) - return strs[i]; - - return "(unknown)"; -} - -static void dice_proc_fixup_string(char *s, unsigned int size) -{ - unsigned int i; - - for (i = 0; i < size; i += 4) - cpu_to_le32s((u32 *)(s + i)); - - for (i = 0; i < size - 2; ++i) { - if (s[i] == '\0') - return; - if (s[i] == '\\' && s[i + 1] == '\\') { - s[i + 2] = '\0'; - return; - } - } - s[size - 1] = '\0'; -} - -static void dice_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - static const char *const section_names[5] = { - "global", "tx", "rx", "ext_sync", "unused2" - }; - static const char *const clock_sources[] = { - "aes1", "aes2", "aes3", "aes4", "aes", "adat", "tdif", - "wc", "arx1", "arx2", "arx3", "arx4", "internal" - }; - static const char *const rates[] = { - "32000", "44100", "48000", "88200", "96000", "176400", "192000", - "any low", "any mid", "any high", "none" - }; - struct snd_dice *dice = entry->private_data; - u32 sections[ARRAY_SIZE(section_names) * 2]; - struct { - u32 number; - u32 size; - } tx_rx_header; - union { - struct { - u32 owner_hi, owner_lo; - u32 notification; - char nick_name[NICK_NAME_SIZE]; - u32 clock_select; - u32 enable; - u32 status; - u32 extended_status; - u32 sample_rate; - u32 version; - u32 clock_caps; - char clock_source_names[CLOCK_SOURCE_NAMES_SIZE]; - } global; - struct { - u32 iso; - u32 number_audio; - u32 number_midi; - u32 speed; - char names[TX_NAMES_SIZE]; - u32 ac3_caps; - u32 ac3_enable; - } tx; - struct { - u32 iso; - u32 seq_start; - u32 number_audio; - u32 number_midi; - char names[RX_NAMES_SIZE]; - u32 ac3_caps; - u32 ac3_enable; - } rx; - struct { - u32 clock_source; - u32 locked; - u32 rate; - u32 adat_user_data; - } ext_sync; - } buf; - unsigned int quadlets, stream, i; - - if (dice_proc_read_mem(dice, sections, 0, ARRAY_SIZE(sections)) < 0) - return; - snd_iprintf(buffer, "sections:\n"); - for (i = 0; i < ARRAY_SIZE(section_names); ++i) - snd_iprintf(buffer, " %s: offset %u, size %u\n", - section_names[i], - sections[i * 2], sections[i * 2 + 1]); - - quadlets = min_t(u32, sections[1], sizeof(buf.global) / 4); - if (dice_proc_read_mem(dice, &buf.global, sections[0], quadlets) < 0) - return; - snd_iprintf(buffer, "global:\n"); - snd_iprintf(buffer, " owner: %04x:%04x%08x\n", - buf.global.owner_hi >> 16, - buf.global.owner_hi & 0xffff, buf.global.owner_lo); - snd_iprintf(buffer, " notification: %08x\n", buf.global.notification); - dice_proc_fixup_string(buf.global.nick_name, NICK_NAME_SIZE); - snd_iprintf(buffer, " nick name: %s\n", buf.global.nick_name); - snd_iprintf(buffer, " clock select: %s %s\n", - str_from_array(clock_sources, ARRAY_SIZE(clock_sources), - buf.global.clock_select & CLOCK_SOURCE_MASK), - str_from_array(rates, ARRAY_SIZE(rates), - (buf.global.clock_select & CLOCK_RATE_MASK) - >> CLOCK_RATE_SHIFT)); - snd_iprintf(buffer, " enable: %u\n", buf.global.enable); - snd_iprintf(buffer, " status: %slocked %s\n", - buf.global.status & STATUS_SOURCE_LOCKED ? "" : "un", - str_from_array(rates, ARRAY_SIZE(rates), - (buf.global.status & - STATUS_NOMINAL_RATE_MASK) - >> CLOCK_RATE_SHIFT)); - snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); - snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); - if (quadlets >= 90) { - snd_iprintf(buffer, " clock caps:"); - for (i = 0; i <= 6; ++i) - if (buf.global.clock_caps & (1 << i)) - snd_iprintf(buffer, " %s", rates[i]); - for (i = 0; i <= 12; ++i) - if (buf.global.clock_caps & (1 << (16 + i))) - snd_iprintf(buffer, " %s", clock_sources[i]); - snd_iprintf(buffer, "\n"); - dice_proc_fixup_string(buf.global.clock_source_names, - CLOCK_SOURCE_NAMES_SIZE); - snd_iprintf(buffer, " clock source names: %s\n", - buf.global.clock_source_names); - } - - if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) - return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); - for (stream = 0; stream < tx_rx_header.number; ++stream) { - if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + - stream * tx_rx_header.size, - quadlets) < 0) - break; - snd_iprintf(buffer, "tx %u:\n", stream); - snd_iprintf(buffer, " iso channel: %d\n", (int)buf.tx.iso); - snd_iprintf(buffer, " audio channels: %u\n", - buf.tx.number_audio); - snd_iprintf(buffer, " midi ports: %u\n", buf.tx.number_midi); - snd_iprintf(buffer, " speed: S%u\n", 100u << buf.tx.speed); - if (quadlets >= 68) { - dice_proc_fixup_string(buf.tx.names, TX_NAMES_SIZE); - snd_iprintf(buffer, " names: %s\n", buf.tx.names); - } - if (quadlets >= 70) { - snd_iprintf(buffer, " ac3 caps: %08x\n", - buf.tx.ac3_caps); - snd_iprintf(buffer, " ac3 enable: %08x\n", - buf.tx.ac3_enable); - } - } - - if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) - return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); - for (stream = 0; stream < tx_rx_header.number; ++stream) { - if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + - stream * tx_rx_header.size, - quadlets) < 0) - break; - snd_iprintf(buffer, "rx %u:\n", stream); - snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); - snd_iprintf(buffer, " audio channels: %u\n", - buf.rx.number_audio); - snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); - if (quadlets >= 68) { - dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); - snd_iprintf(buffer, " names: %s\n", buf.rx.names); - } - if (quadlets >= 70) { - snd_iprintf(buffer, " ac3 caps: %08x\n", - buf.rx.ac3_caps); - snd_iprintf(buffer, " ac3 enable: %08x\n", - buf.rx.ac3_enable); - } - } - - quadlets = min_t(u32, sections[7], sizeof(buf.ext_sync) / 4); - if (quadlets >= 4) { - if (dice_proc_read_mem(dice, &buf.ext_sync, - sections[6], 4) < 0) - return; - snd_iprintf(buffer, "ext status:\n"); - snd_iprintf(buffer, " clock source: %s\n", - str_from_array(clock_sources, - ARRAY_SIZE(clock_sources), - buf.ext_sync.clock_source)); - snd_iprintf(buffer, " locked: %u\n", buf.ext_sync.locked); - snd_iprintf(buffer, " rate: %s\n", - str_from_array(rates, ARRAY_SIZE(rates), - buf.ext_sync.rate)); - snd_iprintf(buffer, " adat user data: "); - if (buf.ext_sync.adat_user_data & ADAT_USER_DATA_NO_DATA) - snd_iprintf(buffer, "-\n"); - else - snd_iprintf(buffer, "%x\n", - buf.ext_sync.adat_user_data); - } -} - -static void dice_create_proc(struct snd_dice *dice) -{ - struct snd_info_entry *entry; - - if (!snd_card_proc_new(dice->card, "dice", &entry)) - snd_info_set_text_ops(entry, dice, dice_proc_read); -} - -static void dice_card_free(struct snd_card *card) -{ - struct snd_dice *dice = card->private_data; - - amdtp_stream_destroy(&dice->rx_stream); - fw_core_remove_address_handler(&dice->notification_handler); - mutex_destroy(&dice->mutex); -} - -#define OUI_WEISS 0x001c6a - -#define DICE_CATEGORY_ID 0x04 -#define WEISS_CATEGORY_ID 0x00 - -static int dice_interface_check(struct fw_unit *unit) -{ - static const int min_values[10] = { - 10, 0x64 / 4, - 10, 0x18 / 4, - 10, 0x18 / 4, - 0, 0, - 0, 0, - }; - struct fw_device *device = fw_parent_device(unit); - struct fw_csr_iterator it; - int key, value, vendor = -1, model = -1, err; - unsigned int category, i; - __be32 pointers[ARRAY_SIZE(min_values)]; - __be32 tx_data[4]; - __be32 version; - - /* - * Check that GUID and unit directory are constructed according to DICE - * rules, i.e., that the specifier ID is the GUID's OUI, and that the - * GUID chip ID consists of the 8-bit category ID, the 10-bit product - * ID, and a 22-bit serial number. - */ - fw_csr_iterator_init(&it, unit->directory); - while (fw_csr_iterator_next(&it, &key, &value)) { - switch (key) { - case CSR_SPECIFIER_ID: - vendor = value; - break; - case CSR_MODEL: - model = value; - break; - } - } - if (vendor == OUI_WEISS) - category = WEISS_CATEGORY_ID; - else - category = DICE_CATEGORY_ID; - if (device->config_rom[3] != ((vendor << 8) | category) || - device->config_rom[4] >> 22 != model) - return -ENODEV; - - /* - * Check that the sub address spaces exist and are located inside the - * private address space. The minimum values are chosen so that all - * minimally required registers are included. - */ - err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE, - pointers, sizeof(pointers), 0); - if (err < 0) - return -ENODEV; - for (i = 0; i < ARRAY_SIZE(pointers); ++i) { - value = be32_to_cpu(pointers[i]); - if (value < min_values[i] || value >= 0x40000) - return -ENODEV; - } - - /* We support playback only. Let capture devices be handled by FFADO. */ - err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[2]) * 4, - tx_data, sizeof(tx_data), 0); - if (err < 0 || (tx_data[0] && tx_data[3])) - return -ENODEV; - - /* - * Check that the implemented DICE driver specification major version - * number matches. - */ - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, - &version, 4, 0); - if (err < 0) - return -ENODEV; - if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { - dev_err(&unit->device, - "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - return -ENODEV; - } - - return 0; -} - -static int highest_supported_mode_rate(struct snd_dice *dice, unsigned int mode) -{ - int i; - - for (i = ARRAY_SIZE(dice_rates) - 1; i >= 0; --i) - if ((dice->clock_caps & (1 << i)) && - rate_index_to_mode(i) == mode) - return i; - - return -1; -} - -static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) -{ - __be32 values[2]; - int rate_index, err; - - rate_index = highest_supported_mode_rate(dice, mode); - if (rate_index < 0) { - dice->rx_channels[mode] = 0; - dice->rx_midi_ports[mode] = 0; - return 0; - } - - err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); - if (err < 0) - return err; - - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - rx_address(dice, RX_NUMBER_AUDIO), - values, 2 * 4, 0); - if (err < 0) - return err; - - dice->rx_channels[mode] = be32_to_cpu(values[0]); - dice->rx_midi_ports[mode] = be32_to_cpu(values[1]); - - return 0; -} - -static int dice_read_params(struct snd_dice *dice) -{ - __be32 pointers[6]; - __be32 value; - int mode, err; - - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE, - pointers, sizeof(pointers), 0); - if (err < 0) - return err; - - dice->global_offset = be32_to_cpu(pointers[0]) * 4; - dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - - /* some very old firmwares don't tell about their clock support */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) { - err = snd_fw_transaction( - dice->unit, TCODE_READ_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_CAPABILITIES), - &value, 4, 0); - if (err < 0) - return err; - dice->clock_caps = be32_to_cpu(value); - } else { - /* this should be supported by any device */ - dice->clock_caps = CLOCK_CAP_RATE_44100 | - CLOCK_CAP_RATE_48000 | - CLOCK_CAP_SOURCE_ARX1 | - CLOCK_CAP_SOURCE_INTERNAL; - } - - for (mode = 2; mode >= 0; --mode) { - err = dice_read_mode_params(dice, mode); - if (err < 0) - return err; - } - - return 0; -} - -static void dice_card_strings(struct snd_dice *dice) -{ - struct snd_card *card = dice->card; - struct fw_device *dev = fw_parent_device(dice->unit); - char vendor[32], model[32]; - unsigned int i; - int err; - - strcpy(card->driver, "DICE"); - - strcpy(card->shortname, "DICE"); - BUILD_BUG_ON(NICK_NAME_SIZE < sizeof(card->shortname)); - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - global_address(dice, GLOBAL_NICK_NAME), - card->shortname, sizeof(card->shortname), 0); - if (err >= 0) { - /* DICE strings are returned in "always-wrong" endianness */ - BUILD_BUG_ON(sizeof(card->shortname) % 4 != 0); - for (i = 0; i < sizeof(card->shortname); i += 4) - swab32s((u32 *)&card->shortname[i]); - card->shortname[sizeof(card->shortname) - 1] = '\0'; - } - - strcpy(vendor, "?"); - fw_csr_string(dev->config_rom + 5, CSR_VENDOR, vendor, sizeof(vendor)); - strcpy(model, "?"); - fw_csr_string(dice->unit->directory, CSR_MODEL, model, sizeof(model)); - snprintf(card->longname, sizeof(card->longname), - "%s %s (serial %u) at %s, S%d", - vendor, model, dev->config_rom[4] & 0x3fffff, - dev_name(&dice->unit->device), 100 << dev->max_speed); - - strcpy(card->mixername, "DICE"); -} - -static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) -{ - struct snd_card *card; - struct snd_dice *dice; - __be32 clock_sel; - int err; - - err = dice_interface_check(unit); - if (err < 0) - return err; - - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*dice), &card); - if (err < 0) - return err; - - dice = card->private_data; - dice->card = card; - spin_lock_init(&dice->lock); - mutex_init(&dice->mutex); - dice->unit = unit; - init_completion(&dice->clock_accepted); - init_waitqueue_head(&dice->hwdep_wait); - - dice->notification_handler.length = 4; - dice->notification_handler.address_callback = dice_notification; - dice->notification_handler.callback_data = dice; - err = fw_core_add_address_handler(&dice->notification_handler, - &fw_high_memory_region); - if (err < 0) - goto err_mutex; - - err = dice_owner_set(dice); - if (err < 0) - goto err_notification_handler; - - err = dice_read_params(dice); - if (err < 0) - goto err_owner; - - err = fw_iso_resources_init(&dice->rx_resources, unit); - if (err < 0) - goto err_owner; - dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; - - err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, - CIP_BLOCKING); - if (err < 0) - goto err_resources; - - card->private_free = dice_card_free; - - dice_card_strings(dice); - - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4, 0); - if (err < 0) - goto error; - clock_sel &= cpu_to_be32(~CLOCK_SOURCE_MASK); - clock_sel |= cpu_to_be32(CLOCK_SOURCE_ARX1); - err = snd_fw_transaction(unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4, 0); - if (err < 0) - goto error; - - err = dice_create_pcm(dice); - if (err < 0) - goto error; - - err = dice_create_hwdep(dice); - if (err < 0) - goto error; - - dice_create_proc(dice); - - err = snd_card_register(card); - if (err < 0) - goto error; - - dev_set_drvdata(&unit->device, dice); - - return 0; - -err_resources: - fw_iso_resources_destroy(&dice->rx_resources); -err_owner: - dice_owner_clear(dice); -err_notification_handler: - fw_core_remove_address_handler(&dice->notification_handler); -err_mutex: - mutex_destroy(&dice->mutex); -error: - snd_card_free(card); - return err; -} - -static void dice_remove(struct fw_unit *unit) -{ - struct snd_dice *dice = dev_get_drvdata(&unit->device); - - amdtp_stream_pcm_abort(&dice->rx_stream); - - snd_card_disconnect(dice->card); - - mutex_lock(&dice->mutex); - - dice_stream_stop(dice); - dice_owner_clear(dice); - - mutex_unlock(&dice->mutex); - - snd_card_free_when_closed(dice->card); -} - -static void dice_bus_reset(struct fw_unit *unit) -{ - struct snd_dice *dice = dev_get_drvdata(&unit->device); - - /* - * On a bus reset, the DICE firmware disables streaming and then goes - * off contemplating its own navel for hundreds of milliseconds before - * it can react to any of our attempts to reenable streaming. This - * means that we lose synchronization anyway, so we force our streams - * to stop so that the application can restart them in an orderly - * manner. - */ - amdtp_stream_pcm_abort(&dice->rx_stream); - - mutex_lock(&dice->mutex); - - dice->global_enabled = false; - dice_stream_stop_packets(dice); - - dice_owner_update(dice); - - fw_iso_resources_update(&dice->rx_resources); - - mutex_unlock(&dice->mutex); -} - -#define DICE_INTERFACE 0x000001 - -static const struct ieee1394_device_id dice_id_table[] = { - { - .match_flags = IEEE1394_MATCH_VERSION, - .version = DICE_INTERFACE, - }, - { } -}; -MODULE_DEVICE_TABLE(ieee1394, dice_id_table); - -static struct fw_driver dice_driver = { - .driver = { - .owner = THIS_MODULE, - .name = KBUILD_MODNAME, - .bus = &fw_bus_type, - }, - .probe = dice_probe, - .update = dice_bus_reset, - .remove = dice_remove, - .id_table = dice_id_table, -}; - -static int __init alsa_dice_init(void) -{ - return driver_register(&dice_driver.driver); -} - -static void __exit alsa_dice_exit(void) -{ - driver_unregister(&dice_driver.driver); -} - -module_init(alsa_dice_init); -module_exit(alsa_dice_exit); diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile new file mode 100644 index 0000000..af05d7e --- /dev/null +++ b/sound/firewire/dice/Makefile @@ -0,0 +1,2 @@ +snd-dice-objs := dice.o +obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h new file mode 100644 index 0000000..27b044f --- /dev/null +++ b/sound/firewire/dice/dice-interface.h @@ -0,0 +1,371 @@ +#ifndef SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED +#define SOUND_FIREWIRE_DICE_INTERFACE_H_INCLUDED + +/* + * DICE device interface definitions + */ + +/* + * Generally, all registers can be read like memory, i.e., with quadlet read or + * block read transactions with at least quadlet-aligned offset and length. + * Writes are not allowed except where noted; quadlet-sized registers must be + * written with a quadlet write transaction. + * + * All values are in big endian. The DICE firmware runs on a little-endian CPU + * and just byte-swaps _all_ quadlets on the bus, so values without endianness + * (e.g. strings) get scrambled and must be byte-swapped again by the driver. + */ + +/* + * Streaming is handled by the "DICE driver" interface. Its registers are + * located in this private address space. + */ +#define DICE_PRIVATE_SPACE 0xffffe0000000uLL + +/* + * The registers are organized in several sections, which are organized + * separately to allow them to be extended individually. Whether a register is + * supported can be detected by checking its offset against its section's size. + * + * The section offset values are relative to DICE_PRIVATE_SPACE; the offset/ + * size values are measured in quadlets. Read-only. + */ +#define DICE_GLOBAL_OFFSET 0x00 +#define DICE_GLOBAL_SIZE 0x04 +#define DICE_TX_OFFSET 0x08 +#define DICE_TX_SIZE 0x0c +#define DICE_RX_OFFSET 0x10 +#define DICE_RX_SIZE 0x14 +#define DICE_EXT_SYNC_OFFSET 0x18 +#define DICE_EXT_SYNC_SIZE 0x1c +#define DICE_UNUSED2_OFFSET 0x20 +#define DICE_UNUSED2_SIZE 0x24 + +/* + * Global settings. + */ + +/* + * Stores the full 64-bit address (node ID and offset in the node's address + * space) where the device will send notifications. Must be changed with + * a compare/swap transaction by the owner. This register is automatically + * cleared on a bus reset. + */ +#define GLOBAL_OWNER 0x000 +#define OWNER_NO_OWNER 0xffff000000000000uLL +#define OWNER_NODE_SHIFT 48 + +/* + * A bitmask with asynchronous events; read-only. When any event(s) happen, + * the bits of previous events are cleared, and the value of this register is + * also written to the address stored in the owner register. + */ +#define GLOBAL_NOTIFICATION 0x008 +/* Some registers in the Rx/Tx sections may have changed. */ +#define NOTIFY_RX_CFG_CHG 0x00000001 +#define NOTIFY_TX_CFG_CHG 0x00000002 +/* Lock status of the current clock source may have changed. */ +#define NOTIFY_LOCK_CHG 0x00000010 +/* Write to the clock select register has been finished. */ +#define NOTIFY_CLOCK_ACCEPTED 0x00000020 +/* Lock status of some clock source has changed. */ +#define NOTIFY_EXT_STATUS 0x00000040 +/* Other bits may be used for device-specific events. */ + +/* + * A name that can be customized for each device; read/write. Padded with zero + * bytes. Quadlets are byte-swapped. The encoding is whatever the host driver + * happens to be using. + */ +#define GLOBAL_NICK_NAME 0x00c +#define NICK_NAME_SIZE 64 + +/* + * The current sample rate and clock source; read/write. Whether a clock + * source or sample rate is supported is device-specific; the internal clock + * source is always available. Low/mid/high = up to 48/96/192 kHz. This + * register can be changed even while streams are running. + */ +#define GLOBAL_CLOCK_SELECT 0x04c +#define CLOCK_SOURCE_MASK 0x000000ff +#define CLOCK_SOURCE_AES1 0x00000000 +#define CLOCK_SOURCE_AES2 0x00000001 +#define CLOCK_SOURCE_AES3 0x00000002 +#define CLOCK_SOURCE_AES4 0x00000003 +#define CLOCK_SOURCE_AES_ANY 0x00000004 +#define CLOCK_SOURCE_ADAT 0x00000005 +#define CLOCK_SOURCE_TDIF 0x00000006 +#define CLOCK_SOURCE_WC 0x00000007 +#define CLOCK_SOURCE_ARX1 0x00000008 +#define CLOCK_SOURCE_ARX2 0x00000009 +#define CLOCK_SOURCE_ARX3 0x0000000a +#define CLOCK_SOURCE_ARX4 0x0000000b +#define CLOCK_SOURCE_INTERNAL 0x0000000c +#define CLOCK_RATE_MASK 0x0000ff00 +#define CLOCK_RATE_32000 0x00000000 +#define CLOCK_RATE_44100 0x00000100 +#define CLOCK_RATE_48000 0x00000200 +#define CLOCK_RATE_88200 0x00000300 +#define CLOCK_RATE_96000 0x00000400 +#define CLOCK_RATE_176400 0x00000500 +#define CLOCK_RATE_192000 0x00000600 +#define CLOCK_RATE_ANY_LOW 0x00000700 +#define CLOCK_RATE_ANY_MID 0x00000800 +#define CLOCK_RATE_ANY_HIGH 0x00000900 +#define CLOCK_RATE_NONE 0x00000a00 +#define CLOCK_RATE_SHIFT 8 + +/* + * Enable streaming; read/write. Writing a non-zero value (re)starts all + * streams that have a valid iso channel set; zero stops all streams. The + * streams' parameters must be configured before starting. This register is + * automatically cleared on a bus reset. + */ +#define GLOBAL_ENABLE 0x050 + +/* + * Status of the sample clock; read-only. + */ +#define GLOBAL_STATUS 0x054 +/* The current clock source is locked. */ +#define STATUS_SOURCE_LOCKED 0x00000001 +/* The actual sample rate; CLOCK_RATE_32000-_192000 or _NONE. */ +#define STATUS_NOMINAL_RATE_MASK 0x0000ff00 + +/* + * Status of all clock sources; read-only. + */ +#define GLOBAL_EXTENDED_STATUS 0x058 +/* + * The _LOCKED bits always show the current status; any change generates + * a notification. + */ +#define EXT_STATUS_AES1_LOCKED 0x00000001 +#define EXT_STATUS_AES2_LOCKED 0x00000002 +#define EXT_STATUS_AES3_LOCKED 0x00000004 +#define EXT_STATUS_AES4_LOCKED 0x00000008 +#define EXT_STATUS_ADAT_LOCKED 0x00000010 +#define EXT_STATUS_TDIF_LOCKED 0x00000020 +#define EXT_STATUS_ARX1_LOCKED 0x00000040 +#define EXT_STATUS_ARX2_LOCKED 0x00000080 +#define EXT_STATUS_ARX3_LOCKED 0x00000100 +#define EXT_STATUS_ARX4_LOCKED 0x00000200 +#define EXT_STATUS_WC_LOCKED 0x00000400 +/* + * The _SLIP bits do not generate notifications; a set bit indicates that an + * error occurred since the last time when this register was read with + * a quadlet read transaction. + */ +#define EXT_STATUS_AES1_SLIP 0x00010000 +#define EXT_STATUS_AES2_SLIP 0x00020000 +#define EXT_STATUS_AES3_SLIP 0x00040000 +#define EXT_STATUS_AES4_SLIP 0x00080000 +#define EXT_STATUS_ADAT_SLIP 0x00100000 +#define EXT_STATUS_TDIF_SLIP 0x00200000 +#define EXT_STATUS_ARX1_SLIP 0x00400000 +#define EXT_STATUS_ARX2_SLIP 0x00800000 +#define EXT_STATUS_ARX3_SLIP 0x01000000 +#define EXT_STATUS_ARX4_SLIP 0x02000000 +#define EXT_STATUS_WC_SLIP 0x04000000 + +/* + * The measured rate of the current clock source, in Hz; read-only. + */ +#define GLOBAL_SAMPLE_RATE 0x05c + +/* + * The version of the DICE driver specification that this device conforms to; + * read-only. + */ +#define GLOBAL_VERSION 0x060 + +/* Some old firmware versions do not have the following global registers: */ + +/* + * Supported sample rates and clock sources; read-only. + */ +#define GLOBAL_CLOCK_CAPABILITIES 0x064 +#define CLOCK_CAP_RATE_32000 0x00000001 +#define CLOCK_CAP_RATE_44100 0x00000002 +#define CLOCK_CAP_RATE_48000 0x00000004 +#define CLOCK_CAP_RATE_88200 0x00000008 +#define CLOCK_CAP_RATE_96000 0x00000010 +#define CLOCK_CAP_RATE_176400 0x00000020 +#define CLOCK_CAP_RATE_192000 0x00000040 +#define CLOCK_CAP_SOURCE_AES1 0x00010000 +#define CLOCK_CAP_SOURCE_AES2 0x00020000 +#define CLOCK_CAP_SOURCE_AES3 0x00040000 +#define CLOCK_CAP_SOURCE_AES4 0x00080000 +#define CLOCK_CAP_SOURCE_AES_ANY 0x00100000 +#define CLOCK_CAP_SOURCE_ADAT 0x00200000 +#define CLOCK_CAP_SOURCE_TDIF 0x00400000 +#define CLOCK_CAP_SOURCE_WC 0x00800000 +#define CLOCK_CAP_SOURCE_ARX1 0x01000000 +#define CLOCK_CAP_SOURCE_ARX2 0x02000000 +#define CLOCK_CAP_SOURCE_ARX3 0x04000000 +#define CLOCK_CAP_SOURCE_ARX4 0x08000000 +#define CLOCK_CAP_SOURCE_INTERNAL 0x10000000 + +/* + * Names of all clock sources; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. Unused clock sources are included. + */ +#define GLOBAL_CLOCK_SOURCE_NAMES 0x068 +#define CLOCK_SOURCE_NAMES_SIZE 256 + +/* + * Capture stream settings. This section includes the number/size registers + * and the registers of all streams. + */ + +/* + * The number of supported capture streams; read-only. + */ +#define TX_NUMBER 0x000 + +/* + * The size of one stream's register block, in quadlets; read-only. The + * registers of the first stream follow immediately afterwards; the registers + * of the following streams are offset by this register's value. + */ +#define TX_SIZE 0x004 + +/* + * The isochronous channel number on which packets are sent, or -1 if the + * stream is not to be used; read/write. + */ +#define TX_ISOCHRONOUS 0x008 + +/* + * The number of audio channels; read-only. There will be one quadlet per + * channel; the first channel is the first quadlet in a data block. + */ +#define TX_NUMBER_AUDIO 0x00c + +/* + * The number of MIDI ports, 0-8; read-only. If > 0, there will be one + * additional quadlet in each data block, following the audio quadlets. + */ +#define TX_NUMBER_MIDI 0x010 + +/* + * The speed at which the packets are sent, SCODE_100-_400; read/write. + */ +#define TX_SPEED 0x014 + +/* + * Names of all audio channels; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. + */ +#define TX_NAMES 0x018 +#define TX_NAMES_SIZE 256 + +/* + * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio + * channel. + */ +#define TX_AC3_CAPABILITIES 0x118 + +/* + * Send audio data with IEC60958 label; read/write. Bitmask with one bit per + * audio channel. This register can be changed even while the stream is + * running. + */ +#define TX_AC3_ENABLE 0x11c + +/* + * Playback stream settings. This section includes the number/size registers + * and the registers of all streams. + */ + +/* + * The number of supported playback streams; read-only. + */ +#define RX_NUMBER 0x000 + +/* + * The size of one stream's register block, in quadlets; read-only. The + * registers of the first stream follow immediately afterwards; the registers + * of the following streams are offset by this register's value. + */ +#define RX_SIZE 0x004 + +/* + * The isochronous channel number on which packets are received, or -1 if the + * stream is not to be used; read/write. + */ +#define RX_ISOCHRONOUS 0x008 + +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x00c + +/* + * The number of audio channels; read-only. There will be one quadlet per + * channel. + */ +#define RX_NUMBER_AUDIO 0x010 + +/* + * The number of MIDI ports, 0-8; read-only. If > 0, there will be one + * additional quadlet in each data block, following the audio quadlets. + */ +#define RX_NUMBER_MIDI 0x014 + +/* + * Names of all audio channels; read-only. Quadlets are byte-swapped. Names + * are separated with one backslash, the list is terminated with two + * backslashes. + */ +#define RX_NAMES 0x018 +#define RX_NAMES_SIZE 256 + +/* + * Audio IEC60958 capabilities; read-only. Bitmask with one bit per audio + * channel. + */ +#define RX_AC3_CAPABILITIES 0x118 + +/* + * Receive audio data with IEC60958 label; read/write. Bitmask with one bit + * per audio channel. This register can be changed even while the stream is + * running. + */ +#define RX_AC3_ENABLE 0x11c + +/* + * Extended synchronization information. + * This section can be read completely with a block read request. + */ + +/* + * Current clock source; read-only. + */ +#define EXT_SYNC_CLOCK_SOURCE 0x000 + +/* + * Clock source is locked (boolean); read-only. + */ +#define EXT_SYNC_LOCKED 0x004 + +/* + * Current sample rate (CLOCK_RATE_* >> CLOCK_RATE_SHIFT), _32000-_192000 or + * _NONE; read-only. + */ +#define EXT_SYNC_RATE 0x008 + +/* + * ADAT user data bits; read-only. + */ +#define EXT_SYNC_ADAT_USER_DATA 0x00c +/* The data bits, if available. */ +#define ADAT_USER_DATA_MASK 0x0f +/* The data bits are not available. */ +#define ADAT_USER_DATA_NO_DATA 0x10 + +#endif diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c new file mode 100644 index 0000000..d3ec778 --- /dev/null +++ b/sound/firewire/dice/dice.c @@ -0,0 +1,1512 @@ +/* + * TC Applied Technologies Digital Interface Communications Engine driver + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../amdtp.h" +#include "../iso-resources.h" +#include "../lib.h" +#include "dice-interface.h" + + +struct snd_dice { + struct snd_card *card; + struct fw_unit *unit; + spinlock_t lock; + struct mutex mutex; + unsigned int global_offset; + unsigned int rx_offset; + unsigned int clock_caps; + unsigned int rx_channels[3]; + unsigned int rx_midi_ports[3]; + struct fw_address_handler notification_handler; + int owner_generation; + int dev_lock_count; /* > 0 driver, < 0 userspace */ + bool dev_lock_changed; + bool global_enabled; + struct completion clock_accepted; + wait_queue_head_t hwdep_wait; + u32 notification_bits; + struct fw_iso_resources rx_resources; + struct amdtp_stream rx_stream; +}; + +MODULE_DESCRIPTION("DICE driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); + +static const unsigned int dice_rates[] = { + /* mode 0 */ + [0] = 32000, + [1] = 44100, + [2] = 48000, + /* mode 1 */ + [3] = 88200, + [4] = 96000, + /* mode 2 */ + [5] = 176400, + [6] = 192000, +}; + +static unsigned int rate_to_index(unsigned int rate) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if (dice_rates[i] == rate) + return i; + + return 0; +} + +static unsigned int rate_index_to_mode(unsigned int rate_index) +{ + return ((int)rate_index - 1) / 2; +} + +static void dice_lock_changed(struct snd_dice *dice) +{ + dice->dev_lock_changed = true; + wake_up(&dice->hwdep_wait); +} + +static int dice_try_lock(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count < 0) { + err = -EBUSY; + goto out; + } + + if (dice->dev_lock_count++ == 0) + dice_lock_changed(dice); + err = 0; + +out: + spin_unlock_irq(&dice->lock); + + return err; +} + +static void dice_unlock(struct snd_dice *dice) +{ + spin_lock_irq(&dice->lock); + + if (WARN_ON(dice->dev_lock_count <= 0)) + goto out; + + if (--dice->dev_lock_count == 0) + dice_lock_changed(dice); + +out: + spin_unlock_irq(&dice->lock); +} + +static inline u64 global_address(struct snd_dice *dice, unsigned int offset) +{ + return DICE_PRIVATE_SPACE + dice->global_offset + offset; +} + +/* TODO: rx index */ +static inline u64 rx_address(struct snd_dice *dice, unsigned int offset) +{ + return DICE_PRIVATE_SPACE + dice->rx_offset + offset; +} + +static int dice_owner_set(struct snd_dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + int err, errors = 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + for (;;) { + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + err = snd_fw_transaction(dice->unit, + TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, + FW_FIXED_GENERATION | + dice->owner_generation); + + if (err == 0) { + if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { + dev_err(&dice->unit->device, + "device is already in use\n"); + err = -EBUSY; + } + break; + } + if (err != -EAGAIN || ++errors >= 3) + break; + + msleep(20); + } + + kfree(buffer); + + return err; +} + +static int dice_owner_update(struct snd_dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + int err; + + if (dice->owner_generation == -1) + return 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + err = snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, + FW_FIXED_GENERATION | dice->owner_generation); + + if (err == 0) { + if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { + dev_err(&dice->unit->device, + "device is already in use\n"); + err = -EBUSY; + } + } else if (err == -EAGAIN) { + err = 0; /* try again later */ + } + + kfree(buffer); + + if (err < 0) + dice->owner_generation = -1; + + return err; +} + +static void dice_owner_clear(struct snd_dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return; + + buffer[0] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + buffer[1] = cpu_to_be64(OWNER_NO_OWNER); + snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + global_address(dice, GLOBAL_OWNER), + buffer, 2 * 8, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); + + kfree(buffer); + + dice->owner_generation = -1; +} + +static int dice_enable_set(struct snd_dice *dice) +{ + __be32 value; + int err; + + value = cpu_to_be32(1); + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_ENABLE), + &value, 4, + FW_FIXED_GENERATION | dice->owner_generation); + if (err < 0) + return err; + + dice->global_enabled = true; + + return 0; +} + +static void dice_enable_clear(struct snd_dice *dice) +{ + __be32 value; + + if (!dice->global_enabled) + return; + + value = 0; + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_ENABLE), + &value, 4, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); + + dice->global_enabled = false; +} + +static void dice_notification(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct snd_dice *dice = callback_data; + u32 bits; + unsigned long flags; + + if (tcode != TCODE_WRITE_QUADLET_REQUEST) { + fw_send_response(card, request, RCODE_TYPE_ERROR); + return; + } + if ((offset & 3) != 0) { + fw_send_response(card, request, RCODE_ADDRESS_ERROR); + return; + } + + bits = be32_to_cpup(data); + + spin_lock_irqsave(&dice->lock, flags); + dice->notification_bits |= bits; + spin_unlock_irqrestore(&dice->lock, flags); + + fw_send_response(card, request, RCODE_COMPLETE); + + if (bits & NOTIFY_CLOCK_ACCEPTED) + complete(&dice->clock_accepted); + wake_up(&dice->hwdep_wait); +} + +static int dice_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_dice *dice = rule->private; + const struct snd_interval *channels = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval allowed_rates = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, mode; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) { + mode = rate_index_to_mode(i); + if ((dice->clock_caps & (1 << i)) && + snd_interval_test(channels, dice->rx_channels[mode])) { + allowed_rates.min = min(allowed_rates.min, + dice_rates[i]); + allowed_rates.max = max(allowed_rates.max, + dice_rates[i]); + } + } + + return snd_interval_refine(rate, &allowed_rates); +} + +static int dice_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_dice *dice = rule->private; + const struct snd_interval *rate = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval allowed_channels = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, mode; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if ((dice->clock_caps & (1 << i)) && + snd_interval_test(rate, dice_rates[i])) { + mode = rate_index_to_mode(i); + allowed_channels.min = min(allowed_channels.min, + dice->rx_channels[mode]); + allowed_channels.max = max(allowed_channels.max, + dice->rx_channels[mode]); + } + + return snd_interval_refine(channels, &allowed_channels); +} + +static int dice_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = UINT_MAX, + .channels_max = 0, + .buffer_bytes_max = 16 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct snd_dice *dice = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int i; + int err; + + err = dice_try_lock(dice); + if (err < 0) + goto error; + + runtime->hw = hardware; + + for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + if (dice->clock_caps & (1 << i)) + runtime->hw.rates |= + snd_pcm_rate_to_rate_bit(dice_rates[i]); + snd_pcm_limit_hw_rates(runtime); + + for (i = 0; i < 3; ++i) + if (dice->rx_channels[i]) { + runtime->hw.channels_min = min(runtime->hw.channels_min, + dice->rx_channels[i]); + runtime->hw.channels_max = max(runtime->hw.channels_max, + dice->rx_channels[i]); + } + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + dice_rate_constraint, dice, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + goto err_lock; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + dice_channels_constraint, dice, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto err_lock; + + err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); + if (err < 0) + goto err_lock; + + return 0; + +err_lock: + dice_unlock(dice); +error: + return err; +} + +static int dice_close(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + dice_unlock(dice); + + return 0; +} + +static int dice_stream_start_packets(struct snd_dice *dice) +{ + int err; + + if (amdtp_stream_running(&dice->rx_stream)) + return 0; + + err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + return err; + + err = dice_enable_set(dice); + if (err < 0) { + amdtp_stream_stop(&dice->rx_stream); + return err; + } + + return 0; +} + +static int dice_stream_start(struct snd_dice *dice) +{ + __be32 channel; + int err; + + if (!dice->rx_resources.allocated) { + err = fw_iso_resources_allocate(&dice->rx_resources, + amdtp_stream_get_max_payload(&dice->rx_stream), + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + goto error; + + channel = cpu_to_be32(dice->rx_resources.channel); + err = snd_fw_transaction(dice->unit, + TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), + &channel, 4, 0); + if (err < 0) + goto err_resources; + } + + err = dice_stream_start_packets(dice); + if (err < 0) + goto err_rx_channel; + + return 0; + +err_rx_channel: + channel = cpu_to_be32((u32)-1); + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); +err_resources: + fw_iso_resources_free(&dice->rx_resources); +error: + return err; +} + +static void dice_stream_stop_packets(struct snd_dice *dice) +{ + if (amdtp_stream_running(&dice->rx_stream)) { + dice_enable_clear(dice); + amdtp_stream_stop(&dice->rx_stream); + } +} + +static void dice_stream_stop(struct snd_dice *dice) +{ + __be32 channel; + + dice_stream_stop_packets(dice); + + if (!dice->rx_resources.allocated) + return; + + channel = cpu_to_be32((u32)-1); + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); + + fw_iso_resources_free(&dice->rx_resources); +} + +static int dice_change_rate(struct snd_dice *dice, unsigned int clock_rate) +{ + __be32 value; + int err; + + reinit_completion(&dice->clock_accepted); + + value = cpu_to_be32(clock_rate | CLOCK_SOURCE_ARX1); + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &value, 4, 0); + if (err < 0) + return err; + + if (!wait_for_completion_timeout(&dice->clock_accepted, + msecs_to_jiffies(100))) + dev_warn(&dice->unit->device, "clock change timed out\n"); + + return 0; +} + +static int dice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_dice *dice = substream->private_data; + unsigned int rate_index, mode, rate, channels, i; + int err; + + mutex_lock(&dice->mutex); + dice_stream_stop(dice); + mutex_unlock(&dice->mutex); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + rate = params_rate(hw_params); + rate_index = rate_to_index(rate); + err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + if (err < 0) + return err; + + /* + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. + */ + channels = params_channels(hw_params); + if (rate_index > 4) { + if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { + err = -ENOSYS; + return err; + } + + rate /= 2; + channels *= 2; + dice->rx_stream.double_pcm_frames = true; + } else { + dice->rx_stream.double_pcm_frames = false; + } + + mode = rate_index_to_mode(rate_index); + amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, + dice->rx_midi_ports[mode]); + if (rate_index > 4) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->rx_stream.pcm_positions[i] = i * 2; + dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + + amdtp_stream_set_pcm_format(&dice->rx_stream, + params_format(hw_params)); + + return 0; +} + +static int dice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + mutex_lock(&dice->mutex); + dice_stream_stop(dice); + mutex_unlock(&dice->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int dice_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + int err; + + mutex_lock(&dice->mutex); + + if (amdtp_streaming_error(&dice->rx_stream)) + dice_stream_stop_packets(dice); + + err = dice_stream_start(dice); + if (err < 0) { + mutex_unlock(&dice->mutex); + return err; + } + + mutex_unlock(&dice->mutex); + + amdtp_stream_pcm_prepare(&dice->rx_stream); + + return 0; +} + +static int dice_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_dice *dice = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_stream_pcm_trigger(&dice->rx_stream, pcm); + + return 0; +} + +static snd_pcm_uframes_t dice_pointer(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + return amdtp_stream_pcm_pointer(&dice->rx_stream); +} + +static int dice_create_pcm(struct snd_dice *dice) +{ + static struct snd_pcm_ops ops = { + .open = dice_open, + .close = dice_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dice_hw_params, + .hw_free = dice_hw_free, + .prepare = dice_prepare, + .trigger = dice_trigger, + .pointer = dice_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = dice; + strcpy(pcm->name, dice->card->shortname); + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->ops = &ops; + + return 0; +} + +static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, + long count, loff_t *offset) +{ + struct snd_dice *dice = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&dice->lock); + + while (!dice->dev_lock_changed && dice->notification_bits == 0) { + prepare_to_wait(&dice->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&dice->lock); + schedule(); + finish_wait(&dice->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&dice->lock); + } + + memset(&event, 0, sizeof(event)); + if (dice->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = dice->dev_lock_count > 0; + dice->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + } else { + event.dice_notification.type = + SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; + event.dice_notification.notification = dice->notification_bits; + dice->notification_bits = 0; + + count = min_t(long, count, sizeof(event.dice_notification)); + } + + spin_unlock_irq(&dice->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_dice *dice = hwdep->private_data; + unsigned int events; + + poll_wait(file, &dice->hwdep_wait, wait); + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_changed || dice->notification_bits != 0) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&dice->lock); + + return events; +} + +static int dice_hwdep_get_info(struct snd_dice *dice, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(dice->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_DICE; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int dice_hwdep_lock(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == 0) { + dice->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&dice->lock); + + return err; +} + +static int dice_hwdep_unlock(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == -1) { + dice->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&dice->lock); + + return err; +} + +static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_dice *dice = hwdep->private_data; + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_count == -1) + dice->dev_lock_count = 0; + spin_unlock_irq(&dice->lock); + + return 0; +} + +static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_dice *dice = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return dice_hwdep_get_info(dice, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return dice_hwdep_lock(dice); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return dice_hwdep_unlock(dice); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int dice_hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return dice_hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define dice_hwdep_compat_ioctl NULL +#endif + +static int dice_create_hwdep(struct snd_dice *dice) +{ + static const struct snd_hwdep_ops ops = { + .read = dice_hwdep_read, + .release = dice_hwdep_release, + .poll = dice_hwdep_poll, + .ioctl = dice_hwdep_ioctl, + .ioctl_compat = dice_hwdep_compat_ioctl, + }; + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(dice->card, "DICE", 0, &hwdep); + if (err < 0) + return err; + strcpy(hwdep->name, "DICE"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_DICE; + hwdep->ops = ops; + hwdep->private_data = dice; + hwdep->exclusive = true; + + return 0; +} + +static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, + unsigned int offset_q, unsigned int quadlets) +{ + unsigned int i; + int err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE + 4 * offset_q, + buffer, 4 * quadlets, 0); + if (err < 0) + return err; + + for (i = 0; i < quadlets; ++i) + be32_to_cpus(&((u32 *)buffer)[i]); + + return 0; +} + +static const char *str_from_array(const char *const strs[], unsigned int count, + unsigned int i) +{ + if (i < count) + return strs[i]; + + return "(unknown)"; +} + +static void dice_proc_fixup_string(char *s, unsigned int size) +{ + unsigned int i; + + for (i = 0; i < size; i += 4) + cpu_to_le32s((u32 *)(s + i)); + + for (i = 0; i < size - 2; ++i) { + if (s[i] == '\0') + return; + if (s[i] == '\\' && s[i + 1] == '\\') { + s[i + 2] = '\0'; + return; + } + } + s[size - 1] = '\0'; +} + +static void dice_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + static const char *const section_names[5] = { + "global", "tx", "rx", "ext_sync", "unused2" + }; + static const char *const clock_sources[] = { + "aes1", "aes2", "aes3", "aes4", "aes", "adat", "tdif", + "wc", "arx1", "arx2", "arx3", "arx4", "internal" + }; + static const char *const rates[] = { + "32000", "44100", "48000", "88200", "96000", "176400", "192000", + "any low", "any mid", "any high", "none" + }; + struct snd_dice *dice = entry->private_data; + u32 sections[ARRAY_SIZE(section_names) * 2]; + struct { + u32 number; + u32 size; + } tx_rx_header; + union { + struct { + u32 owner_hi, owner_lo; + u32 notification; + char nick_name[NICK_NAME_SIZE]; + u32 clock_select; + u32 enable; + u32 status; + u32 extended_status; + u32 sample_rate; + u32 version; + u32 clock_caps; + char clock_source_names[CLOCK_SOURCE_NAMES_SIZE]; + } global; + struct { + u32 iso; + u32 number_audio; + u32 number_midi; + u32 speed; + char names[TX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } tx; + struct { + u32 iso; + u32 seq_start; + u32 number_audio; + u32 number_midi; + char names[RX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } rx; + struct { + u32 clock_source; + u32 locked; + u32 rate; + u32 adat_user_data; + } ext_sync; + } buf; + unsigned int quadlets, stream, i; + + if (dice_proc_read_mem(dice, sections, 0, ARRAY_SIZE(sections)) < 0) + return; + snd_iprintf(buffer, "sections:\n"); + for (i = 0; i < ARRAY_SIZE(section_names); ++i) + snd_iprintf(buffer, " %s: offset %u, size %u\n", + section_names[i], + sections[i * 2], sections[i * 2 + 1]); + + quadlets = min_t(u32, sections[1], sizeof(buf.global) / 4); + if (dice_proc_read_mem(dice, &buf.global, sections[0], quadlets) < 0) + return; + snd_iprintf(buffer, "global:\n"); + snd_iprintf(buffer, " owner: %04x:%04x%08x\n", + buf.global.owner_hi >> 16, + buf.global.owner_hi & 0xffff, buf.global.owner_lo); + snd_iprintf(buffer, " notification: %08x\n", buf.global.notification); + dice_proc_fixup_string(buf.global.nick_name, NICK_NAME_SIZE); + snd_iprintf(buffer, " nick name: %s\n", buf.global.nick_name); + snd_iprintf(buffer, " clock select: %s %s\n", + str_from_array(clock_sources, ARRAY_SIZE(clock_sources), + buf.global.clock_select & CLOCK_SOURCE_MASK), + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.clock_select & CLOCK_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " enable: %u\n", buf.global.enable); + snd_iprintf(buffer, " status: %slocked %s\n", + buf.global.status & STATUS_SOURCE_LOCKED ? "" : "un", + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.status & + STATUS_NOMINAL_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); + snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); + if (quadlets >= 90) { + snd_iprintf(buffer, " clock caps:"); + for (i = 0; i <= 6; ++i) + if (buf.global.clock_caps & (1 << i)) + snd_iprintf(buffer, " %s", rates[i]); + for (i = 0; i <= 12; ++i) + if (buf.global.clock_caps & (1 << (16 + i))) + snd_iprintf(buffer, " %s", clock_sources[i]); + snd_iprintf(buffer, "\n"); + dice_proc_fixup_string(buf.global.clock_source_names, + CLOCK_SOURCE_NAMES_SIZE); + snd_iprintf(buffer, " clock source names: %s\n", + buf.global.clock_source_names); + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "tx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.tx.iso); + snd_iprintf(buffer, " audio channels: %u\n", + buf.tx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.tx.number_midi); + snd_iprintf(buffer, " speed: S%u\n", 100u << buf.tx.speed); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.tx.names, TX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.tx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.tx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.tx.ac3_enable); + } + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "rx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); + snd_iprintf(buffer, " audio channels: %u\n", + buf.rx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.rx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.rx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.rx.ac3_enable); + } + } + + quadlets = min_t(u32, sections[7], sizeof(buf.ext_sync) / 4); + if (quadlets >= 4) { + if (dice_proc_read_mem(dice, &buf.ext_sync, + sections[6], 4) < 0) + return; + snd_iprintf(buffer, "ext status:\n"); + snd_iprintf(buffer, " clock source: %s\n", + str_from_array(clock_sources, + ARRAY_SIZE(clock_sources), + buf.ext_sync.clock_source)); + snd_iprintf(buffer, " locked: %u\n", buf.ext_sync.locked); + snd_iprintf(buffer, " rate: %s\n", + str_from_array(rates, ARRAY_SIZE(rates), + buf.ext_sync.rate)); + snd_iprintf(buffer, " adat user data: "); + if (buf.ext_sync.adat_user_data & ADAT_USER_DATA_NO_DATA) + snd_iprintf(buffer, "-\n"); + else + snd_iprintf(buffer, "%x\n", + buf.ext_sync.adat_user_data); + } +} + +static void dice_create_proc(struct snd_dice *dice) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(dice->card, "dice", &entry)) + snd_info_set_text_ops(entry, dice, dice_proc_read); +} + +static void dice_card_free(struct snd_card *card) +{ + struct snd_dice *dice = card->private_data; + + amdtp_stream_destroy(&dice->rx_stream); + fw_core_remove_address_handler(&dice->notification_handler); + mutex_destroy(&dice->mutex); +} + +#define OUI_WEISS 0x001c6a + +#define DICE_CATEGORY_ID 0x04 +#define WEISS_CATEGORY_ID 0x00 + +static int dice_interface_check(struct fw_unit *unit) +{ + static const int min_values[10] = { + 10, 0x64 / 4, + 10, 0x18 / 4, + 10, 0x18 / 4, + 0, 0, + 0, 0, + }; + struct fw_device *device = fw_parent_device(unit); + struct fw_csr_iterator it; + int key, value, vendor = -1, model = -1, err; + unsigned int category, i; + __be32 pointers[ARRAY_SIZE(min_values)]; + __be32 tx_data[4]; + __be32 version; + + /* + * Check that GUID and unit directory are constructed according to DICE + * rules, i.e., that the specifier ID is the GUID's OUI, and that the + * GUID chip ID consists of the 8-bit category ID, the 10-bit product + * ID, and a 22-bit serial number. + */ + fw_csr_iterator_init(&it, unit->directory); + while (fw_csr_iterator_next(&it, &key, &value)) { + switch (key) { + case CSR_SPECIFIER_ID: + vendor = value; + break; + case CSR_MODEL: + model = value; + break; + } + } + if (vendor == OUI_WEISS) + category = WEISS_CATEGORY_ID; + else + category = DICE_CATEGORY_ID; + if (device->config_rom[3] != ((vendor << 8) | category) || + device->config_rom[4] >> 22 != model) + return -ENODEV; + + /* + * Check that the sub address spaces exist and are located inside the + * private address space. The minimum values are chosen so that all + * minimally required registers are included. + */ + err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE, + pointers, sizeof(pointers), 0); + if (err < 0) + return -ENODEV; + for (i = 0; i < ARRAY_SIZE(pointers); ++i) { + value = be32_to_cpu(pointers[i]); + if (value < min_values[i] || value >= 0x40000) + return -ENODEV; + } + + /* We support playback only. Let capture devices be handled by FFADO. */ + err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[2]) * 4, + tx_data, sizeof(tx_data), 0); + if (err < 0 || (tx_data[0] && tx_data[3])) + return -ENODEV; + + /* + * Check that the implemented DICE driver specification major version + * number matches. + */ + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + DICE_PRIVATE_SPACE + + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, + &version, 4, 0); + if (err < 0) + return -ENODEV; + if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { + dev_err(&unit->device, + "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); + return -ENODEV; + } + + return 0; +} + +static int highest_supported_mode_rate(struct snd_dice *dice, unsigned int mode) +{ + int i; + + for (i = ARRAY_SIZE(dice_rates) - 1; i >= 0; --i) + if ((dice->clock_caps & (1 << i)) && + rate_index_to_mode(i) == mode) + return i; + + return -1; +} + +static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) +{ + __be32 values[2]; + int rate_index, err; + + rate_index = highest_supported_mode_rate(dice, mode); + if (rate_index < 0) { + dice->rx_channels[mode] = 0; + dice->rx_midi_ports[mode] = 0; + return 0; + } + + err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + if (err < 0) + return err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + rx_address(dice, RX_NUMBER_AUDIO), + values, 2 * 4, 0); + if (err < 0) + return err; + + dice->rx_channels[mode] = be32_to_cpu(values[0]); + dice->rx_midi_ports[mode] = be32_to_cpu(values[1]); + + return 0; +} + +static int dice_read_params(struct snd_dice *dice) +{ + __be32 pointers[6]; + __be32 value; + int mode, err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE, + pointers, sizeof(pointers), 0); + if (err < 0) + return err; + + dice->global_offset = be32_to_cpu(pointers[0]) * 4; + dice->rx_offset = be32_to_cpu(pointers[4]) * 4; + + /* some very old firmwares don't tell about their clock support */ + if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) { + err = snd_fw_transaction( + dice->unit, TCODE_READ_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_CAPABILITIES), + &value, 4, 0); + if (err < 0) + return err; + dice->clock_caps = be32_to_cpu(value); + } else { + /* this should be supported by any device */ + dice->clock_caps = CLOCK_CAP_RATE_44100 | + CLOCK_CAP_RATE_48000 | + CLOCK_CAP_SOURCE_ARX1 | + CLOCK_CAP_SOURCE_INTERNAL; + } + + for (mode = 2; mode >= 0; --mode) { + err = dice_read_mode_params(dice, mode); + if (err < 0) + return err; + } + + return 0; +} + +static void dice_card_strings(struct snd_dice *dice) +{ + struct snd_card *card = dice->card; + struct fw_device *dev = fw_parent_device(dice->unit); + char vendor[32], model[32]; + unsigned int i; + int err; + + strcpy(card->driver, "DICE"); + + strcpy(card->shortname, "DICE"); + BUILD_BUG_ON(NICK_NAME_SIZE < sizeof(card->shortname)); + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + global_address(dice, GLOBAL_NICK_NAME), + card->shortname, sizeof(card->shortname), 0); + if (err >= 0) { + /* DICE strings are returned in "always-wrong" endianness */ + BUILD_BUG_ON(sizeof(card->shortname) % 4 != 0); + for (i = 0; i < sizeof(card->shortname); i += 4) + swab32s((u32 *)&card->shortname[i]); + card->shortname[sizeof(card->shortname) - 1] = '\0'; + } + + strcpy(vendor, "?"); + fw_csr_string(dev->config_rom + 5, CSR_VENDOR, vendor, sizeof(vendor)); + strcpy(model, "?"); + fw_csr_string(dice->unit->directory, CSR_MODEL, model, sizeof(model)); + snprintf(card->longname, sizeof(card->longname), + "%s %s (serial %u) at %s, S%d", + vendor, model, dev->config_rom[4] & 0x3fffff, + dev_name(&dice->unit->device), 100 << dev->max_speed); + + strcpy(card->mixername, "DICE"); +} + +static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) +{ + struct snd_card *card; + struct snd_dice *dice; + __be32 clock_sel; + int err; + + err = dice_interface_check(unit); + if (err < 0) + return err; + + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, + sizeof(*dice), &card); + if (err < 0) + return err; + + dice = card->private_data; + dice->card = card; + spin_lock_init(&dice->lock); + mutex_init(&dice->mutex); + dice->unit = unit; + init_completion(&dice->clock_accepted); + init_waitqueue_head(&dice->hwdep_wait); + + dice->notification_handler.length = 4; + dice->notification_handler.address_callback = dice_notification; + dice->notification_handler.callback_data = dice; + err = fw_core_add_address_handler(&dice->notification_handler, + &fw_high_memory_region); + if (err < 0) + goto err_mutex; + + err = dice_owner_set(dice); + if (err < 0) + goto err_notification_handler; + + err = dice_read_params(dice); + if (err < 0) + goto err_owner; + + err = fw_iso_resources_init(&dice->rx_resources, unit); + if (err < 0) + goto err_owner; + dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; + + err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, + CIP_BLOCKING); + if (err < 0) + goto err_resources; + + card->private_free = dice_card_free; + + dice_card_strings(dice); + + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock_sel, 4, 0); + if (err < 0) + goto error; + clock_sel &= cpu_to_be32(~CLOCK_SOURCE_MASK); + clock_sel |= cpu_to_be32(CLOCK_SOURCE_ARX1); + err = snd_fw_transaction(unit, TCODE_WRITE_QUADLET_REQUEST, + global_address(dice, GLOBAL_CLOCK_SELECT), + &clock_sel, 4, 0); + if (err < 0) + goto error; + + err = dice_create_pcm(dice); + if (err < 0) + goto error; + + err = dice_create_hwdep(dice); + if (err < 0) + goto error; + + dice_create_proc(dice); + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, dice); + + return 0; + +err_resources: + fw_iso_resources_destroy(&dice->rx_resources); +err_owner: + dice_owner_clear(dice); +err_notification_handler: + fw_core_remove_address_handler(&dice->notification_handler); +err_mutex: + mutex_destroy(&dice->mutex); +error: + snd_card_free(card); + return err; +} + +static void dice_remove(struct fw_unit *unit) +{ + struct snd_dice *dice = dev_get_drvdata(&unit->device); + + amdtp_stream_pcm_abort(&dice->rx_stream); + + snd_card_disconnect(dice->card); + + mutex_lock(&dice->mutex); + + dice_stream_stop(dice); + dice_owner_clear(dice); + + mutex_unlock(&dice->mutex); + + snd_card_free_when_closed(dice->card); +} + +static void dice_bus_reset(struct fw_unit *unit) +{ + struct snd_dice *dice = dev_get_drvdata(&unit->device); + + /* + * On a bus reset, the DICE firmware disables streaming and then goes + * off contemplating its own navel for hundreds of milliseconds before + * it can react to any of our attempts to reenable streaming. This + * means that we lose synchronization anyway, so we force our streams + * to stop so that the application can restart them in an orderly + * manner. + */ + amdtp_stream_pcm_abort(&dice->rx_stream); + + mutex_lock(&dice->mutex); + + dice->global_enabled = false; + dice_stream_stop_packets(dice); + + dice_owner_update(dice); + + fw_iso_resources_update(&dice->rx_resources); + + mutex_unlock(&dice->mutex); +} + +#define DICE_INTERFACE 0x000001 + +static const struct ieee1394_device_id dice_id_table[] = { + { + .match_flags = IEEE1394_MATCH_VERSION, + .version = DICE_INTERFACE, + }, + { } +}; +MODULE_DEVICE_TABLE(ieee1394, dice_id_table); + +static struct fw_driver dice_driver = { + .driver = { + .owner = THIS_MODULE, + .name = KBUILD_MODNAME, + .bus = &fw_bus_type, + }, + .probe = dice_probe, + .update = dice_bus_reset, + .remove = dice_remove, + .id_table = dice_id_table, +}; + +static int __init alsa_dice_init(void) +{ + return driver_register(&dice_driver.driver); +} + +static void __exit alsa_dice_exit(void) +{ + driver_unregister(&dice_driver.driver); +} + +module_init(alsa_dice_init); +module_exit(alsa_dice_exit); -- cgit v1.1 From 7c2d4c0cf5bacb42bc3079e61d299dfaa3dbdde5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:13 +0900 Subject: ALSA: dice: Split transaction functionality into a file This commit adds a file with some helper functions for transaction, and move some codes into the file with some arrangements. For Dice chipset, well-known FCP or AV/C commands are not used to control devices. It's achieved by read/write transactions into specific addresses. Dice's address area is split into 5 areas. Each area has its own role. The offset for each area can be got by reading head of the address area. By reading these areas, drivers can get to know device status. By writing these areas, drivers can change device status. Dice has a specific mechanism called as 'notification'. When device status is changed, Dice devices tells the event by sending transaction. This notification is sent to an address which drivers register in advance. But this causes an issue to drivers. To handle the notification, drivers need to allocate its own callback function to the address region in host controller. This region is exclusive. For the other applications, drivers must give a mechanism to read the received notification. For this purpose, Dice driver already implements hwdep interface. Dice chipset doesn't allow drivers to register several addresses. In this reason, when this driver is applied to a device, the other drivers should _not_ try to register its own address to the device. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-transaction.c | 387 ++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 487 +++++++-------------------------- sound/firewire/dice/dice.h | 160 +++++++++++ 4 files changed, 653 insertions(+), 383 deletions(-) create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.h diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index af05d7e..9f473cb 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,2 +1,2 @@ -snd-dice-objs := dice.o +snd-dice-objs := dice-transaction.o dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c new file mode 100644 index 0000000..a9b98e0 --- /dev/null +++ b/sound/firewire/dice/dice-transaction.c @@ -0,0 +1,387 @@ +/* + * dice_transaction.c - a part of driver for Dice based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "dice.h" + +#define NOTIFICATION_TIMEOUT_MS 100 + +static u64 get_subaddr(struct snd_dice *dice, enum snd_dice_addr_type type, + u64 offset) +{ + switch (type) { + case SND_DICE_ADDR_TYPE_TX: + offset += dice->tx_offset; + break; + case SND_DICE_ADDR_TYPE_RX: + offset += dice->rx_offset; + break; + case SND_DICE_ADDR_TYPE_SYNC: + offset += dice->sync_offset; + break; + case SND_DICE_ADDR_TYPE_RSRV: + offset += dice->rsrv_offset; + break; + case SND_DICE_ADDR_TYPE_GLOBAL: + default: + offset += dice->global_offset; + break; + }; + offset += DICE_PRIVATE_SPACE; + return offset; +} + +int snd_dice_transaction_write(struct snd_dice *dice, + enum snd_dice_addr_type type, + unsigned int offset, void *buf, unsigned int len) +{ + return snd_fw_transaction(dice->unit, + (len == 4) ? TCODE_WRITE_QUADLET_REQUEST : + TCODE_WRITE_BLOCK_REQUEST, + get_subaddr(dice, type, offset), buf, len, 0); +} + +int snd_dice_transaction_read(struct snd_dice *dice, + enum snd_dice_addr_type type, unsigned int offset, + void *buf, unsigned int len) +{ + return snd_fw_transaction(dice->unit, + (len == 4) ? TCODE_READ_QUADLET_REQUEST : + TCODE_READ_BLOCK_REQUEST, + get_subaddr(dice, type, offset), buf, len, 0); +} + +static unsigned int get_clock_info(struct snd_dice *dice, __be32 *info) +{ + return snd_dice_transaction_read_global(dice, GLOBAL_CLOCK_SELECT, + info, 4); +} + +static int set_clock_info(struct snd_dice *dice, + unsigned int rate, unsigned int source) +{ + unsigned int retries = 3; + unsigned int i; + __be32 info; + u32 mask; + u32 clock; + int err; +retry: + err = get_clock_info(dice, &info); + if (err < 0) + goto end; + + clock = be32_to_cpu(info); + if (source != UINT_MAX) { + mask = CLOCK_SOURCE_MASK; + clock &= ~mask; + clock |= source; + } + if (rate != UINT_MAX) { + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) { + if (snd_dice_rates[i] == rate) + break; + } + if (i == ARRAY_SIZE(snd_dice_rates)) { + err = -EINVAL; + goto end; + } + + mask = CLOCK_RATE_MASK; + clock &= ~mask; + clock |= i << CLOCK_RATE_SHIFT; + } + info = cpu_to_be32(clock); + + if (completion_done(&dice->clock_accepted)) + reinit_completion(&dice->clock_accepted); + + err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT, + &info, 4); + if (err < 0) + goto end; + + /* Timeout means it's invalid request, probably bus reset occurred. */ + if (wait_for_completion_timeout(&dice->clock_accepted, + msecs_to_jiffies(NOTIFICATION_TIMEOUT_MS)) == 0) { + if (retries-- == 0) { + err = -ETIMEDOUT; + goto end; + } + + err = snd_dice_transaction_reinit(dice); + if (err < 0) + goto end; + + msleep(500); /* arbitrary */ + goto retry; + } +end: + return err; +} + +int snd_dice_transaction_get_clock_source(struct snd_dice *dice, + unsigned int *source) +{ + __be32 info; + int err; + + err = get_clock_info(dice, &info); + if (err >= 0) + *source = be32_to_cpu(info) & CLOCK_SOURCE_MASK; + + return err; +} +int snd_dice_transaction_set_clock_source(struct snd_dice *dice, + unsigned int source) +{ + return set_clock_info(dice, UINT_MAX, source); +} + +int snd_dice_transaction_get_rate(struct snd_dice *dice, unsigned int *rate) +{ + __be32 info; + unsigned int index; + int err; + + err = get_clock_info(dice, &info); + if (err < 0) + goto end; + + index = (be32_to_cpu(info) & CLOCK_RATE_MASK) >> CLOCK_RATE_SHIFT; + if (index >= SND_DICE_RATES_COUNT) { + err = -ENOSYS; + goto end; + } + + *rate = snd_dice_rates[index]; +end: + return err; +} +int snd_dice_transaction_set_rate(struct snd_dice *dice, unsigned int rate) +{ + return set_clock_info(dice, rate, UINT_MAX); +} + +int snd_dice_transaction_set_enable(struct snd_dice *dice) +{ + __be32 value; + int err = 0; + + if (dice->global_enabled) + goto end; + + value = cpu_to_be32(1); + err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + get_subaddr(dice, SND_DICE_ADDR_TYPE_GLOBAL, + GLOBAL_ENABLE), + &value, 4, + FW_FIXED_GENERATION | dice->owner_generation); + if (err < 0) + goto end; + + dice->global_enabled = true; +end: + return err; +} + +void snd_dice_transaction_clear_enable(struct snd_dice *dice) +{ + __be32 value; + + value = 0; + snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, + get_subaddr(dice, SND_DICE_ADDR_TYPE_GLOBAL, + GLOBAL_ENABLE), + &value, 4, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); + + dice->global_enabled = false; +} + +static void dice_notification(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct snd_dice *dice = callback_data; + u32 bits; + unsigned long flags; + + if (tcode != TCODE_WRITE_QUADLET_REQUEST) { + fw_send_response(card, request, RCODE_TYPE_ERROR); + return; + } + if ((offset & 3) != 0) { + fw_send_response(card, request, RCODE_ADDRESS_ERROR); + return; + } + + bits = be32_to_cpup(data); + + spin_lock_irqsave(&dice->lock, flags); + dice->notification_bits |= bits; + spin_unlock_irqrestore(&dice->lock, flags); + + fw_send_response(card, request, RCODE_COMPLETE); + + if (bits & NOTIFY_CLOCK_ACCEPTED) + complete(&dice->clock_accepted); + wake_up(&dice->hwdep_wait); +} + +static int register_notification_address(struct snd_dice *dice, bool retry) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + unsigned int retries; + int err; + + retries = (retry) ? 3 : 0; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (!buffer) + return -ENOMEM; + + for (;;) { + buffer[0] = cpu_to_be64(OWNER_NO_OWNER); + buffer[1] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + + dice->owner_generation = device->generation; + smp_rmb(); /* node_id vs. generation */ + err = snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + get_subaddr(dice, + SND_DICE_ADDR_TYPE_GLOBAL, + GLOBAL_OWNER), + buffer, 2 * 8, + FW_FIXED_GENERATION | + dice->owner_generation); + if (err == 0) { + /* success */ + if (buffer[0] == cpu_to_be64(OWNER_NO_OWNER)) + break; + /* The address seems to be already registered. */ + if (buffer[0] == buffer[1]) + break; + + dev_err(&dice->unit->device, + "device is already in use\n"); + err = -EBUSY; + } + if (err != -EAGAIN || retries-- > 0) + break; + + msleep(20); + } + + kfree(buffer); + + if (err < 0) + dice->owner_generation = -1; + + return err; +} + +static void unregister_notification_address(struct snd_dice *dice) +{ + struct fw_device *device = fw_parent_device(dice->unit); + __be64 *buffer; + + buffer = kmalloc(2 * 8, GFP_KERNEL); + if (buffer == NULL) + return; + + buffer[0] = cpu_to_be64( + ((u64)device->card->node_id << OWNER_NODE_SHIFT) | + dice->notification_handler.offset); + buffer[1] = cpu_to_be64(OWNER_NO_OWNER); + snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, + get_subaddr(dice, SND_DICE_ADDR_TYPE_GLOBAL, + GLOBAL_OWNER), + buffer, 2 * 8, FW_QUIET | + FW_FIXED_GENERATION | dice->owner_generation); + + kfree(buffer); + + dice->owner_generation = -1; +} + +void snd_dice_transaction_destroy(struct snd_dice *dice) +{ + struct fw_address_handler *handler = &dice->notification_handler; + + if (handler->callback_data == NULL) + return; + + unregister_notification_address(dice); + + fw_core_remove_address_handler(handler); + handler->callback_data = NULL; +} + +int snd_dice_transaction_reinit(struct snd_dice *dice) +{ + struct fw_address_handler *handler = &dice->notification_handler; + + if (handler->callback_data == NULL) + return -EINVAL; + + return register_notification_address(dice, false); +} + +int snd_dice_transaction_init(struct snd_dice *dice) +{ + struct fw_address_handler *handler = &dice->notification_handler; + __be32 *pointers; + int err; + + /* Use the same way which dice_interface_check() does. */ + pointers = kmalloc(sizeof(__be32) * 10, GFP_KERNEL); + if (pointers == NULL) + return -ENOMEM; + + /* Get offsets for sub-addresses */ + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE, + pointers, sizeof(__be32) * 10, 0); + if (err < 0) + goto end; + + /* Allocation callback in address space over host controller */ + handler->length = 4; + handler->address_callback = dice_notification; + handler->callback_data = dice; + err = fw_core_add_address_handler(handler, &fw_high_memory_region); + if (err < 0) { + handler->callback_data = NULL; + goto end; + } + + /* Register the address space */ + err = register_notification_address(dice, true); + if (err < 0) { + fw_core_remove_address_handler(handler); + handler->callback_data = NULL; + goto end; + } + + dice->global_offset = be32_to_cpu(pointers[0]) * 4; + dice->tx_offset = be32_to_cpu(pointers[2]) * 4; + dice->rx_offset = be32_to_cpu(pointers[4]) * 4; + dice->sync_offset = be32_to_cpu(pointers[6]) * 4; + dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4; + + /* Set up later. */ + if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) + dice->clock_caps = 1; +end: + kfree(pointers); + return err; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index d3ec778..dd62316 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -5,60 +5,13 @@ * Licensed under the terms of the GNU General Public License, version 2. */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../amdtp.h" -#include "../iso-resources.h" -#include "../lib.h" -#include "dice-interface.h" - - -struct snd_dice { - struct snd_card *card; - struct fw_unit *unit; - spinlock_t lock; - struct mutex mutex; - unsigned int global_offset; - unsigned int rx_offset; - unsigned int clock_caps; - unsigned int rx_channels[3]; - unsigned int rx_midi_ports[3]; - struct fw_address_handler notification_handler; - int owner_generation; - int dev_lock_count; /* > 0 driver, < 0 userspace */ - bool dev_lock_changed; - bool global_enabled; - struct completion clock_accepted; - wait_queue_head_t hwdep_wait; - u32 notification_bits; - struct fw_iso_resources rx_resources; - struct amdtp_stream rx_stream; -}; +#include "dice.h" MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -static const unsigned int dice_rates[] = { +const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { /* mode 0 */ [0] = 32000, [1] = 44100, @@ -75,8 +28,8 @@ static unsigned int rate_to_index(unsigned int rate) { unsigned int i; - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) - if (dice_rates[i] == rate) + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) + if (snd_dice_rates[i] == rate) return i; return 0; @@ -128,192 +81,6 @@ out: spin_unlock_irq(&dice->lock); } -static inline u64 global_address(struct snd_dice *dice, unsigned int offset) -{ - return DICE_PRIVATE_SPACE + dice->global_offset + offset; -} - -/* TODO: rx index */ -static inline u64 rx_address(struct snd_dice *dice, unsigned int offset) -{ - return DICE_PRIVATE_SPACE + dice->rx_offset + offset; -} - -static int dice_owner_set(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - int err, errors = 0; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return -ENOMEM; - - for (;;) { - buffer[0] = cpu_to_be64(OWNER_NO_OWNER); - buffer[1] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - - dice->owner_generation = device->generation; - smp_rmb(); /* node_id vs. generation */ - err = snd_fw_transaction(dice->unit, - TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, - FW_FIXED_GENERATION | - dice->owner_generation); - - if (err == 0) { - if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { - dev_err(&dice->unit->device, - "device is already in use\n"); - err = -EBUSY; - } - break; - } - if (err != -EAGAIN || ++errors >= 3) - break; - - msleep(20); - } - - kfree(buffer); - - return err; -} - -static int dice_owner_update(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - int err; - - if (dice->owner_generation == -1) - return 0; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return -ENOMEM; - - buffer[0] = cpu_to_be64(OWNER_NO_OWNER); - buffer[1] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - - dice->owner_generation = device->generation; - smp_rmb(); /* node_id vs. generation */ - err = snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, - FW_FIXED_GENERATION | dice->owner_generation); - - if (err == 0) { - if (buffer[0] != cpu_to_be64(OWNER_NO_OWNER)) { - dev_err(&dice->unit->device, - "device is already in use\n"); - err = -EBUSY; - } - } else if (err == -EAGAIN) { - err = 0; /* try again later */ - } - - kfree(buffer); - - if (err < 0) - dice->owner_generation = -1; - - return err; -} - -static void dice_owner_clear(struct snd_dice *dice) -{ - struct fw_device *device = fw_parent_device(dice->unit); - __be64 *buffer; - - buffer = kmalloc(2 * 8, GFP_KERNEL); - if (!buffer) - return; - - buffer[0] = cpu_to_be64( - ((u64)device->card->node_id << OWNER_NODE_SHIFT) | - dice->notification_handler.offset); - buffer[1] = cpu_to_be64(OWNER_NO_OWNER); - snd_fw_transaction(dice->unit, TCODE_LOCK_COMPARE_SWAP, - global_address(dice, GLOBAL_OWNER), - buffer, 2 * 8, FW_QUIET | - FW_FIXED_GENERATION | dice->owner_generation); - - kfree(buffer); - - dice->owner_generation = -1; -} - -static int dice_enable_set(struct snd_dice *dice) -{ - __be32 value; - int err; - - value = cpu_to_be32(1); - err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_ENABLE), - &value, 4, - FW_FIXED_GENERATION | dice->owner_generation); - if (err < 0) - return err; - - dice->global_enabled = true; - - return 0; -} - -static void dice_enable_clear(struct snd_dice *dice) -{ - __be32 value; - - if (!dice->global_enabled) - return; - - value = 0; - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_ENABLE), - &value, 4, FW_QUIET | - FW_FIXED_GENERATION | dice->owner_generation); - - dice->global_enabled = false; -} - -static void dice_notification(struct fw_card *card, struct fw_request *request, - int tcode, int destination, int source, - int generation, unsigned long long offset, - void *data, size_t length, void *callback_data) -{ - struct snd_dice *dice = callback_data; - u32 bits; - unsigned long flags; - - if (tcode != TCODE_WRITE_QUADLET_REQUEST) { - fw_send_response(card, request, RCODE_TYPE_ERROR); - return; - } - if ((offset & 3) != 0) { - fw_send_response(card, request, RCODE_ADDRESS_ERROR); - return; - } - - bits = be32_to_cpup(data); - - spin_lock_irqsave(&dice->lock, flags); - dice->notification_bits |= bits; - spin_unlock_irqrestore(&dice->lock, flags); - - fw_send_response(card, request, RCODE_COMPLETE); - - if (bits & NOTIFY_CLOCK_ACCEPTED) - complete(&dice->clock_accepted); - wake_up(&dice->hwdep_wait); -} - static int dice_rate_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -327,14 +94,14 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, }; unsigned int i, mode; - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) { + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { mode = rate_index_to_mode(i); if ((dice->clock_caps & (1 << i)) && snd_interval_test(channels, dice->rx_channels[mode])) { allowed_rates.min = min(allowed_rates.min, - dice_rates[i]); + snd_dice_rates[i]); allowed_rates.max = max(allowed_rates.max, - dice_rates[i]); + snd_dice_rates[i]); } } @@ -354,9 +121,9 @@ static int dice_channels_constraint(struct snd_pcm_hw_params *params, }; unsigned int i, mode; - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) if ((dice->clock_caps & (1 << i)) && - snd_interval_test(rate, dice_rates[i])) { + snd_interval_test(rate, snd_dice_rates[i])) { mode = rate_index_to_mode(i); allowed_channels.min = min(allowed_channels.min, dice->rx_channels[mode]); @@ -395,10 +162,10 @@ static int dice_open(struct snd_pcm_substream *substream) runtime->hw = hardware; - for (i = 0; i < ARRAY_SIZE(dice_rates); ++i) + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) if (dice->clock_caps & (1 << i)) runtime->hw.rates |= - snd_pcm_rate_to_rate_bit(dice_rates[i]); + snd_pcm_rate_to_rate_bit(snd_dice_rates[i]); snd_pcm_limit_hw_rates(runtime); for (i = 0; i < 3; ++i) @@ -453,7 +220,7 @@ static int dice_stream_start_packets(struct snd_dice *dice) if (err < 0) return err; - err = dice_enable_set(dice); + err = snd_dice_transaction_set_enable(dice); if (err < 0) { amdtp_stream_stop(&dice->rx_stream); return err; @@ -475,10 +242,8 @@ static int dice_stream_start(struct snd_dice *dice) goto error; channel = cpu_to_be32(dice->rx_resources.channel); - err = snd_fw_transaction(dice->unit, - TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), - &channel, 4, 0); + err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, + &channel, 4); if (err < 0) goto err_resources; } @@ -491,8 +256,7 @@ static int dice_stream_start(struct snd_dice *dice) err_rx_channel: channel = cpu_to_be32((u32)-1); - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); err_resources: fw_iso_resources_free(&dice->rx_resources); error: @@ -502,7 +266,7 @@ error: static void dice_stream_stop_packets(struct snd_dice *dice) { if (amdtp_stream_running(&dice->rx_stream)) { - dice_enable_clear(dice); + snd_dice_transaction_clear_enable(dice); amdtp_stream_stop(&dice->rx_stream); } } @@ -517,33 +281,11 @@ static void dice_stream_stop(struct snd_dice *dice) return; channel = cpu_to_be32((u32)-1); - snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - rx_address(dice, RX_ISOCHRONOUS), &channel, 4, 0); + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); fw_iso_resources_free(&dice->rx_resources); } -static int dice_change_rate(struct snd_dice *dice, unsigned int clock_rate) -{ - __be32 value; - int err; - - reinit_completion(&dice->clock_accepted); - - value = cpu_to_be32(clock_rate | CLOCK_SOURCE_ARX1); - err = snd_fw_transaction(dice->unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &value, 4, 0); - if (err < 0) - return err; - - if (!wait_for_completion_timeout(&dice->clock_accepted, - msecs_to_jiffies(100))) - dev_warn(&dice->unit->device, "clock change timed out\n"); - - return 0; -} - static int dice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -561,8 +303,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; rate = params_rate(hw_params); - rate_index = rate_to_index(rate); - err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + err = snd_dice_transaction_set_rate(dice, rate); if (err < 0) return err; @@ -577,6 +318,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); + rate_index = rate_to_index(rate); if (rate_index > 4) { if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { err = -ENOSYS; @@ -1118,15 +860,6 @@ static void dice_create_proc(struct snd_dice *dice) snd_info_set_text_ops(entry, dice, dice_proc_read); } -static void dice_card_free(struct snd_card *card) -{ - struct snd_dice *dice = card->private_data; - - amdtp_stream_destroy(&dice->rx_stream); - fw_core_remove_address_handler(&dice->notification_handler); - mutex_destroy(&dice->mutex); -} - #define OUI_WEISS 0x001c6a #define DICE_CATEGORY_ID 0x04 @@ -1143,12 +876,17 @@ static int dice_interface_check(struct fw_unit *unit) }; struct fw_device *device = fw_parent_device(unit); struct fw_csr_iterator it; - int key, value, vendor = -1, model = -1, err; + int key, val, vendor = -1, model = -1, err; unsigned int category, i; - __be32 pointers[ARRAY_SIZE(min_values)]; + __be32 *pointers, value; __be32 tx_data[4]; __be32 version; + pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32), + GFP_KERNEL); + if (pointers == NULL) + return -ENOMEM; + /* * Check that GUID and unit directory are constructed according to DICE * rules, i.e., that the specifier ID is the GUID's OUI, and that the @@ -1156,13 +894,13 @@ static int dice_interface_check(struct fw_unit *unit) * ID, and a 22-bit serial number. */ fw_csr_iterator_init(&it, unit->directory); - while (fw_csr_iterator_next(&it, &key, &value)) { + while (fw_csr_iterator_next(&it, &key, &val)) { switch (key) { case CSR_SPECIFIER_ID: - vendor = value; + vendor = val; break; case CSR_MODEL: - model = value; + model = val; break; } } @@ -1171,8 +909,10 @@ static int dice_interface_check(struct fw_unit *unit) else category = DICE_CATEGORY_ID; if (device->config_rom[3] != ((vendor << 8) | category) || - device->config_rom[4] >> 22 != model) - return -ENODEV; + device->config_rom[4] >> 22 != model) { + err = -ENODEV; + goto end; + } /* * Check that the sub address spaces exist and are located inside the @@ -1180,14 +920,18 @@ static int dice_interface_check(struct fw_unit *unit) * minimally required registers are included. */ err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE, - pointers, sizeof(pointers), 0); - if (err < 0) - return -ENODEV; - for (i = 0; i < ARRAY_SIZE(pointers); ++i) { + DICE_PRIVATE_SPACE, pointers, + sizeof(__be32) * ARRAY_SIZE(min_values), 0); + if (err < 0) { + err = -ENODEV; + goto end; + } + for (i = 0; i < ARRAY_SIZE(min_values); ++i) { value = be32_to_cpu(pointers[i]); - if (value < min_values[i] || value >= 0x40000) - return -ENODEV; + if (value < min_values[i] || value >= 0x40000) { + err = -ENODEV; + goto end; + } } /* We support playback only. Let capture devices be handled by FFADO. */ @@ -1195,8 +939,10 @@ static int dice_interface_check(struct fw_unit *unit) DICE_PRIVATE_SPACE + be32_to_cpu(pointers[2]) * 4, tx_data, sizeof(tx_data), 0); - if (err < 0 || (tx_data[0] && tx_data[3])) - return -ENODEV; + if (err < 0 || (tx_data[0] && tx_data[3])) { + err = -ENODEV; + goto end; + } /* * Check that the implemented DICE driver specification major version @@ -1206,22 +952,25 @@ static int dice_interface_check(struct fw_unit *unit) DICE_PRIVATE_SPACE + be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION, &version, 4, 0); - if (err < 0) - return -ENODEV; + if (err < 0) { + err = -ENODEV; + goto end; + } if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) { dev_err(&unit->device, "unknown DICE version: 0x%08x\n", be32_to_cpu(version)); - return -ENODEV; + err = -ENODEV; + goto end; } - - return 0; +end: + return err; } static int highest_supported_mode_rate(struct snd_dice *dice, unsigned int mode) { int i; - for (i = ARRAY_SIZE(dice_rates) - 1; i >= 0; --i) + for (i = ARRAY_SIZE(snd_dice_rates) - 1; i >= 0; --i) if ((dice->clock_caps & (1 << i)) && rate_index_to_mode(i) == mode) return i; @@ -1241,13 +990,12 @@ static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) return 0; } - err = dice_change_rate(dice, rate_index << CLOCK_RATE_SHIFT); + err = snd_dice_transaction_set_rate(dice, snd_dice_rates[rate_index]); if (err < 0) return err; - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - rx_address(dice, RX_NUMBER_AUDIO), - values, 2 * 4, 0); + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, + values, sizeof(values)); if (err < 0) return err; @@ -1259,25 +1007,14 @@ static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) static int dice_read_params(struct snd_dice *dice) { - __be32 pointers[6]; __be32 value; int mode, err; - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE, - pointers, sizeof(pointers), 0); - if (err < 0) - return err; - - dice->global_offset = be32_to_cpu(pointers[0]) * 4; - dice->rx_offset = be32_to_cpu(pointers[4]) * 4; - /* some very old firmwares don't tell about their clock support */ - if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4) { - err = snd_fw_transaction( - dice->unit, TCODE_READ_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_CAPABILITIES), - &value, 4, 0); + if (dice->clock_caps > 0) { + err = snd_dice_transaction_read_global(dice, + GLOBAL_CLOCK_CAPABILITIES, + &value, 4); if (err < 0) return err; dice->clock_caps = be32_to_cpu(value); @@ -1310,9 +1047,9 @@ static void dice_card_strings(struct snd_dice *dice) strcpy(card->shortname, "DICE"); BUILD_BUG_ON(NICK_NAME_SIZE < sizeof(card->shortname)); - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - global_address(dice, GLOBAL_NICK_NAME), - card->shortname, sizeof(card->shortname), 0); + err = snd_dice_transaction_read_global(dice, GLOBAL_NICK_NAME, + card->shortname, + sizeof(card->shortname)); if (err >= 0) { /* DICE strings are returned in "always-wrong" endianness */ BUILD_BUG_ON(sizeof(card->shortname) % 4 != 0); @@ -1333,70 +1070,50 @@ static void dice_card_strings(struct snd_dice *dice) strcpy(card->mixername, "DICE"); } +static void dice_card_free(struct snd_card *card) +{ + struct snd_dice *dice = card->private_data; + + snd_dice_transaction_destroy(dice); + mutex_destroy(&dice->mutex); +} + static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_card *card; struct snd_dice *dice; - __be32 clock_sel; int err; err = dice_interface_check(unit); if (err < 0) - return err; + goto end; err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, sizeof(*dice), &card); if (err < 0) - return err; + goto end; dice = card->private_data; dice->card = card; + dice->unit = unit; + card->private_free = dice_card_free; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); - dice->unit = unit; init_completion(&dice->clock_accepted); init_waitqueue_head(&dice->hwdep_wait); - dice->notification_handler.length = 4; - dice->notification_handler.address_callback = dice_notification; - dice->notification_handler.callback_data = dice; - err = fw_core_add_address_handler(&dice->notification_handler, - &fw_high_memory_region); - if (err < 0) - goto err_mutex; - - err = dice_owner_set(dice); + err = snd_dice_transaction_init(dice); if (err < 0) - goto err_notification_handler; + goto error; err = dice_read_params(dice); if (err < 0) - goto err_owner; - - err = fw_iso_resources_init(&dice->rx_resources, unit); - if (err < 0) - goto err_owner; - dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; - - err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, - CIP_BLOCKING); - if (err < 0) - goto err_resources; - - card->private_free = dice_card_free; + goto error; dice_card_strings(dice); - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4, 0); - if (err < 0) - goto error; - clock_sel &= cpu_to_be32(~CLOCK_SOURCE_MASK); - clock_sel |= cpu_to_be32(CLOCK_SOURCE_ARX1); - err = snd_fw_transaction(unit, TCODE_WRITE_QUADLET_REQUEST, - global_address(dice, GLOBAL_CLOCK_SELECT), - &clock_sel, 4, 0); + err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); if (err < 0) goto error; @@ -1410,22 +1127,28 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice_create_proc(dice); - err = snd_card_register(card); + err = fw_iso_resources_init(&dice->rx_resources, unit); if (err < 0) goto error; + dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; - dev_set_drvdata(&unit->device, dice); + err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, + CIP_BLOCKING); + if (err < 0) { + fw_iso_resources_destroy(&dice->rx_resources); + goto error; + } - return 0; + err = snd_card_register(card); + if (err < 0) { + amdtp_stream_destroy(&dice->rx_stream); + fw_iso_resources_destroy(&dice->rx_resources); + goto error; + } -err_resources: - fw_iso_resources_destroy(&dice->rx_resources); -err_owner: - dice_owner_clear(dice); -err_notification_handler: - fw_core_remove_address_handler(&dice->notification_handler); -err_mutex: - mutex_destroy(&dice->mutex); + dev_set_drvdata(&unit->device, dice); +end: + return err; error: snd_card_free(card); return err; @@ -1442,7 +1165,6 @@ static void dice_remove(struct fw_unit *unit) mutex_lock(&dice->mutex); dice_stream_stop(dice); - dice_owner_clear(dice); mutex_unlock(&dice->mutex); @@ -1453,6 +1175,9 @@ static void dice_bus_reset(struct fw_unit *unit) { struct snd_dice *dice = dev_get_drvdata(&unit->device); + /* The handler address register becomes initialized. */ + snd_dice_transaction_reinit(dice); + /* * On a bus reset, the DICE firmware disables streaming and then goes * off contemplating its own navel for hundreds of milliseconds before @@ -1466,10 +1191,8 @@ static void dice_bus_reset(struct fw_unit *unit) mutex_lock(&dice->mutex); dice->global_enabled = false; - dice_stream_stop_packets(dice); - - dice_owner_update(dice); + dice_stream_stop_packets(dice); fw_iso_resources_update(&dice->rx_resources); mutex_unlock(&dice->mutex); diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h new file mode 100644 index 0000000..c756e62 --- /dev/null +++ b/sound/firewire/dice/dice.h @@ -0,0 +1,160 @@ +/* + * dice.h - a part of driver for Dice based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#ifndef SOUND_DICE_H_INCLUDED +#define SOUND_DICE_H_INCLUDED + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../amdtp.h" +#include "../iso-resources.h" +#include "../lib.h" +#include "dice-interface.h" + +struct snd_dice { + struct snd_card *card; + struct fw_unit *unit; + spinlock_t lock; + struct mutex mutex; + + /* Offsets for sub-addresses */ + unsigned int global_offset; + unsigned int rx_offset; + unsigned int tx_offset; + unsigned int sync_offset; + unsigned int rsrv_offset; + + unsigned int clock_caps; + unsigned int rx_channels[3]; + unsigned int rx_midi_ports[3]; + struct fw_address_handler notification_handler; + int owner_generation; + int dev_lock_count; /* > 0 driver, < 0 userspace */ + bool dev_lock_changed; + bool global_enabled; + struct completion clock_accepted; + wait_queue_head_t hwdep_wait; + u32 notification_bits; + struct fw_iso_resources rx_resources; + struct amdtp_stream rx_stream; +}; + +enum snd_dice_addr_type { + SND_DICE_ADDR_TYPE_PRIVATE, + SND_DICE_ADDR_TYPE_GLOBAL, + SND_DICE_ADDR_TYPE_TX, + SND_DICE_ADDR_TYPE_RX, + SND_DICE_ADDR_TYPE_SYNC, + SND_DICE_ADDR_TYPE_RSRV, +}; + +int snd_dice_transaction_write(struct snd_dice *dice, + enum snd_dice_addr_type type, + unsigned int offset, + void *buf, unsigned int len); +int snd_dice_transaction_read(struct snd_dice *dice, + enum snd_dice_addr_type type, unsigned int offset, + void *buf, unsigned int len); + +static inline int snd_dice_transaction_write_global(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_write(dice, + SND_DICE_ADDR_TYPE_GLOBAL, offset, + buf, len); +} +static inline int snd_dice_transaction_read_global(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_read(dice, + SND_DICE_ADDR_TYPE_GLOBAL, offset, + buf, len); +} +static inline int snd_dice_transaction_write_tx(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_write(dice, SND_DICE_ADDR_TYPE_TX, offset, + buf, len); +} +static inline int snd_dice_transaction_read_tx(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_read(dice, SND_DICE_ADDR_TYPE_TX, offset, + buf, len); +} +static inline int snd_dice_transaction_write_rx(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_write(dice, SND_DICE_ADDR_TYPE_RX, offset, + buf, len); +} +static inline int snd_dice_transaction_read_rx(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_read(dice, SND_DICE_ADDR_TYPE_RX, offset, + buf, len); +} +static inline int snd_dice_transaction_write_sync(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_write(dice, SND_DICE_ADDR_TYPE_SYNC, offset, + buf, len); +} +static inline int snd_dice_transaction_read_sync(struct snd_dice *dice, + unsigned int offset, + void *buf, unsigned int len) +{ + return snd_dice_transaction_read(dice, SND_DICE_ADDR_TYPE_SYNC, offset, + buf, len); +} + +int snd_dice_transaction_set_clock_source(struct snd_dice *dice, + unsigned int source); +int snd_dice_transaction_get_clock_source(struct snd_dice *dice, + unsigned int *source); +int snd_dice_transaction_set_rate(struct snd_dice *dice, unsigned int rate); +int snd_dice_transaction_get_rate(struct snd_dice *dice, unsigned int *rate); +int snd_dice_transaction_set_enable(struct snd_dice *dice); +void snd_dice_transaction_clear_enable(struct snd_dice *dice); +int snd_dice_transaction_init(struct snd_dice *dice); +int snd_dice_transaction_reinit(struct snd_dice *dice); +void snd_dice_transaction_destroy(struct snd_dice *dice); + +#define SND_DICE_RATES_COUNT 7 +extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; + +#endif -- cgit v1.1 From 6eb6c81eee2a6270b39ca02a446f3ccece24b6f8 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:14 +0900 Subject: ALSA: dice: Split stream functionality into a file This commit adds a file with some helper functions for streaming, and move some codes into the file with some arrangements. Well-known CMP is not used to start/stop streams for Dice chipset. It's achieved by writing to specific address. We call this way as 'enable'. When devices are 'enabled', streaming starts in registered isochronous channel. Some helper functions are already implemented in previous commit. Basically, the stream is compliant to IEC 61883-6, so-called as AMDTP. But Dice has a specific quirk, so called-as 'Dual Wire'. This quirk is applied at 176.4/192.0kHz. In this mode, each packet includes double number of events than number in the specification, and stream runs at a half of sampling rate. There is another quirk at bus reset. Dice chipset handles drivers' request but don't re-enable streaming. So stream should be stopped. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-stream.c | 206 ++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 299 +++++++++----------------------------- sound/firewire/dice/dice.h | 14 ++ 4 files changed, 288 insertions(+), 233 deletions(-) create mode 100644 sound/firewire/dice/dice-stream.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9f473cb..867864c 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,2 +1,2 @@ -snd-dice-objs := dice-transaction.o dice.o +snd-dice-objs := dice-transaction.o dice-stream.o dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c new file mode 100644 index 0000000..c25b9fb --- /dev/null +++ b/sound/firewire/dice/dice-stream.c @@ -0,0 +1,206 @@ +/* + * dice_stream.c - a part of driver for DICE based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "dice.h" + +const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { + /* mode 0 */ + [0] = 32000, + [1] = 44100, + [2] = 48000, + /* mode 1 */ + [3] = 88200, + [4] = 96000, + /* mode 2 */ + [5] = 176400, + [6] = 192000, +}; + +int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, + unsigned int *mode) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) { + if (!(dice->clock_caps & BIT(i))) + continue; + if (snd_dice_rates[i] != rate) + continue; + + *mode = (i - 1) / 2; + return 0; + } + return -EINVAL; +} + +int snd_dice_stream_start_packets(struct snd_dice *dice) +{ + int err; + + if (amdtp_stream_running(&dice->rx_stream)) + return 0; + + err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + return err; + + err = snd_dice_transaction_set_enable(dice); + if (err < 0) { + amdtp_stream_stop(&dice->rx_stream); + return err; + } + + return 0; +} + +int snd_dice_stream_start(struct snd_dice *dice) +{ + __be32 channel; + int err; + + if (!dice->rx_resources.allocated) { + err = fw_iso_resources_allocate(&dice->rx_resources, + amdtp_stream_get_max_payload(&dice->rx_stream), + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + goto error; + + channel = cpu_to_be32(dice->rx_resources.channel); + err = snd_dice_transaction_write_tx(dice, RX_ISOCHRONOUS, + &channel, 4); + if (err < 0) + goto err_resources; + } + + err = snd_dice_stream_start_packets(dice); + if (err < 0) + goto err_rx_channel; + + return 0; + +err_rx_channel: + channel = cpu_to_be32((u32)-1); + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); +err_resources: + fw_iso_resources_free(&dice->rx_resources); +error: + return err; +} + +void snd_dice_stream_stop_packets(struct snd_dice *dice) +{ + if (amdtp_stream_running(&dice->rx_stream)) { + snd_dice_transaction_clear_enable(dice); + amdtp_stream_stop(&dice->rx_stream); + } +} + +void snd_dice_stream_stop(struct snd_dice *dice) +{ + __be32 channel; + + snd_dice_stream_stop_packets(dice); + + if (!dice->rx_resources.allocated) + return; + + channel = cpu_to_be32((u32)-1); + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); + + fw_iso_resources_free(&dice->rx_resources); +} + +int snd_dice_stream_init(struct snd_dice *dice) +{ + int err; + + err = fw_iso_resources_init(&dice->rx_resources, dice->unit); + if (err < 0) + goto end; + dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; + + err = amdtp_stream_init(&dice->rx_stream, dice->unit, AMDTP_OUT_STREAM, + CIP_BLOCKING); + if (err < 0) + goto error; + + err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); + if (err < 0) + goto error; +end: + return err; +error: + amdtp_stream_destroy(&dice->rx_stream); + fw_iso_resources_destroy(&dice->rx_resources); + return err; +} + +void snd_dice_stream_destroy(struct snd_dice *dice) +{ + amdtp_stream_pcm_abort(&dice->rx_stream); + snd_dice_stream_stop(dice); + amdtp_stream_destroy(&dice->rx_stream); + fw_iso_resources_destroy(&dice->rx_resources); +} + +void snd_dice_stream_update(struct snd_dice *dice) +{ + /* + * On a bus reset, the DICE firmware disables streaming and then goes + * off contemplating its own navel for hundreds of milliseconds before + * it can react to any of our attempts to reenable streaming. This + * means that we lose synchronization anyway, so we force our streams + * to stop so that the application can restart them in an orderly + * manner. + */ + dice->global_enabled = false; + + amdtp_stream_pcm_abort(&dice->rx_stream); + snd_dice_stream_stop_packets(dice); + fw_iso_resources_update(&dice->rx_resources); +} + +static void dice_lock_changed(struct snd_dice *dice) +{ + dice->dev_lock_changed = true; + wake_up(&dice->hwdep_wait); +} + +int snd_dice_stream_lock_try(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count < 0) { + err = -EBUSY; + goto out; + } + + if (dice->dev_lock_count++ == 0) + dice_lock_changed(dice); + err = 0; +out: + spin_unlock_irq(&dice->lock); + return err; +} + +void snd_dice_stream_lock_release(struct snd_dice *dice) +{ + spin_lock_irq(&dice->lock); + + if (WARN_ON(dice->dev_lock_count <= 0)) + goto out; + + if (--dice->dev_lock_count == 0) + dice_lock_changed(dice); +out: + spin_unlock_irq(&dice->lock); +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index dd62316..e032b1c 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -11,127 +11,62 @@ MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { - /* mode 0 */ - [0] = 32000, - [1] = 44100, - [2] = 48000, - /* mode 1 */ - [3] = 88200, - [4] = 96000, - /* mode 2 */ - [5] = 176400, - [6] = 192000, -}; - -static unsigned int rate_to_index(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) - if (snd_dice_rates[i] == rate) - return i; - - return 0; -} - -static unsigned int rate_index_to_mode(unsigned int rate_index) -{ - return ((int)rate_index - 1) / 2; -} - -static void dice_lock_changed(struct snd_dice *dice) -{ - dice->dev_lock_changed = true; - wake_up(&dice->hwdep_wait); -} - -static int dice_try_lock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count < 0) { - err = -EBUSY; - goto out; - } - - if (dice->dev_lock_count++ == 0) - dice_lock_changed(dice); - err = 0; - -out: - spin_unlock_irq(&dice->lock); - - return err; -} - -static void dice_unlock(struct snd_dice *dice) -{ - spin_lock_irq(&dice->lock); - - if (WARN_ON(dice->dev_lock_count <= 0)) - goto out; - - if (--dice->dev_lock_count == 0) - dice_lock_changed(dice); - -out: - spin_unlock_irq(&dice->lock); -} - static int dice_rate_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_dice *dice = rule->private; - const struct snd_interval *channels = + + const struct snd_interval *c = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval allowed_rates = { + struct snd_interval rates = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, mode; + unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - mode = rate_index_to_mode(i); - if ((dice->clock_caps & (1 << i)) && - snd_interval_test(channels, dice->rx_channels[mode])) { - allowed_rates.min = min(allowed_rates.min, - snd_dice_rates[i]); - allowed_rates.max = max(allowed_rates.max, - snd_dice_rates[i]); - } + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + + if (!snd_interval_test(c, pcm_channels[mode])) + continue; + + rates.min = min(rates.min, rate); + rates.max = max(rates.max, rate); } - return snd_interval_refine(rate, &allowed_rates); + return snd_interval_refine(r, &rates); } static int dice_channels_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_dice *dice = rule->private; - const struct snd_interval *rate = + + const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval allowed_channels = { + struct snd_interval channels = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, mode; + unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) - if ((dice->clock_caps & (1 << i)) && - snd_interval_test(rate, snd_dice_rates[i])) { - mode = rate_index_to_mode(i); - allowed_channels.min = min(allowed_channels.min, - dice->rx_channels[mode]); - allowed_channels.max = max(allowed_channels.max, - dice->rx_channels[mode]); - } + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; - return snd_interval_refine(channels, &allowed_channels); + if (!snd_interval_test(r, rate)) + continue; + + channels.min = min(channels.min, pcm_channels[mode]); + channels.max = max(channels.max, pcm_channels[mode]); + } + + return snd_interval_refine(c, &channels); } static int dice_open(struct snd_pcm_substream *substream) @@ -156,7 +91,7 @@ static int dice_open(struct snd_pcm_substream *substream) unsigned int i; int err; - err = dice_try_lock(dice); + err = snd_dice_stream_lock_try(dice); if (err < 0) goto error; @@ -194,7 +129,7 @@ static int dice_open(struct snd_pcm_substream *substream) return 0; err_lock: - dice_unlock(dice); + snd_dice_stream_lock_release(dice); error: return err; } @@ -203,98 +138,20 @@ static int dice_close(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; - dice_unlock(dice); + snd_dice_stream_lock_release(dice); return 0; } -static int dice_stream_start_packets(struct snd_dice *dice) -{ - int err; - - if (amdtp_stream_running(&dice->rx_stream)) - return 0; - - err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, - fw_parent_device(dice->unit)->max_speed); - if (err < 0) - return err; - - err = snd_dice_transaction_set_enable(dice); - if (err < 0) { - amdtp_stream_stop(&dice->rx_stream); - return err; - } - - return 0; -} - -static int dice_stream_start(struct snd_dice *dice) -{ - __be32 channel; - int err; - - if (!dice->rx_resources.allocated) { - err = fw_iso_resources_allocate(&dice->rx_resources, - amdtp_stream_get_max_payload(&dice->rx_stream), - fw_parent_device(dice->unit)->max_speed); - if (err < 0) - goto error; - - channel = cpu_to_be32(dice->rx_resources.channel); - err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, - &channel, 4); - if (err < 0) - goto err_resources; - } - - err = dice_stream_start_packets(dice); - if (err < 0) - goto err_rx_channel; - - return 0; - -err_rx_channel: - channel = cpu_to_be32((u32)-1); - snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); -err_resources: - fw_iso_resources_free(&dice->rx_resources); -error: - return err; -} - -static void dice_stream_stop_packets(struct snd_dice *dice) -{ - if (amdtp_stream_running(&dice->rx_stream)) { - snd_dice_transaction_clear_enable(dice); - amdtp_stream_stop(&dice->rx_stream); - } -} - -static void dice_stream_stop(struct snd_dice *dice) -{ - __be32 channel; - - dice_stream_stop_packets(dice); - - if (!dice->rx_resources.allocated) - return; - - channel = cpu_to_be32((u32)-1); - snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); - - fw_iso_resources_free(&dice->rx_resources); -} - static int dice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; - unsigned int rate_index, mode, rate, channels, i; + unsigned int mode, rate, channels, i; int err; mutex_lock(&dice->mutex); - dice_stream_stop(dice); + snd_dice_stream_stop(dice); mutex_unlock(&dice->mutex); err = snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -307,6 +164,10 @@ static int dice_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; + err = snd_dice_stream_get_rate_mode(dice, rate, &mode); + if (err < 0) + return err; + /* * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in * one data block of AMDTP packet. Thus sampling transfer frequency is @@ -318,8 +179,7 @@ static int dice_hw_params(struct snd_pcm_substream *substream, * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); - rate_index = rate_to_index(rate); - if (rate_index > 4) { + if (mode > 1) { if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { err = -ENOSYS; return err; @@ -332,10 +192,9 @@ static int dice_hw_params(struct snd_pcm_substream *substream, dice->rx_stream.double_pcm_frames = false; } - mode = rate_index_to_mode(rate_index); amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, dice->rx_midi_ports[mode]); - if (rate_index > 4) { + if (mode > 4) { channels /= 2; for (i = 0; i < channels; i++) { @@ -355,7 +214,7 @@ static int dice_hw_free(struct snd_pcm_substream *substream) struct snd_dice *dice = substream->private_data; mutex_lock(&dice->mutex); - dice_stream_stop(dice); + snd_dice_stream_stop(dice); mutex_unlock(&dice->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -369,9 +228,9 @@ static int dice_prepare(struct snd_pcm_substream *substream) mutex_lock(&dice->mutex); if (amdtp_streaming_error(&dice->rx_stream)) - dice_stream_stop_packets(dice); + snd_dice_stream_stop_packets(dice); - err = dice_stream_start(dice); + err = snd_dice_stream_start(dice); if (err < 0) { mutex_unlock(&dice->mutex); return err; @@ -966,31 +825,37 @@ end: return err; } -static int highest_supported_mode_rate(struct snd_dice *dice, unsigned int mode) +static int highest_supported_mode_rate(struct snd_dice *dice, + unsigned int mode, unsigned int *rate) { - int i; + unsigned int i, m; - for (i = ARRAY_SIZE(snd_dice_rates) - 1; i >= 0; --i) - if ((dice->clock_caps & (1 << i)) && - rate_index_to_mode(i) == mode) - return i; + for (i = ARRAY_SIZE(snd_dice_rates); i > 0; i--) { + *rate = snd_dice_rates[i - 1]; + if (snd_dice_stream_get_rate_mode(dice, *rate, &m) < 0) + continue; + if (mode == m) + break; + } + if (i == 0) + return -EINVAL; - return -1; + return 0; } static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) { __be32 values[2]; - int rate_index, err; + unsigned int rate; + int err; - rate_index = highest_supported_mode_rate(dice, mode); - if (rate_index < 0) { + if (highest_supported_mode_rate(dice, mode, &rate) < 0) { dice->rx_channels[mode] = 0; dice->rx_midi_ports[mode] = 0; return 0; } - err = snd_dice_transaction_set_rate(dice, snd_dice_rates[rate_index]); + err = snd_dice_transaction_set_rate(dice, rate); if (err < 0) return err; @@ -1113,10 +978,6 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice_card_strings(dice); - err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); - if (err < 0) - goto error; - err = dice_create_pcm(dice); if (err < 0) goto error; @@ -1127,22 +988,13 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice_create_proc(dice); - err = fw_iso_resources_init(&dice->rx_resources, unit); + err = snd_dice_stream_init(dice); if (err < 0) goto error; - dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; - - err = amdtp_stream_init(&dice->rx_stream, unit, AMDTP_OUT_STREAM, - CIP_BLOCKING); - if (err < 0) { - fw_iso_resources_destroy(&dice->rx_resources); - goto error; - } err = snd_card_register(card); if (err < 0) { - amdtp_stream_destroy(&dice->rx_stream); - fw_iso_resources_destroy(&dice->rx_resources); + snd_dice_stream_destroy(dice); goto error; } @@ -1158,13 +1010,11 @@ static void dice_remove(struct fw_unit *unit) { struct snd_dice *dice = dev_get_drvdata(&unit->device); - amdtp_stream_pcm_abort(&dice->rx_stream); - snd_card_disconnect(dice->card); mutex_lock(&dice->mutex); - dice_stream_stop(dice); + snd_dice_stream_destroy(dice); mutex_unlock(&dice->mutex); @@ -1178,23 +1028,8 @@ static void dice_bus_reset(struct fw_unit *unit) /* The handler address register becomes initialized. */ snd_dice_transaction_reinit(dice); - /* - * On a bus reset, the DICE firmware disables streaming and then goes - * off contemplating its own navel for hundreds of milliseconds before - * it can react to any of our attempts to reenable streaming. This - * means that we lose synchronization anyway, so we force our streams - * to stop so that the application can restart them in an orderly - * manner. - */ - amdtp_stream_pcm_abort(&dice->rx_stream); - mutex_lock(&dice->mutex); - - dice->global_enabled = false; - - dice_stream_stop_packets(dice); - fw_iso_resources_update(&dice->rx_resources); - + snd_dice_stream_update(dice); mutex_unlock(&dice->mutex); } diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index c756e62..ca4090d 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -157,4 +157,18 @@ void snd_dice_transaction_destroy(struct snd_dice *dice); #define SND_DICE_RATES_COUNT 7 extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; +int snd_dice_stream_get_rate_mode(struct snd_dice *dice, + unsigned int rate, unsigned int *mode); + +int snd_dice_stream_start_packets(struct snd_dice *dice); +int snd_dice_stream_start(struct snd_dice *dice); +void snd_dice_stream_stop_packets(struct snd_dice *dice); +void snd_dice_stream_stop(struct snd_dice *dice); +int snd_dice_stream_init(struct snd_dice *dice); +void snd_dice_stream_destroy(struct snd_dice *dice); +void snd_dice_stream_update(struct snd_dice *dice); + +int snd_dice_stream_lock_try(struct snd_dice *dice); +void snd_dice_stream_lock_release(struct snd_dice *dice); + #endif -- cgit v1.1 From c50fb91f53626e3bdae3ffebfee586786f970f7c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:15 +0900 Subject: ALSA: dice: Split PCM functionality into a file This commit adds a file and move some codes related to PCM functionality. Currently PCM playback is supported. PCM capture will be supported in followed commits. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-pcm.c | 296 ++++++++++++++++++++++++++++++++++++++ sound/firewire/dice/dice-stream.c | 9 +- sound/firewire/dice/dice.c | 288 +------------------------------------ sound/firewire/dice/dice.h | 2 + 5 files changed, 305 insertions(+), 292 deletions(-) create mode 100644 sound/firewire/dice/dice-pcm.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 867864c..1ddaeca 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,2 +1,2 @@ -snd-dice-objs := dice-transaction.o dice-stream.o dice.o +snd-dice-objs := dice-transaction.o dice-stream.o dice-pcm.o dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c new file mode 100644 index 0000000..deacb53 --- /dev/null +++ b/sound/firewire/dice/dice-pcm.c @@ -0,0 +1,296 @@ +/* + * dice_pcm.c - a part of driver for DICE based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "dice.h" + +static int dice_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_dice *dice = rule->private; + + const struct snd_interval *c = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval rates = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + + if (!snd_interval_test(c, pcm_channels[mode])) + continue; + + rates.min = min(rates.min, rate); + rates.max = max(rates.max, rate); + } + + return snd_interval_refine(r, &rates); +} + +static int dice_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_dice *dice = rule->private; + + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *c = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval channels = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + + if (!snd_interval_test(r, rate)) + continue; + + channels.min = min(channels.min, pcm_channels[mode]); + channels.max = max(channels.max, pcm_channels[mode]); + } + + return snd_interval_refine(c, &channels); +} + + +static int pcm_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = UINT_MAX, + .channels_max = 0, + .buffer_bytes_max = 16 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct snd_dice *dice = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int i; + int err; + + err = snd_dice_stream_lock_try(dice); + if (err < 0) + goto error; + + runtime->hw = hardware; + + for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { + if (dice->clock_caps & (1 << i)) + runtime->hw.rates |= + snd_pcm_rate_to_rate_bit(snd_dice_rates[i]); + } + snd_pcm_limit_hw_rates(runtime); + + for (i = 0; i < 3; ++i) { + if (dice->rx_channels[i]) { + runtime->hw.channels_min = min(runtime->hw.channels_min, + dice->rx_channels[i]); + runtime->hw.channels_max = max(runtime->hw.channels_max, + dice->rx_channels[i]); + } + } + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + dice_rate_constraint, dice, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + goto err_lock; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + dice_channels_constraint, dice, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto err_lock; + + err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); + if (err < 0) + goto err_lock; + + return 0; + +err_lock: + snd_dice_stream_lock_release(dice); +error: + return err; +} + +static int pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + snd_dice_stream_lock_release(dice); + + return 0; +} + +static int playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_dice *dice = substream->private_data; + unsigned int mode, rate, channels, i; + int err; + + mutex_lock(&dice->mutex); + snd_dice_stream_stop(dice); + mutex_unlock(&dice->mutex); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + rate = params_rate(hw_params); + err = snd_dice_transaction_set_rate(dice, rate); + if (err < 0) + return err; + + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + return err; + + /* + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. + */ + channels = params_channels(hw_params); + if (mode > 1) { + if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { + err = -ENOSYS; + return err; + } + + rate /= 2; + channels *= 2; + dice->rx_stream.double_pcm_frames = true; + } else { + dice->rx_stream.double_pcm_frames = false; + } + + amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, + dice->rx_midi_ports[mode]); + if (mode > 1) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->rx_stream.pcm_positions[i] = i * 2; + dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + + amdtp_stream_set_pcm_format(&dice->rx_stream, + params_format(hw_params)); + + return 0; +} + +static int playback_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + mutex_lock(&dice->mutex); + snd_dice_stream_stop(dice); + mutex_unlock(&dice->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + int err; + + mutex_lock(&dice->mutex); + + if (amdtp_streaming_error(&dice->rx_stream)) + snd_dice_stream_stop_packets(dice); + + err = snd_dice_stream_start(dice); + if (err < 0) { + mutex_unlock(&dice->mutex); + return err; + } + + mutex_unlock(&dice->mutex); + + amdtp_stream_pcm_prepare(&dice->rx_stream); + + return 0; +} + +static int playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_dice *dice = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&dice->rx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&dice->rx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + return amdtp_stream_pcm_pointer(&dice->rx_stream); +} + +int snd_dice_create_pcm(struct snd_dice *dice) +{ + static struct snd_pcm_ops playback_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = playback_hw_params, + .hw_free = playback_hw_free, + .prepare = playback_prepare, + .trigger = playback_trigger, + .pointer = playback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = dice; + strcpy(pcm->name, dice->card->shortname); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); + + return 0; +} diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index c25b9fb..4c4c4ff 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -96,10 +96,11 @@ error: void snd_dice_stream_stop_packets(struct snd_dice *dice) { - if (amdtp_stream_running(&dice->rx_stream)) { - snd_dice_transaction_clear_enable(dice); - amdtp_stream_stop(&dice->rx_stream); - } + if (!amdtp_stream_running(&dice->rx_stream)) + return; + + snd_dice_transaction_clear_enable(dice); + amdtp_stream_stop(&dice->rx_stream); } void snd_dice_stream_stop(struct snd_dice *dice) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index e032b1c..b76ed06 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -11,292 +11,6 @@ MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -static int dice_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_dice *dice = rule->private; - - const struct snd_interval *c = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *r = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval rates = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - rate = snd_dice_rates[i]; - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - continue; - - if (!snd_interval_test(c, pcm_channels[mode])) - continue; - - rates.min = min(rates.min, rate); - rates.max = max(rates.max, rate); - } - - return snd_interval_refine(r, &rates); -} - -static int dice_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_dice *dice = rule->private; - - const struct snd_interval *r = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *c = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval channels = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - rate = snd_dice_rates[i]; - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - continue; - - if (!snd_interval_test(r, rate)) - continue; - - channels.min = min(channels.min, pcm_channels[mode]); - channels.max = max(channels.max, pcm_channels[mode]); - } - - return snd_interval_refine(c, &channels); -} - -static int dice_open(struct snd_pcm_substream *substream) -{ - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = UINT_MAX, - .channels_max = 0, - .buffer_bytes_max = 16 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct snd_dice *dice = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int i; - int err; - - err = snd_dice_stream_lock_try(dice); - if (err < 0) - goto error; - - runtime->hw = hardware; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) - if (dice->clock_caps & (1 << i)) - runtime->hw.rates |= - snd_pcm_rate_to_rate_bit(snd_dice_rates[i]); - snd_pcm_limit_hw_rates(runtime); - - for (i = 0; i < 3; ++i) - if (dice->rx_channels[i]) { - runtime->hw.channels_min = min(runtime->hw.channels_min, - dice->rx_channels[i]); - runtime->hw.channels_max = max(runtime->hw.channels_max, - dice->rx_channels[i]); - } - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - dice_rate_constraint, dice, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - goto err_lock; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dice_channels_constraint, dice, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - goto err_lock; - - err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); - if (err < 0) - goto err_lock; - - return 0; - -err_lock: - snd_dice_stream_lock_release(dice); -error: - return err; -} - -static int dice_close(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - snd_dice_stream_lock_release(dice); - - return 0; -} - -static int dice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_dice *dice = substream->private_data; - unsigned int mode, rate, channels, i; - int err; - - mutex_lock(&dice->mutex); - snd_dice_stream_stop(dice); - mutex_unlock(&dice->mutex); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - - rate = params_rate(hw_params); - err = snd_dice_transaction_set_rate(dice, rate); - if (err < 0) - return err; - - err = snd_dice_stream_get_rate_mode(dice, rate, &mode); - if (err < 0) - return err; - - /* - * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in - * one data block of AMDTP packet. Thus sampling transfer frequency is - * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are - * transferred on AMDTP packets at 96 kHz. Two successive samples of a - * channel are stored consecutively in the packet. This quirk is called - * as 'Dual Wire'. - * For this quirk, blocking mode is required and PCM buffer size should - * be aligned to SYT_INTERVAL. - */ - channels = params_channels(hw_params); - if (mode > 1) { - if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { - err = -ENOSYS; - return err; - } - - rate /= 2; - channels *= 2; - dice->rx_stream.double_pcm_frames = true; - } else { - dice->rx_stream.double_pcm_frames = false; - } - - amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, - dice->rx_midi_ports[mode]); - if (mode > 4) { - channels /= 2; - - for (i = 0; i < channels; i++) { - dice->rx_stream.pcm_positions[i] = i * 2; - dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; - } - } - - amdtp_stream_set_pcm_format(&dice->rx_stream, - params_format(hw_params)); - - return 0; -} - -static int dice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - mutex_lock(&dice->mutex); - snd_dice_stream_stop(dice); - mutex_unlock(&dice->mutex); - - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -static int dice_prepare(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - int err; - - mutex_lock(&dice->mutex); - - if (amdtp_streaming_error(&dice->rx_stream)) - snd_dice_stream_stop_packets(dice); - - err = snd_dice_stream_start(dice); - if (err < 0) { - mutex_unlock(&dice->mutex); - return err; - } - - mutex_unlock(&dice->mutex); - - amdtp_stream_pcm_prepare(&dice->rx_stream); - - return 0; -} - -static int dice_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_dice *dice = substream->private_data; - struct snd_pcm_substream *pcm; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pcm = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pcm = NULL; - break; - default: - return -EINVAL; - } - amdtp_stream_pcm_trigger(&dice->rx_stream, pcm); - - return 0; -} - -static snd_pcm_uframes_t dice_pointer(struct snd_pcm_substream *substream) -{ - struct snd_dice *dice = substream->private_data; - - return amdtp_stream_pcm_pointer(&dice->rx_stream); -} - -static int dice_create_pcm(struct snd_dice *dice) -{ - static struct snd_pcm_ops ops = { - .open = dice_open, - .close = dice_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = dice_hw_params, - .hw_free = dice_hw_free, - .prepare = dice_prepare, - .trigger = dice_trigger, - .pointer = dice_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, - }; - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); - if (err < 0) - return err; - pcm->private_data = dice; - strcpy(pcm->name, dice->card->shortname); - pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->ops = &ops; - - return 0; -} - static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, loff_t *offset) { @@ -978,7 +692,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) dice_card_strings(dice); - err = dice_create_pcm(dice); + err = snd_dice_create_pcm(dice); if (err < 0) goto error; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index ca4090d..4d9e55b 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -171,4 +171,6 @@ void snd_dice_stream_update(struct snd_dice *dice); int snd_dice_stream_lock_try(struct snd_dice *dice); void snd_dice_stream_lock_release(struct snd_dice *dice); +int snd_dice_create_pcm(struct snd_dice *dice); + #endif -- cgit v1.1 From 19af57b46dda93b34902739673d5f37d8c6d0d5f Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:16 +0900 Subject: ALSA: dice: Split hwdep functionality into a file This commit adds a file and move some codes related to hwdep functionality. This interface is designed for mixer/control application. By using hwdep interface, the application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Additionally, this interface give a way to read Dice notification. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 3 +- sound/firewire/dice/dice-hwdep.c | 190 +++++++++++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 182 +------------------------------------ sound/firewire/dice/dice.h | 2 + 4 files changed, 195 insertions(+), 182 deletions(-) create mode 100644 sound/firewire/dice/dice-hwdep.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 1ddaeca..73b0e38 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,2 +1,3 @@ -snd-dice-objs := dice-transaction.o dice-stream.o dice-pcm.o dice.o +snd-dice-objs := dice-transaction.o dice-stream.o dice-pcm.o dice-hwdep.o \ + dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-hwdep.c b/sound/firewire/dice/dice-hwdep.c new file mode 100644 index 0000000..a4dc02a --- /dev/null +++ b/sound/firewire/dice/dice-hwdep.c @@ -0,0 +1,190 @@ +/* + * dice_hwdep.c - a part of driver for DICE based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "dice.h" + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, + long count, loff_t *offset) +{ + struct snd_dice *dice = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&dice->lock); + + while (!dice->dev_lock_changed && dice->notification_bits == 0) { + prepare_to_wait(&dice->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&dice->lock); + schedule(); + finish_wait(&dice->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&dice->lock); + } + + memset(&event, 0, sizeof(event)); + if (dice->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = dice->dev_lock_count > 0; + dice->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + } else { + event.dice_notification.type = + SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; + event.dice_notification.notification = dice->notification_bits; + dice->notification_bits = 0; + + count = min_t(long, count, sizeof(event.dice_notification)); + } + + spin_unlock_irq(&dice->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_dice *dice = hwdep->private_data; + unsigned int events; + + poll_wait(file, &dice->hwdep_wait, wait); + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_changed || dice->notification_bits != 0) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&dice->lock); + + return events; +} + +static int hwdep_get_info(struct snd_dice *dice, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(dice->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_DICE; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == 0) { + dice->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&dice->lock); + + return err; +} + +static int hwdep_unlock(struct snd_dice *dice) +{ + int err; + + spin_lock_irq(&dice->lock); + + if (dice->dev_lock_count == -1) { + dice->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&dice->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_dice *dice = hwdep->private_data; + + spin_lock_irq(&dice->lock); + if (dice->dev_lock_count == -1) + dice->dev_lock_count = 0; + spin_unlock_irq(&dice->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_dice *dice = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(dice, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(dice); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(dice); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +int snd_dice_create_hwdep(struct snd_dice *dice) +{ + static const struct snd_hwdep_ops ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, + }; + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(dice->card, "DICE", 0, &hwdep); + if (err < 0) + return err; + strcpy(hwdep->name, "DICE"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_DICE; + hwdep->ops = ops; + hwdep->private_data = dice; + hwdep->exclusive = true; + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index b76ed06..dbc1239 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -11,186 +11,6 @@ MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -static long dice_hwdep_read(struct snd_hwdep *hwdep, char __user *buf, - long count, loff_t *offset) -{ - struct snd_dice *dice = hwdep->private_data; - DEFINE_WAIT(wait); - union snd_firewire_event event; - - spin_lock_irq(&dice->lock); - - while (!dice->dev_lock_changed && dice->notification_bits == 0) { - prepare_to_wait(&dice->hwdep_wait, &wait, TASK_INTERRUPTIBLE); - spin_unlock_irq(&dice->lock); - schedule(); - finish_wait(&dice->hwdep_wait, &wait); - if (signal_pending(current)) - return -ERESTARTSYS; - spin_lock_irq(&dice->lock); - } - - memset(&event, 0, sizeof(event)); - if (dice->dev_lock_changed) { - event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; - event.lock_status.status = dice->dev_lock_count > 0; - dice->dev_lock_changed = false; - - count = min_t(long, count, sizeof(event.lock_status)); - } else { - event.dice_notification.type = - SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION; - event.dice_notification.notification = dice->notification_bits; - dice->notification_bits = 0; - - count = min_t(long, count, sizeof(event.dice_notification)); - } - - spin_unlock_irq(&dice->lock); - - if (copy_to_user(buf, &event, count)) - return -EFAULT; - - return count; -} - -static unsigned int dice_hwdep_poll(struct snd_hwdep *hwdep, struct file *file, - poll_table *wait) -{ - struct snd_dice *dice = hwdep->private_data; - unsigned int events; - - poll_wait(file, &dice->hwdep_wait, wait); - - spin_lock_irq(&dice->lock); - if (dice->dev_lock_changed || dice->notification_bits != 0) - events = POLLIN | POLLRDNORM; - else - events = 0; - spin_unlock_irq(&dice->lock); - - return events; -} - -static int dice_hwdep_get_info(struct snd_dice *dice, void __user *arg) -{ - struct fw_device *dev = fw_parent_device(dice->unit); - struct snd_firewire_get_info info; - - memset(&info, 0, sizeof(info)); - info.type = SNDRV_FIREWIRE_TYPE_DICE; - info.card = dev->card->index; - *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); - *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); - strlcpy(info.device_name, dev_name(&dev->device), - sizeof(info.device_name)); - - if (copy_to_user(arg, &info, sizeof(info))) - return -EFAULT; - - return 0; -} - -static int dice_hwdep_lock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count == 0) { - dice->dev_lock_count = -1; - err = 0; - } else { - err = -EBUSY; - } - - spin_unlock_irq(&dice->lock); - - return err; -} - -static int dice_hwdep_unlock(struct snd_dice *dice) -{ - int err; - - spin_lock_irq(&dice->lock); - - if (dice->dev_lock_count == -1) { - dice->dev_lock_count = 0; - err = 0; - } else { - err = -EBADFD; - } - - spin_unlock_irq(&dice->lock); - - return err; -} - -static int dice_hwdep_release(struct snd_hwdep *hwdep, struct file *file) -{ - struct snd_dice *dice = hwdep->private_data; - - spin_lock_irq(&dice->lock); - if (dice->dev_lock_count == -1) - dice->dev_lock_count = 0; - spin_unlock_irq(&dice->lock); - - return 0; -} - -static int dice_hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct snd_dice *dice = hwdep->private_data; - - switch (cmd) { - case SNDRV_FIREWIRE_IOCTL_GET_INFO: - return dice_hwdep_get_info(dice, (void __user *)arg); - case SNDRV_FIREWIRE_IOCTL_LOCK: - return dice_hwdep_lock(dice); - case SNDRV_FIREWIRE_IOCTL_UNLOCK: - return dice_hwdep_unlock(dice); - default: - return -ENOIOCTLCMD; - } -} - -#ifdef CONFIG_COMPAT -static int dice_hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, - unsigned int cmd, unsigned long arg) -{ - return dice_hwdep_ioctl(hwdep, file, cmd, - (unsigned long)compat_ptr(arg)); -} -#else -#define dice_hwdep_compat_ioctl NULL -#endif - -static int dice_create_hwdep(struct snd_dice *dice) -{ - static const struct snd_hwdep_ops ops = { - .read = dice_hwdep_read, - .release = dice_hwdep_release, - .poll = dice_hwdep_poll, - .ioctl = dice_hwdep_ioctl, - .ioctl_compat = dice_hwdep_compat_ioctl, - }; - struct snd_hwdep *hwdep; - int err; - - err = snd_hwdep_new(dice->card, "DICE", 0, &hwdep); - if (err < 0) - return err; - strcpy(hwdep->name, "DICE"); - hwdep->iface = SNDRV_HWDEP_IFACE_FW_DICE; - hwdep->ops = ops; - hwdep->private_data = dice; - hwdep->exclusive = true; - - return 0; -} - static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, unsigned int offset_q, unsigned int quadlets) { @@ -696,7 +516,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto error; - err = dice_create_hwdep(dice); + err = snd_dice_create_hwdep(dice); if (err < 0) goto error; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 4d9e55b..dcc8c78 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -173,4 +173,6 @@ void snd_dice_stream_lock_release(struct snd_dice *dice); int snd_dice_create_pcm(struct snd_dice *dice); +int snd_dice_create_hwdep(struct snd_dice *dice); + #endif -- cgit v1.1 From 04d426a039691bf114997a8af877682fdffcebd7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:17 +0900 Subject: ALSA: dice: Split proc interface into a file This commit adds a file and move some codes related to proc output. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-proc.c | 252 ++++++++++++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 244 +------------------------------------- sound/firewire/dice/dice.h | 2 + 4 files changed, 257 insertions(+), 245 deletions(-) create mode 100644 sound/firewire/dice/dice-proc.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 73b0e38..9a48289 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ -snd-dice-objs := dice-transaction.o dice-stream.o dice-pcm.o dice-hwdep.o \ - dice.o +snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-pcm.o \ + dice-hwdep.o dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c new file mode 100644 index 0000000..f5c1d1b --- /dev/null +++ b/sound/firewire/dice/dice-proc.c @@ -0,0 +1,252 @@ +/* + * dice_proc.c - a part of driver for Dice based devices + * + * Copyright (c) Clemens Ladisch + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "dice.h" + +static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, + unsigned int offset_q, unsigned int quadlets) +{ + unsigned int i; + int err; + + err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, + DICE_PRIVATE_SPACE + 4 * offset_q, + buffer, 4 * quadlets, 0); + if (err < 0) + return err; + + for (i = 0; i < quadlets; ++i) + be32_to_cpus(&((u32 *)buffer)[i]); + + return 0; +} + +static const char *str_from_array(const char *const strs[], unsigned int count, + unsigned int i) +{ + if (i < count) + return strs[i]; + + return "(unknown)"; +} + +static void dice_proc_fixup_string(char *s, unsigned int size) +{ + unsigned int i; + + for (i = 0; i < size; i += 4) + cpu_to_le32s((u32 *)(s + i)); + + for (i = 0; i < size - 2; ++i) { + if (s[i] == '\0') + return; + if (s[i] == '\\' && s[i + 1] == '\\') { + s[i + 2] = '\0'; + return; + } + } + s[size - 1] = '\0'; +} + +static void dice_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + static const char *const section_names[5] = { + "global", "tx", "rx", "ext_sync", "unused2" + }; + static const char *const clock_sources[] = { + "aes1", "aes2", "aes3", "aes4", "aes", "adat", "tdif", + "wc", "arx1", "arx2", "arx3", "arx4", "internal" + }; + static const char *const rates[] = { + "32000", "44100", "48000", "88200", "96000", "176400", "192000", + "any low", "any mid", "any high", "none" + }; + struct snd_dice *dice = entry->private_data; + u32 sections[ARRAY_SIZE(section_names) * 2]; + struct { + u32 number; + u32 size; + } tx_rx_header; + union { + struct { + u32 owner_hi, owner_lo; + u32 notification; + char nick_name[NICK_NAME_SIZE]; + u32 clock_select; + u32 enable; + u32 status; + u32 extended_status; + u32 sample_rate; + u32 version; + u32 clock_caps; + char clock_source_names[CLOCK_SOURCE_NAMES_SIZE]; + } global; + struct { + u32 iso; + u32 number_audio; + u32 number_midi; + u32 speed; + char names[TX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } tx; + struct { + u32 iso; + u32 seq_start; + u32 number_audio; + u32 number_midi; + char names[RX_NAMES_SIZE]; + u32 ac3_caps; + u32 ac3_enable; + } rx; + struct { + u32 clock_source; + u32 locked; + u32 rate; + u32 adat_user_data; + } ext_sync; + } buf; + unsigned int quadlets, stream, i; + + if (dice_proc_read_mem(dice, sections, 0, ARRAY_SIZE(sections)) < 0) + return; + snd_iprintf(buffer, "sections:\n"); + for (i = 0; i < ARRAY_SIZE(section_names); ++i) + snd_iprintf(buffer, " %s: offset %u, size %u\n", + section_names[i], + sections[i * 2], sections[i * 2 + 1]); + + quadlets = min_t(u32, sections[1], sizeof(buf.global) / 4); + if (dice_proc_read_mem(dice, &buf.global, sections[0], quadlets) < 0) + return; + snd_iprintf(buffer, "global:\n"); + snd_iprintf(buffer, " owner: %04x:%04x%08x\n", + buf.global.owner_hi >> 16, + buf.global.owner_hi & 0xffff, buf.global.owner_lo); + snd_iprintf(buffer, " notification: %08x\n", buf.global.notification); + dice_proc_fixup_string(buf.global.nick_name, NICK_NAME_SIZE); + snd_iprintf(buffer, " nick name: %s\n", buf.global.nick_name); + snd_iprintf(buffer, " clock select: %s %s\n", + str_from_array(clock_sources, ARRAY_SIZE(clock_sources), + buf.global.clock_select & CLOCK_SOURCE_MASK), + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.clock_select & CLOCK_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " enable: %u\n", buf.global.enable); + snd_iprintf(buffer, " status: %slocked %s\n", + buf.global.status & STATUS_SOURCE_LOCKED ? "" : "un", + str_from_array(rates, ARRAY_SIZE(rates), + (buf.global.status & + STATUS_NOMINAL_RATE_MASK) + >> CLOCK_RATE_SHIFT)); + snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); + snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); + snd_iprintf(buffer, " version: %u.%u.%u.%u\n", + (buf.global.version >> 24) & 0xff, + (buf.global.version >> 16) & 0xff, + (buf.global.version >> 8) & 0xff, + (buf.global.version >> 0) & 0xff); + if (quadlets >= 90) { + snd_iprintf(buffer, " clock caps:"); + for (i = 0; i <= 6; ++i) + if (buf.global.clock_caps & (1 << i)) + snd_iprintf(buffer, " %s", rates[i]); + for (i = 0; i <= 12; ++i) + if (buf.global.clock_caps & (1 << (16 + i))) + snd_iprintf(buffer, " %s", clock_sources[i]); + snd_iprintf(buffer, "\n"); + dice_proc_fixup_string(buf.global.clock_source_names, + CLOCK_SOURCE_NAMES_SIZE); + snd_iprintf(buffer, " clock source names: %s\n", + buf.global.clock_source_names); + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "tx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.tx.iso); + snd_iprintf(buffer, " audio channels: %u\n", + buf.tx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.tx.number_midi); + snd_iprintf(buffer, " speed: S%u\n", 100u << buf.tx.speed); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.tx.names, TX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.tx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.tx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.tx.ac3_enable); + } + } + + if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) + return; + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); + for (stream = 0; stream < tx_rx_header.number; ++stream) { + if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + + stream * tx_rx_header.size, + quadlets) < 0) + break; + snd_iprintf(buffer, "rx %u:\n", stream); + snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); + snd_iprintf(buffer, " audio channels: %u\n", + buf.rx.number_audio); + snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + if (quadlets >= 68) { + dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); + snd_iprintf(buffer, " names: %s\n", buf.rx.names); + } + if (quadlets >= 70) { + snd_iprintf(buffer, " ac3 caps: %08x\n", + buf.rx.ac3_caps); + snd_iprintf(buffer, " ac3 enable: %08x\n", + buf.rx.ac3_enable); + } + } + + quadlets = min_t(u32, sections[7], sizeof(buf.ext_sync) / 4); + if (quadlets >= 4) { + if (dice_proc_read_mem(dice, &buf.ext_sync, + sections[6], 4) < 0) + return; + snd_iprintf(buffer, "ext status:\n"); + snd_iprintf(buffer, " clock source: %s\n", + str_from_array(clock_sources, + ARRAY_SIZE(clock_sources), + buf.ext_sync.clock_source)); + snd_iprintf(buffer, " locked: %u\n", buf.ext_sync.locked); + snd_iprintf(buffer, " rate: %s\n", + str_from_array(rates, ARRAY_SIZE(rates), + buf.ext_sync.rate)); + snd_iprintf(buffer, " adat user data: "); + if (buf.ext_sync.adat_user_data & ADAT_USER_DATA_NO_DATA) + snd_iprintf(buffer, "-\n"); + else + snd_iprintf(buffer, "%x\n", + buf.ext_sync.adat_user_data); + } +} + +void snd_dice_create_proc(struct snd_dice *dice) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(dice->card, "dice", &entry)) + snd_info_set_text_ops(entry, dice, dice_proc_read); +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index dbc1239..8e2c172 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -11,248 +11,6 @@ MODULE_DESCRIPTION("DICE driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -static int dice_proc_read_mem(struct snd_dice *dice, void *buffer, - unsigned int offset_q, unsigned int quadlets) -{ - unsigned int i; - int err; - - err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE + 4 * offset_q, - buffer, 4 * quadlets, 0); - if (err < 0) - return err; - - for (i = 0; i < quadlets; ++i) - be32_to_cpus(&((u32 *)buffer)[i]); - - return 0; -} - -static const char *str_from_array(const char *const strs[], unsigned int count, - unsigned int i) -{ - if (i < count) - return strs[i]; - - return "(unknown)"; -} - -static void dice_proc_fixup_string(char *s, unsigned int size) -{ - unsigned int i; - - for (i = 0; i < size; i += 4) - cpu_to_le32s((u32 *)(s + i)); - - for (i = 0; i < size - 2; ++i) { - if (s[i] == '\0') - return; - if (s[i] == '\\' && s[i + 1] == '\\') { - s[i + 2] = '\0'; - return; - } - } - s[size - 1] = '\0'; -} - -static void dice_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - static const char *const section_names[5] = { - "global", "tx", "rx", "ext_sync", "unused2" - }; - static const char *const clock_sources[] = { - "aes1", "aes2", "aes3", "aes4", "aes", "adat", "tdif", - "wc", "arx1", "arx2", "arx3", "arx4", "internal" - }; - static const char *const rates[] = { - "32000", "44100", "48000", "88200", "96000", "176400", "192000", - "any low", "any mid", "any high", "none" - }; - struct snd_dice *dice = entry->private_data; - u32 sections[ARRAY_SIZE(section_names) * 2]; - struct { - u32 number; - u32 size; - } tx_rx_header; - union { - struct { - u32 owner_hi, owner_lo; - u32 notification; - char nick_name[NICK_NAME_SIZE]; - u32 clock_select; - u32 enable; - u32 status; - u32 extended_status; - u32 sample_rate; - u32 version; - u32 clock_caps; - char clock_source_names[CLOCK_SOURCE_NAMES_SIZE]; - } global; - struct { - u32 iso; - u32 number_audio; - u32 number_midi; - u32 speed; - char names[TX_NAMES_SIZE]; - u32 ac3_caps; - u32 ac3_enable; - } tx; - struct { - u32 iso; - u32 seq_start; - u32 number_audio; - u32 number_midi; - char names[RX_NAMES_SIZE]; - u32 ac3_caps; - u32 ac3_enable; - } rx; - struct { - u32 clock_source; - u32 locked; - u32 rate; - u32 adat_user_data; - } ext_sync; - } buf; - unsigned int quadlets, stream, i; - - if (dice_proc_read_mem(dice, sections, 0, ARRAY_SIZE(sections)) < 0) - return; - snd_iprintf(buffer, "sections:\n"); - for (i = 0; i < ARRAY_SIZE(section_names); ++i) - snd_iprintf(buffer, " %s: offset %u, size %u\n", - section_names[i], - sections[i * 2], sections[i * 2 + 1]); - - quadlets = min_t(u32, sections[1], sizeof(buf.global) / 4); - if (dice_proc_read_mem(dice, &buf.global, sections[0], quadlets) < 0) - return; - snd_iprintf(buffer, "global:\n"); - snd_iprintf(buffer, " owner: %04x:%04x%08x\n", - buf.global.owner_hi >> 16, - buf.global.owner_hi & 0xffff, buf.global.owner_lo); - snd_iprintf(buffer, " notification: %08x\n", buf.global.notification); - dice_proc_fixup_string(buf.global.nick_name, NICK_NAME_SIZE); - snd_iprintf(buffer, " nick name: %s\n", buf.global.nick_name); - snd_iprintf(buffer, " clock select: %s %s\n", - str_from_array(clock_sources, ARRAY_SIZE(clock_sources), - buf.global.clock_select & CLOCK_SOURCE_MASK), - str_from_array(rates, ARRAY_SIZE(rates), - (buf.global.clock_select & CLOCK_RATE_MASK) - >> CLOCK_RATE_SHIFT)); - snd_iprintf(buffer, " enable: %u\n", buf.global.enable); - snd_iprintf(buffer, " status: %slocked %s\n", - buf.global.status & STATUS_SOURCE_LOCKED ? "" : "un", - str_from_array(rates, ARRAY_SIZE(rates), - (buf.global.status & - STATUS_NOMINAL_RATE_MASK) - >> CLOCK_RATE_SHIFT)); - snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status); - snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate); - snd_iprintf(buffer, " version: %u.%u.%u.%u\n", - (buf.global.version >> 24) & 0xff, - (buf.global.version >> 16) & 0xff, - (buf.global.version >> 8) & 0xff, - (buf.global.version >> 0) & 0xff); - if (quadlets >= 90) { - snd_iprintf(buffer, " clock caps:"); - for (i = 0; i <= 6; ++i) - if (buf.global.clock_caps & (1 << i)) - snd_iprintf(buffer, " %s", rates[i]); - for (i = 0; i <= 12; ++i) - if (buf.global.clock_caps & (1 << (16 + i))) - snd_iprintf(buffer, " %s", clock_sources[i]); - snd_iprintf(buffer, "\n"); - dice_proc_fixup_string(buf.global.clock_source_names, - CLOCK_SOURCE_NAMES_SIZE); - snd_iprintf(buffer, " clock source names: %s\n", - buf.global.clock_source_names); - } - - if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) - return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); - for (stream = 0; stream < tx_rx_header.number; ++stream) { - if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + - stream * tx_rx_header.size, - quadlets) < 0) - break; - snd_iprintf(buffer, "tx %u:\n", stream); - snd_iprintf(buffer, " iso channel: %d\n", (int)buf.tx.iso); - snd_iprintf(buffer, " audio channels: %u\n", - buf.tx.number_audio); - snd_iprintf(buffer, " midi ports: %u\n", buf.tx.number_midi); - snd_iprintf(buffer, " speed: S%u\n", 100u << buf.tx.speed); - if (quadlets >= 68) { - dice_proc_fixup_string(buf.tx.names, TX_NAMES_SIZE); - snd_iprintf(buffer, " names: %s\n", buf.tx.names); - } - if (quadlets >= 70) { - snd_iprintf(buffer, " ac3 caps: %08x\n", - buf.tx.ac3_caps); - snd_iprintf(buffer, " ac3 enable: %08x\n", - buf.tx.ac3_enable); - } - } - - if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) - return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); - for (stream = 0; stream < tx_rx_header.number; ++stream) { - if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + - stream * tx_rx_header.size, - quadlets) < 0) - break; - snd_iprintf(buffer, "rx %u:\n", stream); - snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); - snd_iprintf(buffer, " audio channels: %u\n", - buf.rx.number_audio); - snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); - if (quadlets >= 68) { - dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); - snd_iprintf(buffer, " names: %s\n", buf.rx.names); - } - if (quadlets >= 70) { - snd_iprintf(buffer, " ac3 caps: %08x\n", - buf.rx.ac3_caps); - snd_iprintf(buffer, " ac3 enable: %08x\n", - buf.rx.ac3_enable); - } - } - - quadlets = min_t(u32, sections[7], sizeof(buf.ext_sync) / 4); - if (quadlets >= 4) { - if (dice_proc_read_mem(dice, &buf.ext_sync, - sections[6], 4) < 0) - return; - snd_iprintf(buffer, "ext status:\n"); - snd_iprintf(buffer, " clock source: %s\n", - str_from_array(clock_sources, - ARRAY_SIZE(clock_sources), - buf.ext_sync.clock_source)); - snd_iprintf(buffer, " locked: %u\n", buf.ext_sync.locked); - snd_iprintf(buffer, " rate: %s\n", - str_from_array(rates, ARRAY_SIZE(rates), - buf.ext_sync.rate)); - snd_iprintf(buffer, " adat user data: "); - if (buf.ext_sync.adat_user_data & ADAT_USER_DATA_NO_DATA) - snd_iprintf(buffer, "-\n"); - else - snd_iprintf(buffer, "%x\n", - buf.ext_sync.adat_user_data); - } -} - -static void dice_create_proc(struct snd_dice *dice) -{ - struct snd_info_entry *entry; - - if (!snd_card_proc_new(dice->card, "dice", &entry)) - snd_info_set_text_ops(entry, dice, dice_proc_read); -} - #define OUI_WEISS 0x001c6a #define DICE_CATEGORY_ID 0x04 @@ -520,7 +278,7 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) if (err < 0) goto error; - dice_create_proc(dice); + snd_dice_create_proc(dice); err = snd_dice_stream_init(dice); if (err < 0) diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index dcc8c78..969189a 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -175,4 +175,6 @@ int snd_dice_create_pcm(struct snd_dice *dice); int snd_dice_create_hwdep(struct snd_dice *dice); +void snd_dice_create_proc(struct snd_dice *dice); + #endif -- cgit v1.1 From 2c2416c83e7345fc51af49132c40ec7d337d4132 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:18 +0900 Subject: ALSA: dice: Add new functions for constraints of PCM parameters This commit adds a new functions and some arrangement for PCM restriction. This arrangement is due to the number of channels which each Dice device has. I note that minimum number for period becomes 2, instead of 1 because its PCM functionality has SNDRV_PCM_INFO_BATCH, this means that the driver uses double (or more) buffering so the minimum number for period should be 2. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-pcm.c | 109 ++++++++++++++++++++++++----------------- 1 file changed, 65 insertions(+), 44 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index deacb53..2e531bd 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -67,71 +67,92 @@ static int dice_channels_constraint(struct snd_pcm_hw_params *params, return snd_interval_refine(c, &channels); } - -static int pcm_open(struct snd_pcm_substream *substream) +static void limit_channels_and_rates(struct snd_dice *dice, + struct snd_pcm_runtime *runtime, + unsigned int *pcm_channels) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = UINT_MAX, - .channels_max = 0, - .buffer_bytes_max = 16 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct snd_dice *dice = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int i; - int err; + struct snd_pcm_hardware *hw = &runtime->hw; + unsigned int i, rate, mode; - err = snd_dice_stream_lock_try(dice); - if (err < 0) - goto error; - - runtime->hw = hardware; + hw->channels_min = UINT_MAX; + hw->channels_max = 0; for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - if (dice->clock_caps & (1 << i)) - runtime->hw.rates |= - snd_pcm_rate_to_rate_bit(snd_dice_rates[i]); + rate = snd_dice_rates[i]; + if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) + continue; + hw->rates |= snd_pcm_rate_to_rate_bit(rate); + + if (pcm_channels[mode] == 0) + continue; + hw->channels_min = min(hw->channels_min, pcm_channels[mode]); + hw->channels_max = max(hw->channels_max, pcm_channels[mode]); } + snd_pcm_limit_hw_rates(runtime); +} - for (i = 0; i < 3; ++i) { - if (dice->rx_channels[i]) { - runtime->hw.channels_min = min(runtime->hw.channels_min, - dice->rx_channels[i]); - runtime->hw.channels_max = max(runtime->hw.channels_max, - dice->rx_channels[i]); - } - } +static void limit_period_and_buffer(struct snd_pcm_hardware *hw) +{ + hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ + hw->periods_max = UINT_MAX; + + hw->period_bytes_min = 4 * hw->channels_max; /* byte for a frame */ + + /* Just to prevent from allocating much pages. */ + hw->period_bytes_max = hw->period_bytes_min * 2048; + hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; +} + +static int init_hw_info(struct snd_dice *dice, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hardware *hw = &runtime->hw; + int err; + + hw->info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER; + hw->formats = AMDTP_OUT_PCM_FORMAT_BITS; + + limit_channels_and_rates(dice, runtime, dice->rx_channels); + limit_period_and_buffer(hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, dice_rate_constraint, dice, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) - goto err_lock; + goto end; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dice_channels_constraint, dice, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) - goto err_lock; + goto end; err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); - if (err < 0) - goto err_lock; +end: + return err; +} - return 0; +static int pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + int err; -err_lock: + err = snd_dice_stream_lock_try(dice); + if (err < 0) + goto end; + + err = init_hw_info(dice, substream); + if (err < 0) + goto err_locked; +end: + return err; +err_locked: snd_dice_stream_lock_release(dice); -error: return err; } -- cgit v1.1 From 60cea522963fc0b3893d78e568666d105f117ff9 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:24 +0900 Subject: ALSA: dice: remove experimental state Some developers test this driver, thus it's better to remove its experimental state. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 46dff64..a433149 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -13,12 +13,12 @@ config SND_FIREWIRE_LIB select SND_RAWMIDI config SND_DICE - tristate "DICE-based DACs (EXPERIMENTAL)" + tristate "DICE-based DACs support" select SND_HWDEP select SND_FIREWIRE_LIB help Say Y here to include support for many DACs based on the DICE - chip family (DICE-II/Jr/Mini) from TC Applied Technologies. + chip family (DICE-II/Jr/Mini) which TC Applied Technologies produces. At the moment, this driver supports playback only. If you want to use devices that support capturing, use FFADO instead. -- cgit v1.1 From 8832c5a74ba3506c51b6637ac78941fcd21afbef Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:25 +0900 Subject: ALSA: speakers: Rename to oxfw and rename some members This commit renames 'firewire-speakers' to 'oxfw' to enhance support for devices which based on OXFW970/971. A line for MODULE_ALIAS is added. Additionally, to help for works in followed paches, some members in private structure are renamed. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 12 +- sound/firewire/Makefile | 4 +- sound/firewire/oxfw.c | 793 ++++++++++++++++++++++++++++++++++++++++++++++ sound/firewire/speakers.c | 792 --------------------------------------------- 4 files changed, 802 insertions(+), 799 deletions(-) create mode 100644 sound/firewire/oxfw.c delete mode 100644 sound/firewire/speakers.c diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index a433149..2a5b9a6 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -26,15 +26,17 @@ config SND_DICE To compile this driver as a module, choose M here: the module will be called snd-dice. -config SND_FIREWIRE_SPEAKERS - tristate "FireWire speakers" +config SND_OXFW + tristate "Oxford Semiconductor FW970/971 chipset support" select SND_FIREWIRE_LIB help - Say Y here to include support for the Griffin FireWave Surround - and the LaCie FireWire Speakers. + Say Y here to include support for FireWire devices based on + Oxford Semiconductor FW970/971 chipset. + * Griffin Firewave + * LaCie Firewire Speakers To compile this driver as a module, choose M here: the module - will be called snd-firewire-speakers. + will be called snd-oxfw. config SND_ISIGHT tristate "Apple iSight microphone" diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index c50761c..5ed6fb7 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,12 +1,12 @@ snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp.o -snd-firewire-speakers-objs := speakers.o +snd-oxfw-objs := oxfw.o snd-isight-objs := isight.o snd-scs1x-objs := scs1x.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o obj-$(CONFIG_SND_DICE) += dice/ -obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o +obj-$(CONFIG_SND_OXFW) += snd-oxfw.o obj-$(CONFIG_SND_ISIGHT) += snd-isight.o obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o obj-$(CONFIG_SND_FIREWORKS) += fireworks/ diff --git a/sound/firewire/oxfw.c b/sound/firewire/oxfw.c new file mode 100644 index 0000000..b7498e9 --- /dev/null +++ b/sound/firewire/oxfw.c @@ -0,0 +1,793 @@ +/* + * oxfw.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "cmp.h" +#include "fcp.h" +#include "amdtp.h" +#include "lib.h" + +#define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000) +/* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */ + +#define OXFORD_HARDWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x90020) +#define OXFORD_HARDWARE_ID_OXFW970 0x39443841 +#define OXFORD_HARDWARE_ID_OXFW971 0x39373100 + +#define VENDOR_GRIFFIN 0x001292 +#define VENDOR_LACIE 0x00d04b + +#define SPECIFIER_1394TA 0x00a02d +#define VERSION_AVC 0x010001 + +struct device_info { + const char *driver_name; + const char *short_name; + const char *long_name; + int (*pcm_constraints)(struct snd_pcm_runtime *runtime); + unsigned int mixer_channels; + u8 mute_fb_id; + u8 volume_fb_id; +}; + +struct snd_oxfw { + struct snd_card *card; + struct fw_unit *unit; + const struct device_info *device_info; + struct mutex mutex; + struct cmp_connection in_conn; + struct amdtp_stream rx_stream; + bool mute; + s16 volume[6]; + s16 volume_min; + s16 volume_max; +}; + +MODULE_DESCRIPTION("Oxford Semiconductor FW970/971 driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("snd-firewire-speakers"); + +static int firewave_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static unsigned int stereo_rates[] = { 48000, 96000 }; + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + + /* two channels work only at 48/96 kHz */ + if (snd_interval_max(channels) < 6) + return snd_interval_list(rate, 2, stereo_rates, 0); + return 0; +} + +static int firewave_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static const struct snd_interval all_channels = { .min = 6, .max = 6 }; + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + /* 32/44.1 kHz work only with all six channels */ + if (snd_interval_max(rate) < 48000) + return snd_interval_refine(channels, &all_channels); + return 0; +} + +static int firewave_constraints(struct snd_pcm_runtime *runtime) +{ + static unsigned int channels_list[] = { 2, 6 }; + static struct snd_pcm_hw_constraint_list channels_list_constraint = { + .count = 2, + .list = channels_list, + }; + int err; + + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000; + runtime->hw.channels_max = 6; + + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &channels_list_constraint); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + firewave_rate_constraint, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + firewave_channels_constraint, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + + return 0; +} + +static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) +{ + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + + return 0; +} + +static int oxfw_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 4 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + runtime->hw = hardware; + + err = oxfw->device_info->pcm_constraints(runtime); + if (err < 0) + return err; + err = snd_pcm_limit_hw_rates(runtime); + if (err < 0) + return err; + + err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); + if (err < 0) + return err; + + return 0; +} + +static int oxfw_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static void oxfw_stop_stream(struct snd_oxfw *oxfw) +{ + if (amdtp_stream_running(&oxfw->rx_stream)) { + amdtp_stream_stop(&oxfw->rx_stream); + cmp_connection_break(&oxfw->in_conn); + } +} + +static int oxfw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + goto error; + + amdtp_stream_set_parameters(&oxfw->rx_stream, + params_rate(hw_params), + params_channels(hw_params), + 0); + + amdtp_stream_set_pcm_format(&oxfw->rx_stream, + params_format(hw_params)); + + err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), + AVC_GENERAL_PLUG_DIR_IN, 0); + if (err < 0) { + dev_err(&oxfw->unit->device, "failed to set sample rate\n"); + goto err_buffer; + } + + return 0; + +err_buffer: + snd_pcm_lib_free_vmalloc_buffer(substream); +error: + return err; +} + +static int oxfw_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int oxfw_prepare(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + if (amdtp_streaming_error(&oxfw->rx_stream)) + oxfw_stop_stream(oxfw); + + if (!amdtp_stream_running(&oxfw->rx_stream)) { + err = cmp_connection_establish(&oxfw->in_conn, + amdtp_stream_get_max_payload(&oxfw->rx_stream)); + if (err < 0) + goto err_mutex; + + err = amdtp_stream_start(&oxfw->rx_stream, + oxfw->in_conn.resources.channel, + oxfw->in_conn.speed); + if (err < 0) + goto err_connection; + } + + mutex_unlock(&oxfw->mutex); + + amdtp_stream_pcm_prepare(&oxfw->rx_stream); + + return 0; + +err_connection: + cmp_connection_break(&oxfw->in_conn); +err_mutex: + mutex_unlock(&oxfw->mutex); + + return err; +} + +static int oxfw_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_stream_pcm_trigger(&oxfw->rx_stream, pcm); + return 0; +} + +static snd_pcm_uframes_t oxfw_pointer(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + return amdtp_stream_pcm_pointer(&oxfw->rx_stream); +} + +static int oxfw_create_pcm(struct snd_oxfw *oxfw) +{ + static struct snd_pcm_ops ops = { + .open = oxfw_open, + .close = oxfw_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxfw_hw_params, + .hw_free = oxfw_hw_free, + .prepare = oxfw_prepare, + .trigger = oxfw_trigger, + .pointer = oxfw_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(oxfw->card, "OXFW", 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = oxfw; + strcpy(pcm->name, oxfw->device_info->short_name); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); + return 0; +} + +enum control_action { CTL_READ, CTL_WRITE }; +enum control_attribute { + CTL_MIN = 0x02, + CTL_MAX = 0x03, + CTL_CURRENT = 0x10, +}; + +static int oxfw_mute_command(struct snd_oxfw *oxfw, bool *value, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(11, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->mute_fb_id; /* function block ID */ + buf[5] = 0x10; /* control attribute: current */ + buf[6] = 0x02; /* selector length */ + buf[7] = 0x00; /* audio channel number */ + buf[8] = 0x01; /* control selector: mute */ + buf[9] = 0x01; /* control data length */ + if (action == CTL_READ) + buf[10] = 0xff; + else + buf[10] = *value ? 0x70 : 0x60; + + err = fcp_avc_transaction(oxfw->unit, buf, 11, buf, 11, 0x3fe); + if (err < 0) + goto error; + if (err < 11) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "mute command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = buf[10] == 0x70; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_volume_command(struct snd_oxfw *oxfw, s16 *value, + unsigned int channel, + enum control_attribute attribute, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(12, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->volume_fb_id; /* function block ID */ + buf[5] = attribute; /* control attribute */ + buf[6] = 0x02; /* selector length */ + buf[7] = channel; /* audio channel number */ + buf[8] = 0x02; /* control selector: volume */ + buf[9] = 0x02; /* control data length */ + if (action == CTL_READ) { + buf[10] = 0xff; + buf[11] = 0xff; + } else { + buf[10] = *value >> 8; + buf[11] = *value; + } + + err = fcp_avc_transaction(oxfw->unit, buf, 12, buf, 12, 0x3fe); + if (err < 0) + goto error; + if (err < 12) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "volume command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = (buf[10] << 8) | buf[11]; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_mute_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + + value->value.integer.value[0] = !oxfw->mute; + + return 0; +} + +static int oxfw_mute_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + bool mute; + int err; + + mute = !value->value.integer.value[0]; + + if (mute == oxfw->mute) + return 0; + + err = oxfw_mute_command(oxfw, &mute, CTL_WRITE); + if (err < 0) + return err; + oxfw->mute = mute; + + return 1; +} + +static int oxfw_volume_info(struct snd_kcontrol *control, + struct snd_ctl_elem_info *info) +{ + struct snd_oxfw *oxfw = control->private_data; + + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = oxfw->device_info->mixer_channels; + info->value.integer.min = oxfw->volume_min; + info->value.integer.max = oxfw->volume_max; + + return 0; +} + +static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; + +static int oxfw_volume_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + value->value.integer.value[channel_map[i]] = oxfw->volume[i]; + + return 0; +} + +static int oxfw_volume_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i, changed_channels; + bool equal_values = true; + s16 volume; + int err; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + if (value->value.integer.value[i] < oxfw->volume_min || + value->value.integer.value[i] > oxfw->volume_max) + return -EINVAL; + if (value->value.integer.value[i] != + value->value.integer.value[0]) + equal_values = false; + } + + changed_channels = 0; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + if (value->value.integer.value[channel_map[i]] != + oxfw->volume[i]) + changed_channels |= 1 << (i + 1); + + if (equal_values && changed_channels != 0) + changed_channels = 1 << 0; + + for (i = 0; i <= oxfw->device_info->mixer_channels; ++i) { + volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; + if (changed_channels & (1 << i)) { + err = oxfw_volume_command(oxfw, &volume, i, + CTL_CURRENT, CTL_WRITE); + if (err < 0) + return err; + } + if (i > 0) + oxfw->volume[i - 1] = volume; + } + + return changed_channels != 0; +} + +static int oxfw_create_mixer(struct snd_oxfw *oxfw) +{ + static const struct snd_kcontrol_new controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = oxfw_mute_get, + .put = oxfw_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .info = oxfw_volume_info, + .get = oxfw_volume_get, + .put = oxfw_volume_put, + }, + }; + unsigned int i, first_ch; + int err; + + err = oxfw_volume_command(oxfw, &oxfw->volume_min, + 0, CTL_MIN, CTL_READ); + if (err < 0) + return err; + err = oxfw_volume_command(oxfw, &oxfw->volume_max, + 0, CTL_MAX, CTL_READ); + if (err < 0) + return err; + + err = oxfw_mute_command(oxfw, &oxfw->mute, CTL_READ); + if (err < 0) + return err; + + first_ch = oxfw->device_info->mixer_channels == 1 ? 0 : 1; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + err = oxfw_volume_command(oxfw, &oxfw->volume[i], + first_ch + i, CTL_CURRENT, CTL_READ); + if (err < 0) + return err; + } + + for (i = 0; i < ARRAY_SIZE(controls); ++i) { + err = snd_ctl_add(oxfw->card, + snd_ctl_new1(&controls[i], oxfw)); + if (err < 0) + return err; + } + + return 0; +} + +static u32 oxfw_read_firmware_version(struct fw_unit *unit) +{ + __be32 data; + int err; + + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); + return err >= 0 ? be32_to_cpu(data) : 0; +} + +static void oxfw_card_free(struct snd_card *card) +{ + struct snd_oxfw *oxfw = card->private_data; + + amdtp_stream_destroy(&oxfw->rx_stream); + cmp_connection_destroy(&oxfw->in_conn); + fw_unit_put(oxfw->unit); + mutex_destroy(&oxfw->mutex); +} + +static int oxfw_probe(struct fw_unit *unit, + const struct ieee1394_device_id *id) +{ + struct fw_device *fw_dev = fw_parent_device(unit); + struct snd_card *card; + struct snd_oxfw *oxfw; + u32 firmware; + int err; + + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, + sizeof(*oxfw), &card); + if (err < 0) + return err; + + oxfw = card->private_data; + oxfw->card = card; + mutex_init(&oxfw->mutex); + oxfw->unit = fw_unit_get(unit); + oxfw->device_info = (const struct device_info *)id->driver_data; + + err = cmp_connection_init(&oxfw->in_conn, unit, CMP_INPUT, 0); + if (err < 0) + goto err_unit; + + err = amdtp_stream_init(&oxfw->rx_stream, unit, AMDTP_OUT_STREAM, + CIP_NONBLOCKING); + if (err < 0) + goto err_connection; + + card->private_free = oxfw_card_free; + + strcpy(card->driver, oxfw->device_info->driver_name); + strcpy(card->shortname, oxfw->device_info->short_name); + firmware = oxfw_read_firmware_version(unit); + snprintf(card->longname, sizeof(card->longname), + "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d", + oxfw->device_info->long_name, + firmware >> 20, firmware & 0xffff, + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&unit->device), 100 << fw_dev->max_speed); + strcpy(card->mixername, "OXFW"); + + err = oxfw_create_pcm(oxfw); + if (err < 0) + goto error; + + err = oxfw_create_mixer(oxfw); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, oxfw); + + return 0; + +err_connection: + cmp_connection_destroy(&oxfw->in_conn); +err_unit: + fw_unit_put(oxfw->unit); + mutex_destroy(&oxfw->mutex); +error: + snd_card_free(card); + return err; +} + +static void oxfw_bus_reset(struct fw_unit *unit) +{ + struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + + fcp_bus_reset(oxfw->unit); + + if (cmp_connection_update(&oxfw->in_conn) < 0) { + amdtp_stream_pcm_abort(&oxfw->rx_stream); + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + return; + } + + amdtp_stream_update(&oxfw->rx_stream); +} + +static void oxfw_remove(struct fw_unit *unit) +{ + struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + + amdtp_stream_pcm_abort(&oxfw->rx_stream); + snd_card_disconnect(oxfw->card); + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + snd_card_free_when_closed(oxfw->card); +} + +static const struct device_info griffin_firewave = { + .driver_name = "FireWave", + .short_name = "FireWave", + .long_name = "Griffin FireWave Surround", + .pcm_constraints = firewave_constraints, + .mixer_channels = 6, + .mute_fb_id = 0x01, + .volume_fb_id = 0x02, +}; + +static const struct device_info lacie_speakers = { + .driver_name = "FWSpeakers", + .short_name = "FireWire Speakers", + .long_name = "LaCie FireWire Speakers", + .pcm_constraints = lacie_speakers_constraints, + .mixer_channels = 1, + .mute_fb_id = 0x01, + .volume_fb_id = 0x01, +}; + +static const struct ieee1394_device_id oxfw_id_table[] = { + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VENDOR_GRIFFIN, + .model_id = 0x00f970, + .specifier_id = SPECIFIER_1394TA, + .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&griffin_firewave, + }, + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VENDOR_LACIE, + .model_id = 0x00f970, + .specifier_id = SPECIFIER_1394TA, + .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&lacie_speakers, + }, + { } +}; +MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); + +static struct fw_driver oxfw_driver = { + .driver = { + .owner = THIS_MODULE, + .name = KBUILD_MODNAME, + .bus = &fw_bus_type, + }, + .probe = oxfw_probe, + .update = oxfw_bus_reset, + .remove = oxfw_remove, + .id_table = oxfw_id_table, +}; + +static int __init snd_oxfw_init(void) +{ + return driver_register(&oxfw_driver.driver); +} + +static void __exit snd_oxfw_exit(void) +{ + driver_unregister(&oxfw_driver.driver); +} + +module_init(snd_oxfw_init); +module_exit(snd_oxfw_exit); diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c deleted file mode 100644 index 768d40d..0000000 --- a/sound/firewire/speakers.c +++ /dev/null @@ -1,792 +0,0 @@ -/* - * OXFW970-based speakers driver - * - * Copyright (c) Clemens Ladisch - * Licensed under the terms of the GNU General Public License, version 2. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "cmp.h" -#include "fcp.h" -#include "amdtp.h" -#include "lib.h" - -#define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000) -/* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */ - -#define OXFORD_HARDWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x90020) -#define OXFORD_HARDWARE_ID_OXFW970 0x39443841 -#define OXFORD_HARDWARE_ID_OXFW971 0x39373100 - -#define VENDOR_GRIFFIN 0x001292 -#define VENDOR_LACIE 0x00d04b - -#define SPECIFIER_1394TA 0x00a02d -#define VERSION_AVC 0x010001 - -struct device_info { - const char *driver_name; - const char *short_name; - const char *long_name; - int (*pcm_constraints)(struct snd_pcm_runtime *runtime); - unsigned int mixer_channels; - u8 mute_fb_id; - u8 volume_fb_id; -}; - -struct fwspk { - struct snd_card *card; - struct fw_unit *unit; - const struct device_info *device_info; - struct mutex mutex; - struct cmp_connection connection; - struct amdtp_stream stream; - bool mute; - s16 volume[6]; - s16 volume_min; - s16 volume_max; -}; - -MODULE_DESCRIPTION("FireWire speakers driver"); -MODULE_AUTHOR("Clemens Ladisch "); -MODULE_LICENSE("GPL v2"); - -static int firewave_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static unsigned int stereo_rates[] = { 48000, 96000 }; - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - - /* two channels work only at 48/96 kHz */ - if (snd_interval_max(channels) < 6) - return snd_interval_list(rate, 2, stereo_rates, 0); - return 0; -} - -static int firewave_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static const struct snd_interval all_channels = { .min = 6, .max = 6 }; - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - - /* 32/44.1 kHz work only with all six channels */ - if (snd_interval_max(rate) < 48000) - return snd_interval_refine(channels, &all_channels); - return 0; -} - -static int firewave_constraints(struct snd_pcm_runtime *runtime) -{ - static unsigned int channels_list[] = { 2, 6 }; - static struct snd_pcm_hw_constraint_list channels_list_constraint = { - .count = 2, - .list = channels_list, - }; - int err; - - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000; - runtime->hw.channels_max = 6; - - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &channels_list_constraint); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - firewave_rate_constraint, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - firewave_channels_constraint, NULL, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - - return 0; -} - -static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) -{ - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000; - - return 0; -} - -static int fwspk_open(struct snd_pcm_substream *substream) -{ - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 4 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct fwspk *fwspk = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - runtime->hw = hardware; - - err = fwspk->device_info->pcm_constraints(runtime); - if (err < 0) - return err; - err = snd_pcm_limit_hw_rates(runtime); - if (err < 0) - return err; - - err = amdtp_stream_add_pcm_hw_constraints(&fwspk->stream, runtime); - if (err < 0) - return err; - - return 0; -} - -static int fwspk_close(struct snd_pcm_substream *substream) -{ - return 0; -} - -static void fwspk_stop_stream(struct fwspk *fwspk) -{ - if (amdtp_stream_running(&fwspk->stream)) { - amdtp_stream_stop(&fwspk->stream); - cmp_connection_break(&fwspk->connection); - } -} - -static int fwspk_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct fwspk *fwspk = substream->private_data; - int err; - - mutex_lock(&fwspk->mutex); - fwspk_stop_stream(fwspk); - mutex_unlock(&fwspk->mutex); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - goto error; - - amdtp_stream_set_parameters(&fwspk->stream, - params_rate(hw_params), - params_channels(hw_params), - 0); - - amdtp_stream_set_pcm_format(&fwspk->stream, - params_format(hw_params)); - - err = avc_general_set_sig_fmt(fwspk->unit, params_rate(hw_params), - AVC_GENERAL_PLUG_DIR_IN, 0); - if (err < 0) { - dev_err(&fwspk->unit->device, "failed to set sample rate\n"); - goto err_buffer; - } - - return 0; - -err_buffer: - snd_pcm_lib_free_vmalloc_buffer(substream); -error: - return err; -} - -static int fwspk_hw_free(struct snd_pcm_substream *substream) -{ - struct fwspk *fwspk = substream->private_data; - - mutex_lock(&fwspk->mutex); - fwspk_stop_stream(fwspk); - mutex_unlock(&fwspk->mutex); - - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -static int fwspk_prepare(struct snd_pcm_substream *substream) -{ - struct fwspk *fwspk = substream->private_data; - int err; - - mutex_lock(&fwspk->mutex); - - if (amdtp_streaming_error(&fwspk->stream)) - fwspk_stop_stream(fwspk); - - if (!amdtp_stream_running(&fwspk->stream)) { - err = cmp_connection_establish(&fwspk->connection, - amdtp_stream_get_max_payload(&fwspk->stream)); - if (err < 0) - goto err_mutex; - - err = amdtp_stream_start(&fwspk->stream, - fwspk->connection.resources.channel, - fwspk->connection.speed); - if (err < 0) - goto err_connection; - } - - mutex_unlock(&fwspk->mutex); - - amdtp_stream_pcm_prepare(&fwspk->stream); - - return 0; - -err_connection: - cmp_connection_break(&fwspk->connection); -err_mutex: - mutex_unlock(&fwspk->mutex); - - return err; -} - -static int fwspk_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct fwspk *fwspk = substream->private_data; - struct snd_pcm_substream *pcm; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pcm = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pcm = NULL; - break; - default: - return -EINVAL; - } - amdtp_stream_pcm_trigger(&fwspk->stream, pcm); - return 0; -} - -static snd_pcm_uframes_t fwspk_pointer(struct snd_pcm_substream *substream) -{ - struct fwspk *fwspk = substream->private_data; - - return amdtp_stream_pcm_pointer(&fwspk->stream); -} - -static int fwspk_create_pcm(struct fwspk *fwspk) -{ - static struct snd_pcm_ops ops = { - .open = fwspk_open, - .close = fwspk_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = fwspk_hw_params, - .hw_free = fwspk_hw_free, - .prepare = fwspk_prepare, - .trigger = fwspk_trigger, - .pointer = fwspk_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, - }; - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(fwspk->card, "OXFW970", 0, 1, 0, &pcm); - if (err < 0) - return err; - pcm->private_data = fwspk; - strcpy(pcm->name, fwspk->device_info->short_name); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); - return 0; -} - -enum control_action { CTL_READ, CTL_WRITE }; -enum control_attribute { - CTL_MIN = 0x02, - CTL_MAX = 0x03, - CTL_CURRENT = 0x10, -}; - -static int fwspk_mute_command(struct fwspk *fwspk, bool *value, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(11, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = fwspk->device_info->mute_fb_id; /* function block ID */ - buf[5] = 0x10; /* control attribute: current */ - buf[6] = 0x02; /* selector length */ - buf[7] = 0x00; /* audio channel number */ - buf[8] = 0x01; /* control selector: mute */ - buf[9] = 0x01; /* control data length */ - if (action == CTL_READ) - buf[10] = 0xff; - else - buf[10] = *value ? 0x70 : 0x60; - - err = fcp_avc_transaction(fwspk->unit, buf, 11, buf, 11, 0x3fe); - if (err < 0) - goto error; - if (err < 11) { - dev_err(&fwspk->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&fwspk->unit->device, "mute command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = buf[10] == 0x70; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int fwspk_volume_command(struct fwspk *fwspk, s16 *value, - unsigned int channel, - enum control_attribute attribute, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(12, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = fwspk->device_info->volume_fb_id; /* function block ID */ - buf[5] = attribute; /* control attribute */ - buf[6] = 0x02; /* selector length */ - buf[7] = channel; /* audio channel number */ - buf[8] = 0x02; /* control selector: volume */ - buf[9] = 0x02; /* control data length */ - if (action == CTL_READ) { - buf[10] = 0xff; - buf[11] = 0xff; - } else { - buf[10] = *value >> 8; - buf[11] = *value; - } - - err = fcp_avc_transaction(fwspk->unit, buf, 12, buf, 12, 0x3fe); - if (err < 0) - goto error; - if (err < 12) { - dev_err(&fwspk->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&fwspk->unit->device, "volume command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = (buf[10] << 8) | buf[11]; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int fwspk_mute_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct fwspk *fwspk = control->private_data; - - value->value.integer.value[0] = !fwspk->mute; - - return 0; -} - -static int fwspk_mute_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct fwspk *fwspk = control->private_data; - bool mute; - int err; - - mute = !value->value.integer.value[0]; - - if (mute == fwspk->mute) - return 0; - - err = fwspk_mute_command(fwspk, &mute, CTL_WRITE); - if (err < 0) - return err; - fwspk->mute = mute; - - return 1; -} - -static int fwspk_volume_info(struct snd_kcontrol *control, - struct snd_ctl_elem_info *info) -{ - struct fwspk *fwspk = control->private_data; - - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = fwspk->device_info->mixer_channels; - info->value.integer.min = fwspk->volume_min; - info->value.integer.max = fwspk->volume_max; - - return 0; -} - -static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; - -static int fwspk_volume_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct fwspk *fwspk = control->private_data; - unsigned int i; - - for (i = 0; i < fwspk->device_info->mixer_channels; ++i) - value->value.integer.value[channel_map[i]] = fwspk->volume[i]; - - return 0; -} - -static int fwspk_volume_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct fwspk *fwspk = control->private_data; - unsigned int i, changed_channels; - bool equal_values = true; - s16 volume; - int err; - - for (i = 0; i < fwspk->device_info->mixer_channels; ++i) { - if (value->value.integer.value[i] < fwspk->volume_min || - value->value.integer.value[i] > fwspk->volume_max) - return -EINVAL; - if (value->value.integer.value[i] != - value->value.integer.value[0]) - equal_values = false; - } - - changed_channels = 0; - for (i = 0; i < fwspk->device_info->mixer_channels; ++i) - if (value->value.integer.value[channel_map[i]] != - fwspk->volume[i]) - changed_channels |= 1 << (i + 1); - - if (equal_values && changed_channels != 0) - changed_channels = 1 << 0; - - for (i = 0; i <= fwspk->device_info->mixer_channels; ++i) { - volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; - if (changed_channels & (1 << i)) { - err = fwspk_volume_command(fwspk, &volume, i, - CTL_CURRENT, CTL_WRITE); - if (err < 0) - return err; - } - if (i > 0) - fwspk->volume[i - 1] = volume; - } - - return changed_channels != 0; -} - -static int fwspk_create_mixer(struct fwspk *fwspk) -{ - static const struct snd_kcontrol_new controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = fwspk_mute_get, - .put = fwspk_mute_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Volume", - .info = fwspk_volume_info, - .get = fwspk_volume_get, - .put = fwspk_volume_put, - }, - }; - unsigned int i, first_ch; - int err; - - err = fwspk_volume_command(fwspk, &fwspk->volume_min, - 0, CTL_MIN, CTL_READ); - if (err < 0) - return err; - err = fwspk_volume_command(fwspk, &fwspk->volume_max, - 0, CTL_MAX, CTL_READ); - if (err < 0) - return err; - - err = fwspk_mute_command(fwspk, &fwspk->mute, CTL_READ); - if (err < 0) - return err; - - first_ch = fwspk->device_info->mixer_channels == 1 ? 0 : 1; - for (i = 0; i < fwspk->device_info->mixer_channels; ++i) { - err = fwspk_volume_command(fwspk, &fwspk->volume[i], - first_ch + i, CTL_CURRENT, CTL_READ); - if (err < 0) - return err; - } - - for (i = 0; i < ARRAY_SIZE(controls); ++i) { - err = snd_ctl_add(fwspk->card, - snd_ctl_new1(&controls[i], fwspk)); - if (err < 0) - return err; - } - - return 0; -} - -static u32 fwspk_read_firmware_version(struct fw_unit *unit) -{ - __be32 data; - int err; - - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); - return err >= 0 ? be32_to_cpu(data) : 0; -} - -static void fwspk_card_free(struct snd_card *card) -{ - struct fwspk *fwspk = card->private_data; - - amdtp_stream_destroy(&fwspk->stream); - cmp_connection_destroy(&fwspk->connection); - fw_unit_put(fwspk->unit); - mutex_destroy(&fwspk->mutex); -} - -static int fwspk_probe(struct fw_unit *unit, - const struct ieee1394_device_id *id) -{ - struct fw_device *fw_dev = fw_parent_device(unit); - struct snd_card *card; - struct fwspk *fwspk; - u32 firmware; - int err; - - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*fwspk), &card); - if (err < 0) - return err; - - fwspk = card->private_data; - fwspk->card = card; - mutex_init(&fwspk->mutex); - fwspk->unit = fw_unit_get(unit); - fwspk->device_info = (const struct device_info *)id->driver_data; - - err = cmp_connection_init(&fwspk->connection, unit, CMP_INPUT, 0); - if (err < 0) - goto err_unit; - - err = amdtp_stream_init(&fwspk->stream, unit, AMDTP_OUT_STREAM, - CIP_NONBLOCKING); - if (err < 0) - goto err_connection; - - card->private_free = fwspk_card_free; - - strcpy(card->driver, fwspk->device_info->driver_name); - strcpy(card->shortname, fwspk->device_info->short_name); - firmware = fwspk_read_firmware_version(unit); - snprintf(card->longname, sizeof(card->longname), - "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d", - fwspk->device_info->long_name, - firmware >> 20, firmware & 0xffff, - fw_dev->config_rom[3], fw_dev->config_rom[4], - dev_name(&unit->device), 100 << fw_dev->max_speed); - strcpy(card->mixername, "OXFW970"); - - err = fwspk_create_pcm(fwspk); - if (err < 0) - goto error; - - err = fwspk_create_mixer(fwspk); - if (err < 0) - goto error; - - err = snd_card_register(card); - if (err < 0) - goto error; - - dev_set_drvdata(&unit->device, fwspk); - - return 0; - -err_connection: - cmp_connection_destroy(&fwspk->connection); -err_unit: - fw_unit_put(fwspk->unit); - mutex_destroy(&fwspk->mutex); -error: - snd_card_free(card); - return err; -} - -static void fwspk_bus_reset(struct fw_unit *unit) -{ - struct fwspk *fwspk = dev_get_drvdata(&unit->device); - - fcp_bus_reset(fwspk->unit); - - if (cmp_connection_update(&fwspk->connection) < 0) { - amdtp_stream_pcm_abort(&fwspk->stream); - mutex_lock(&fwspk->mutex); - fwspk_stop_stream(fwspk); - mutex_unlock(&fwspk->mutex); - return; - } - - amdtp_stream_update(&fwspk->stream); -} - -static void fwspk_remove(struct fw_unit *unit) -{ - struct fwspk *fwspk = dev_get_drvdata(&unit->device); - - amdtp_stream_pcm_abort(&fwspk->stream); - snd_card_disconnect(fwspk->card); - - mutex_lock(&fwspk->mutex); - fwspk_stop_stream(fwspk); - mutex_unlock(&fwspk->mutex); - - snd_card_free_when_closed(fwspk->card); -} - -static const struct device_info griffin_firewave = { - .driver_name = "FireWave", - .short_name = "FireWave", - .long_name = "Griffin FireWave Surround", - .pcm_constraints = firewave_constraints, - .mixer_channels = 6, - .mute_fb_id = 0x01, - .volume_fb_id = 0x02, -}; - -static const struct device_info lacie_speakers = { - .driver_name = "FWSpeakers", - .short_name = "FireWire Speakers", - .long_name = "LaCie FireWire Speakers", - .pcm_constraints = lacie_speakers_constraints, - .mixer_channels = 1, - .mute_fb_id = 0x01, - .volume_fb_id = 0x01, -}; - -static const struct ieee1394_device_id fwspk_id_table[] = { - { - .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_MODEL_ID | - IEEE1394_MATCH_SPECIFIER_ID | - IEEE1394_MATCH_VERSION, - .vendor_id = VENDOR_GRIFFIN, - .model_id = 0x00f970, - .specifier_id = SPECIFIER_1394TA, - .version = VERSION_AVC, - .driver_data = (kernel_ulong_t)&griffin_firewave, - }, - { - .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_MODEL_ID | - IEEE1394_MATCH_SPECIFIER_ID | - IEEE1394_MATCH_VERSION, - .vendor_id = VENDOR_LACIE, - .model_id = 0x00f970, - .specifier_id = SPECIFIER_1394TA, - .version = VERSION_AVC, - .driver_data = (kernel_ulong_t)&lacie_speakers, - }, - { } -}; -MODULE_DEVICE_TABLE(ieee1394, fwspk_id_table); - -static struct fw_driver fwspk_driver = { - .driver = { - .owner = THIS_MODULE, - .name = KBUILD_MODNAME, - .bus = &fw_bus_type, - }, - .probe = fwspk_probe, - .update = fwspk_bus_reset, - .remove = fwspk_remove, - .id_table = fwspk_id_table, -}; - -static int __init alsa_fwspk_init(void) -{ - return driver_register(&fwspk_driver.driver); -} - -static void __exit alsa_fwspk_exit(void) -{ - driver_unregister(&fwspk_driver.driver); -} - -module_init(alsa_fwspk_init); -module_exit(alsa_fwspk_exit); -- cgit v1.1 From 1a4e39c2e5ca2eb494a53ecd73055562f690bca0 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:26 +0900 Subject: ALSA: oxfw: Move to its own directory Followed commits add much codes. To make the work easy, this commit creates own directory and move current file to it. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/Makefile | 2 +- sound/firewire/oxfw.c | 793 ------------------------------------------- sound/firewire/oxfw/Makefile | 2 + sound/firewire/oxfw/oxfw.c | 793 +++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 796 insertions(+), 794 deletions(-) delete mode 100644 sound/firewire/oxfw.c create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 5ed6fb7..8b37f08 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -6,7 +6,7 @@ snd-scs1x-objs := scs1x.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o obj-$(CONFIG_SND_DICE) += dice/ -obj-$(CONFIG_SND_OXFW) += snd-oxfw.o +obj-$(CONFIG_SND_OXFW) += oxfw/ obj-$(CONFIG_SND_ISIGHT) += snd-isight.o obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o obj-$(CONFIG_SND_FIREWORKS) += fireworks/ diff --git a/sound/firewire/oxfw.c b/sound/firewire/oxfw.c deleted file mode 100644 index b7498e9..0000000 --- a/sound/firewire/oxfw.c +++ /dev/null @@ -1,793 +0,0 @@ -/* - * oxfw.c - a part of driver for OXFW970/971 based devices - * - * Copyright (c) Clemens Ladisch - * Licensed under the terms of the GNU General Public License, version 2. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "cmp.h" -#include "fcp.h" -#include "amdtp.h" -#include "lib.h" - -#define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000) -/* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */ - -#define OXFORD_HARDWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x90020) -#define OXFORD_HARDWARE_ID_OXFW970 0x39443841 -#define OXFORD_HARDWARE_ID_OXFW971 0x39373100 - -#define VENDOR_GRIFFIN 0x001292 -#define VENDOR_LACIE 0x00d04b - -#define SPECIFIER_1394TA 0x00a02d -#define VERSION_AVC 0x010001 - -struct device_info { - const char *driver_name; - const char *short_name; - const char *long_name; - int (*pcm_constraints)(struct snd_pcm_runtime *runtime); - unsigned int mixer_channels; - u8 mute_fb_id; - u8 volume_fb_id; -}; - -struct snd_oxfw { - struct snd_card *card; - struct fw_unit *unit; - const struct device_info *device_info; - struct mutex mutex; - struct cmp_connection in_conn; - struct amdtp_stream rx_stream; - bool mute; - s16 volume[6]; - s16 volume_min; - s16 volume_max; -}; - -MODULE_DESCRIPTION("Oxford Semiconductor FW970/971 driver"); -MODULE_AUTHOR("Clemens Ladisch "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("snd-firewire-speakers"); - -static int firewave_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static unsigned int stereo_rates[] = { 48000, 96000 }; - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - - /* two channels work only at 48/96 kHz */ - if (snd_interval_max(channels) < 6) - return snd_interval_list(rate, 2, stereo_rates, 0); - return 0; -} - -static int firewave_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static const struct snd_interval all_channels = { .min = 6, .max = 6 }; - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - - /* 32/44.1 kHz work only with all six channels */ - if (snd_interval_max(rate) < 48000) - return snd_interval_refine(channels, &all_channels); - return 0; -} - -static int firewave_constraints(struct snd_pcm_runtime *runtime) -{ - static unsigned int channels_list[] = { 2, 6 }; - static struct snd_pcm_hw_constraint_list channels_list_constraint = { - .count = 2, - .list = channels_list, - }; - int err; - - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000; - runtime->hw.channels_max = 6; - - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &channels_list_constraint); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - firewave_rate_constraint, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - firewave_channels_constraint, NULL, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - - return 0; -} - -static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) -{ - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000; - - return 0; -} - -static int oxfw_open(struct snd_pcm_substream *substream) -{ - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 4 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct snd_oxfw *oxfw = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - runtime->hw = hardware; - - err = oxfw->device_info->pcm_constraints(runtime); - if (err < 0) - return err; - err = snd_pcm_limit_hw_rates(runtime); - if (err < 0) - return err; - - err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); - if (err < 0) - return err; - - return 0; -} - -static int oxfw_close(struct snd_pcm_substream *substream) -{ - return 0; -} - -static void oxfw_stop_stream(struct snd_oxfw *oxfw) -{ - if (amdtp_stream_running(&oxfw->rx_stream)) { - amdtp_stream_stop(&oxfw->rx_stream); - cmp_connection_break(&oxfw->in_conn); - } -} - -static int oxfw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_oxfw *oxfw = substream->private_data; - int err; - - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - goto error; - - amdtp_stream_set_parameters(&oxfw->rx_stream, - params_rate(hw_params), - params_channels(hw_params), - 0); - - amdtp_stream_set_pcm_format(&oxfw->rx_stream, - params_format(hw_params)); - - err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), - AVC_GENERAL_PLUG_DIR_IN, 0); - if (err < 0) { - dev_err(&oxfw->unit->device, "failed to set sample rate\n"); - goto err_buffer; - } - - return 0; - -err_buffer: - snd_pcm_lib_free_vmalloc_buffer(substream); -error: - return err; -} - -static int oxfw_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); - - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -static int oxfw_prepare(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - int err; - - mutex_lock(&oxfw->mutex); - - if (amdtp_streaming_error(&oxfw->rx_stream)) - oxfw_stop_stream(oxfw); - - if (!amdtp_stream_running(&oxfw->rx_stream)) { - err = cmp_connection_establish(&oxfw->in_conn, - amdtp_stream_get_max_payload(&oxfw->rx_stream)); - if (err < 0) - goto err_mutex; - - err = amdtp_stream_start(&oxfw->rx_stream, - oxfw->in_conn.resources.channel, - oxfw->in_conn.speed); - if (err < 0) - goto err_connection; - } - - mutex_unlock(&oxfw->mutex); - - amdtp_stream_pcm_prepare(&oxfw->rx_stream); - - return 0; - -err_connection: - cmp_connection_break(&oxfw->in_conn); -err_mutex: - mutex_unlock(&oxfw->mutex); - - return err; -} - -static int oxfw_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_oxfw *oxfw = substream->private_data; - struct snd_pcm_substream *pcm; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pcm = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pcm = NULL; - break; - default: - return -EINVAL; - } - amdtp_stream_pcm_trigger(&oxfw->rx_stream, pcm); - return 0; -} - -static snd_pcm_uframes_t oxfw_pointer(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - - return amdtp_stream_pcm_pointer(&oxfw->rx_stream); -} - -static int oxfw_create_pcm(struct snd_oxfw *oxfw) -{ - static struct snd_pcm_ops ops = { - .open = oxfw_open, - .close = oxfw_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = oxfw_hw_params, - .hw_free = oxfw_hw_free, - .prepare = oxfw_prepare, - .trigger = oxfw_trigger, - .pointer = oxfw_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, - }; - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(oxfw->card, "OXFW", 0, 1, 0, &pcm); - if (err < 0) - return err; - pcm->private_data = oxfw; - strcpy(pcm->name, oxfw->device_info->short_name); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); - return 0; -} - -enum control_action { CTL_READ, CTL_WRITE }; -enum control_attribute { - CTL_MIN = 0x02, - CTL_MAX = 0x03, - CTL_CURRENT = 0x10, -}; - -static int oxfw_mute_command(struct snd_oxfw *oxfw, bool *value, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(11, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = oxfw->device_info->mute_fb_id; /* function block ID */ - buf[5] = 0x10; /* control attribute: current */ - buf[6] = 0x02; /* selector length */ - buf[7] = 0x00; /* audio channel number */ - buf[8] = 0x01; /* control selector: mute */ - buf[9] = 0x01; /* control data length */ - if (action == CTL_READ) - buf[10] = 0xff; - else - buf[10] = *value ? 0x70 : 0x60; - - err = fcp_avc_transaction(oxfw->unit, buf, 11, buf, 11, 0x3fe); - if (err < 0) - goto error; - if (err < 11) { - dev_err(&oxfw->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&oxfw->unit->device, "mute command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = buf[10] == 0x70; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int oxfw_volume_command(struct snd_oxfw *oxfw, s16 *value, - unsigned int channel, - enum control_attribute attribute, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(12, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = oxfw->device_info->volume_fb_id; /* function block ID */ - buf[5] = attribute; /* control attribute */ - buf[6] = 0x02; /* selector length */ - buf[7] = channel; /* audio channel number */ - buf[8] = 0x02; /* control selector: volume */ - buf[9] = 0x02; /* control data length */ - if (action == CTL_READ) { - buf[10] = 0xff; - buf[11] = 0xff; - } else { - buf[10] = *value >> 8; - buf[11] = *value; - } - - err = fcp_avc_transaction(oxfw->unit, buf, 12, buf, 12, 0x3fe); - if (err < 0) - goto error; - if (err < 12) { - dev_err(&oxfw->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&oxfw->unit->device, "volume command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = (buf[10] << 8) | buf[11]; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int oxfw_mute_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - - value->value.integer.value[0] = !oxfw->mute; - - return 0; -} - -static int oxfw_mute_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - bool mute; - int err; - - mute = !value->value.integer.value[0]; - - if (mute == oxfw->mute) - return 0; - - err = oxfw_mute_command(oxfw, &mute, CTL_WRITE); - if (err < 0) - return err; - oxfw->mute = mute; - - return 1; -} - -static int oxfw_volume_info(struct snd_kcontrol *control, - struct snd_ctl_elem_info *info) -{ - struct snd_oxfw *oxfw = control->private_data; - - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = oxfw->device_info->mixer_channels; - info->value.integer.min = oxfw->volume_min; - info->value.integer.max = oxfw->volume_max; - - return 0; -} - -static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; - -static int oxfw_volume_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - unsigned int i; - - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) - value->value.integer.value[channel_map[i]] = oxfw->volume[i]; - - return 0; -} - -static int oxfw_volume_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - unsigned int i, changed_channels; - bool equal_values = true; - s16 volume; - int err; - - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { - if (value->value.integer.value[i] < oxfw->volume_min || - value->value.integer.value[i] > oxfw->volume_max) - return -EINVAL; - if (value->value.integer.value[i] != - value->value.integer.value[0]) - equal_values = false; - } - - changed_channels = 0; - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) - if (value->value.integer.value[channel_map[i]] != - oxfw->volume[i]) - changed_channels |= 1 << (i + 1); - - if (equal_values && changed_channels != 0) - changed_channels = 1 << 0; - - for (i = 0; i <= oxfw->device_info->mixer_channels; ++i) { - volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; - if (changed_channels & (1 << i)) { - err = oxfw_volume_command(oxfw, &volume, i, - CTL_CURRENT, CTL_WRITE); - if (err < 0) - return err; - } - if (i > 0) - oxfw->volume[i - 1] = volume; - } - - return changed_channels != 0; -} - -static int oxfw_create_mixer(struct snd_oxfw *oxfw) -{ - static const struct snd_kcontrol_new controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = oxfw_mute_get, - .put = oxfw_mute_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Volume", - .info = oxfw_volume_info, - .get = oxfw_volume_get, - .put = oxfw_volume_put, - }, - }; - unsigned int i, first_ch; - int err; - - err = oxfw_volume_command(oxfw, &oxfw->volume_min, - 0, CTL_MIN, CTL_READ); - if (err < 0) - return err; - err = oxfw_volume_command(oxfw, &oxfw->volume_max, - 0, CTL_MAX, CTL_READ); - if (err < 0) - return err; - - err = oxfw_mute_command(oxfw, &oxfw->mute, CTL_READ); - if (err < 0) - return err; - - first_ch = oxfw->device_info->mixer_channels == 1 ? 0 : 1; - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { - err = oxfw_volume_command(oxfw, &oxfw->volume[i], - first_ch + i, CTL_CURRENT, CTL_READ); - if (err < 0) - return err; - } - - for (i = 0; i < ARRAY_SIZE(controls); ++i) { - err = snd_ctl_add(oxfw->card, - snd_ctl_new1(&controls[i], oxfw)); - if (err < 0) - return err; - } - - return 0; -} - -static u32 oxfw_read_firmware_version(struct fw_unit *unit) -{ - __be32 data; - int err; - - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); - return err >= 0 ? be32_to_cpu(data) : 0; -} - -static void oxfw_card_free(struct snd_card *card) -{ - struct snd_oxfw *oxfw = card->private_data; - - amdtp_stream_destroy(&oxfw->rx_stream); - cmp_connection_destroy(&oxfw->in_conn); - fw_unit_put(oxfw->unit); - mutex_destroy(&oxfw->mutex); -} - -static int oxfw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *id) -{ - struct fw_device *fw_dev = fw_parent_device(unit); - struct snd_card *card; - struct snd_oxfw *oxfw; - u32 firmware; - int err; - - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*oxfw), &card); - if (err < 0) - return err; - - oxfw = card->private_data; - oxfw->card = card; - mutex_init(&oxfw->mutex); - oxfw->unit = fw_unit_get(unit); - oxfw->device_info = (const struct device_info *)id->driver_data; - - err = cmp_connection_init(&oxfw->in_conn, unit, CMP_INPUT, 0); - if (err < 0) - goto err_unit; - - err = amdtp_stream_init(&oxfw->rx_stream, unit, AMDTP_OUT_STREAM, - CIP_NONBLOCKING); - if (err < 0) - goto err_connection; - - card->private_free = oxfw_card_free; - - strcpy(card->driver, oxfw->device_info->driver_name); - strcpy(card->shortname, oxfw->device_info->short_name); - firmware = oxfw_read_firmware_version(unit); - snprintf(card->longname, sizeof(card->longname), - "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d", - oxfw->device_info->long_name, - firmware >> 20, firmware & 0xffff, - fw_dev->config_rom[3], fw_dev->config_rom[4], - dev_name(&unit->device), 100 << fw_dev->max_speed); - strcpy(card->mixername, "OXFW"); - - err = oxfw_create_pcm(oxfw); - if (err < 0) - goto error; - - err = oxfw_create_mixer(oxfw); - if (err < 0) - goto error; - - err = snd_card_register(card); - if (err < 0) - goto error; - - dev_set_drvdata(&unit->device, oxfw); - - return 0; - -err_connection: - cmp_connection_destroy(&oxfw->in_conn); -err_unit: - fw_unit_put(oxfw->unit); - mutex_destroy(&oxfw->mutex); -error: - snd_card_free(card); - return err; -} - -static void oxfw_bus_reset(struct fw_unit *unit) -{ - struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - - fcp_bus_reset(oxfw->unit); - - if (cmp_connection_update(&oxfw->in_conn) < 0) { - amdtp_stream_pcm_abort(&oxfw->rx_stream); - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); - return; - } - - amdtp_stream_update(&oxfw->rx_stream); -} - -static void oxfw_remove(struct fw_unit *unit) -{ - struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - - amdtp_stream_pcm_abort(&oxfw->rx_stream); - snd_card_disconnect(oxfw->card); - - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); - - snd_card_free_when_closed(oxfw->card); -} - -static const struct device_info griffin_firewave = { - .driver_name = "FireWave", - .short_name = "FireWave", - .long_name = "Griffin FireWave Surround", - .pcm_constraints = firewave_constraints, - .mixer_channels = 6, - .mute_fb_id = 0x01, - .volume_fb_id = 0x02, -}; - -static const struct device_info lacie_speakers = { - .driver_name = "FWSpeakers", - .short_name = "FireWire Speakers", - .long_name = "LaCie FireWire Speakers", - .pcm_constraints = lacie_speakers_constraints, - .mixer_channels = 1, - .mute_fb_id = 0x01, - .volume_fb_id = 0x01, -}; - -static const struct ieee1394_device_id oxfw_id_table[] = { - { - .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_MODEL_ID | - IEEE1394_MATCH_SPECIFIER_ID | - IEEE1394_MATCH_VERSION, - .vendor_id = VENDOR_GRIFFIN, - .model_id = 0x00f970, - .specifier_id = SPECIFIER_1394TA, - .version = VERSION_AVC, - .driver_data = (kernel_ulong_t)&griffin_firewave, - }, - { - .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_MODEL_ID | - IEEE1394_MATCH_SPECIFIER_ID | - IEEE1394_MATCH_VERSION, - .vendor_id = VENDOR_LACIE, - .model_id = 0x00f970, - .specifier_id = SPECIFIER_1394TA, - .version = VERSION_AVC, - .driver_data = (kernel_ulong_t)&lacie_speakers, - }, - { } -}; -MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); - -static struct fw_driver oxfw_driver = { - .driver = { - .owner = THIS_MODULE, - .name = KBUILD_MODNAME, - .bus = &fw_bus_type, - }, - .probe = oxfw_probe, - .update = oxfw_bus_reset, - .remove = oxfw_remove, - .id_table = oxfw_id_table, -}; - -static int __init snd_oxfw_init(void) -{ - return driver_register(&oxfw_driver.driver); -} - -static void __exit snd_oxfw_exit(void) -{ - driver_unregister(&oxfw_driver.driver); -} - -module_init(snd_oxfw_init); -module_exit(snd_oxfw_exit); diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile new file mode 100644 index 0000000..9ca49c0 --- /dev/null +++ b/sound/firewire/oxfw/Makefile @@ -0,0 +1,2 @@ +snd-oxfw-objs := oxfw.o +obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c new file mode 100644 index 0000000..55c687e --- /dev/null +++ b/sound/firewire/oxfw/oxfw.c @@ -0,0 +1,793 @@ +/* + * oxfw.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "../cmp.h" +#include "../fcp.h" +#include "../amdtp.h" +#include "../lib.h" + +#define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000) +/* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */ + +#define OXFORD_HARDWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x90020) +#define OXFORD_HARDWARE_ID_OXFW970 0x39443841 +#define OXFORD_HARDWARE_ID_OXFW971 0x39373100 + +#define VENDOR_GRIFFIN 0x001292 +#define VENDOR_LACIE 0x00d04b + +#define SPECIFIER_1394TA 0x00a02d +#define VERSION_AVC 0x010001 + +struct device_info { + const char *driver_name; + const char *short_name; + const char *long_name; + int (*pcm_constraints)(struct snd_pcm_runtime *runtime); + unsigned int mixer_channels; + u8 mute_fb_id; + u8 volume_fb_id; +}; + +struct snd_oxfw { + struct snd_card *card; + struct fw_unit *unit; + const struct device_info *device_info; + struct mutex mutex; + struct cmp_connection in_conn; + struct amdtp_stream rx_stream; + bool mute; + s16 volume[6]; + s16 volume_min; + s16 volume_max; +}; + +MODULE_DESCRIPTION("Oxford Semiconductor FW970/971 driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("snd-firewire-speakers"); + +static int firewave_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static unsigned int stereo_rates[] = { 48000, 96000 }; + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + + /* two channels work only at 48/96 kHz */ + if (snd_interval_max(channels) < 6) + return snd_interval_list(rate, 2, stereo_rates, 0); + return 0; +} + +static int firewave_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static const struct snd_interval all_channels = { .min = 6, .max = 6 }; + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + /* 32/44.1 kHz work only with all six channels */ + if (snd_interval_max(rate) < 48000) + return snd_interval_refine(channels, &all_channels); + return 0; +} + +static int firewave_constraints(struct snd_pcm_runtime *runtime) +{ + static unsigned int channels_list[] = { 2, 6 }; + static struct snd_pcm_hw_constraint_list channels_list_constraint = { + .count = 2, + .list = channels_list, + }; + int err; + + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000; + runtime->hw.channels_max = 6; + + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &channels_list_constraint); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + firewave_rate_constraint, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + firewave_channels_constraint, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + + return 0; +} + +static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) +{ + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + + return 0; +} + +static int oxfw_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 4 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + runtime->hw = hardware; + + err = oxfw->device_info->pcm_constraints(runtime); + if (err < 0) + return err; + err = snd_pcm_limit_hw_rates(runtime); + if (err < 0) + return err; + + err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); + if (err < 0) + return err; + + return 0; +} + +static int oxfw_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static void oxfw_stop_stream(struct snd_oxfw *oxfw) +{ + if (amdtp_stream_running(&oxfw->rx_stream)) { + amdtp_stream_stop(&oxfw->rx_stream); + cmp_connection_break(&oxfw->in_conn); + } +} + +static int oxfw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + goto error; + + amdtp_stream_set_parameters(&oxfw->rx_stream, + params_rate(hw_params), + params_channels(hw_params), + 0); + + amdtp_stream_set_pcm_format(&oxfw->rx_stream, + params_format(hw_params)); + + err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), + AVC_GENERAL_PLUG_DIR_IN, 0); + if (err < 0) { + dev_err(&oxfw->unit->device, "failed to set sample rate\n"); + goto err_buffer; + } + + return 0; + +err_buffer: + snd_pcm_lib_free_vmalloc_buffer(substream); +error: + return err; +} + +static int oxfw_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int oxfw_prepare(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + if (amdtp_streaming_error(&oxfw->rx_stream)) + oxfw_stop_stream(oxfw); + + if (!amdtp_stream_running(&oxfw->rx_stream)) { + err = cmp_connection_establish(&oxfw->in_conn, + amdtp_stream_get_max_payload(&oxfw->rx_stream)); + if (err < 0) + goto err_mutex; + + err = amdtp_stream_start(&oxfw->rx_stream, + oxfw->in_conn.resources.channel, + oxfw->in_conn.speed); + if (err < 0) + goto err_connection; + } + + mutex_unlock(&oxfw->mutex); + + amdtp_stream_pcm_prepare(&oxfw->rx_stream); + + return 0; + +err_connection: + cmp_connection_break(&oxfw->in_conn); +err_mutex: + mutex_unlock(&oxfw->mutex); + + return err; +} + +static int oxfw_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_stream_pcm_trigger(&oxfw->rx_stream, pcm); + return 0; +} + +static snd_pcm_uframes_t oxfw_pointer(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + return amdtp_stream_pcm_pointer(&oxfw->rx_stream); +} + +static int oxfw_create_pcm(struct snd_oxfw *oxfw) +{ + static struct snd_pcm_ops ops = { + .open = oxfw_open, + .close = oxfw_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = oxfw_hw_params, + .hw_free = oxfw_hw_free, + .prepare = oxfw_prepare, + .trigger = oxfw_trigger, + .pointer = oxfw_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(oxfw->card, "OXFW", 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = oxfw; + strcpy(pcm->name, oxfw->device_info->short_name); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); + return 0; +} + +enum control_action { CTL_READ, CTL_WRITE }; +enum control_attribute { + CTL_MIN = 0x02, + CTL_MAX = 0x03, + CTL_CURRENT = 0x10, +}; + +static int oxfw_mute_command(struct snd_oxfw *oxfw, bool *value, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(11, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->mute_fb_id; /* function block ID */ + buf[5] = 0x10; /* control attribute: current */ + buf[6] = 0x02; /* selector length */ + buf[7] = 0x00; /* audio channel number */ + buf[8] = 0x01; /* control selector: mute */ + buf[9] = 0x01; /* control data length */ + if (action == CTL_READ) + buf[10] = 0xff; + else + buf[10] = *value ? 0x70 : 0x60; + + err = fcp_avc_transaction(oxfw->unit, buf, 11, buf, 11, 0x3fe); + if (err < 0) + goto error; + if (err < 11) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "mute command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = buf[10] == 0x70; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_volume_command(struct snd_oxfw *oxfw, s16 *value, + unsigned int channel, + enum control_attribute attribute, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(12, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->volume_fb_id; /* function block ID */ + buf[5] = attribute; /* control attribute */ + buf[6] = 0x02; /* selector length */ + buf[7] = channel; /* audio channel number */ + buf[8] = 0x02; /* control selector: volume */ + buf[9] = 0x02; /* control data length */ + if (action == CTL_READ) { + buf[10] = 0xff; + buf[11] = 0xff; + } else { + buf[10] = *value >> 8; + buf[11] = *value; + } + + err = fcp_avc_transaction(oxfw->unit, buf, 12, buf, 12, 0x3fe); + if (err < 0) + goto error; + if (err < 12) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "volume command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = (buf[10] << 8) | buf[11]; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_mute_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + + value->value.integer.value[0] = !oxfw->mute; + + return 0; +} + +static int oxfw_mute_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + bool mute; + int err; + + mute = !value->value.integer.value[0]; + + if (mute == oxfw->mute) + return 0; + + err = oxfw_mute_command(oxfw, &mute, CTL_WRITE); + if (err < 0) + return err; + oxfw->mute = mute; + + return 1; +} + +static int oxfw_volume_info(struct snd_kcontrol *control, + struct snd_ctl_elem_info *info) +{ + struct snd_oxfw *oxfw = control->private_data; + + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = oxfw->device_info->mixer_channels; + info->value.integer.min = oxfw->volume_min; + info->value.integer.max = oxfw->volume_max; + + return 0; +} + +static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; + +static int oxfw_volume_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + value->value.integer.value[channel_map[i]] = oxfw->volume[i]; + + return 0; +} + +static int oxfw_volume_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i, changed_channels; + bool equal_values = true; + s16 volume; + int err; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + if (value->value.integer.value[i] < oxfw->volume_min || + value->value.integer.value[i] > oxfw->volume_max) + return -EINVAL; + if (value->value.integer.value[i] != + value->value.integer.value[0]) + equal_values = false; + } + + changed_channels = 0; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + if (value->value.integer.value[channel_map[i]] != + oxfw->volume[i]) + changed_channels |= 1 << (i + 1); + + if (equal_values && changed_channels != 0) + changed_channels = 1 << 0; + + for (i = 0; i <= oxfw->device_info->mixer_channels; ++i) { + volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; + if (changed_channels & (1 << i)) { + err = oxfw_volume_command(oxfw, &volume, i, + CTL_CURRENT, CTL_WRITE); + if (err < 0) + return err; + } + if (i > 0) + oxfw->volume[i - 1] = volume; + } + + return changed_channels != 0; +} + +static int oxfw_create_mixer(struct snd_oxfw *oxfw) +{ + static const struct snd_kcontrol_new controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = oxfw_mute_get, + .put = oxfw_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .info = oxfw_volume_info, + .get = oxfw_volume_get, + .put = oxfw_volume_put, + }, + }; + unsigned int i, first_ch; + int err; + + err = oxfw_volume_command(oxfw, &oxfw->volume_min, + 0, CTL_MIN, CTL_READ); + if (err < 0) + return err; + err = oxfw_volume_command(oxfw, &oxfw->volume_max, + 0, CTL_MAX, CTL_READ); + if (err < 0) + return err; + + err = oxfw_mute_command(oxfw, &oxfw->mute, CTL_READ); + if (err < 0) + return err; + + first_ch = oxfw->device_info->mixer_channels == 1 ? 0 : 1; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + err = oxfw_volume_command(oxfw, &oxfw->volume[i], + first_ch + i, CTL_CURRENT, CTL_READ); + if (err < 0) + return err; + } + + for (i = 0; i < ARRAY_SIZE(controls); ++i) { + err = snd_ctl_add(oxfw->card, + snd_ctl_new1(&controls[i], oxfw)); + if (err < 0) + return err; + } + + return 0; +} + +static u32 oxfw_read_firmware_version(struct fw_unit *unit) +{ + __be32 data; + int err; + + err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, + OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); + return err >= 0 ? be32_to_cpu(data) : 0; +} + +static void oxfw_card_free(struct snd_card *card) +{ + struct snd_oxfw *oxfw = card->private_data; + + amdtp_stream_destroy(&oxfw->rx_stream); + cmp_connection_destroy(&oxfw->in_conn); + fw_unit_put(oxfw->unit); + mutex_destroy(&oxfw->mutex); +} + +static int oxfw_probe(struct fw_unit *unit, + const struct ieee1394_device_id *id) +{ + struct fw_device *fw_dev = fw_parent_device(unit); + struct snd_card *card; + struct snd_oxfw *oxfw; + u32 firmware; + int err; + + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, + sizeof(*oxfw), &card); + if (err < 0) + return err; + + oxfw = card->private_data; + oxfw->card = card; + mutex_init(&oxfw->mutex); + oxfw->unit = fw_unit_get(unit); + oxfw->device_info = (const struct device_info *)id->driver_data; + + err = cmp_connection_init(&oxfw->in_conn, unit, CMP_INPUT, 0); + if (err < 0) + goto err_unit; + + err = amdtp_stream_init(&oxfw->rx_stream, unit, AMDTP_OUT_STREAM, + CIP_NONBLOCKING); + if (err < 0) + goto err_connection; + + card->private_free = oxfw_card_free; + + strcpy(card->driver, oxfw->device_info->driver_name); + strcpy(card->shortname, oxfw->device_info->short_name); + firmware = oxfw_read_firmware_version(unit); + snprintf(card->longname, sizeof(card->longname), + "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d", + oxfw->device_info->long_name, + firmware >> 20, firmware & 0xffff, + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&unit->device), 100 << fw_dev->max_speed); + strcpy(card->mixername, "OXFW"); + + err = oxfw_create_pcm(oxfw); + if (err < 0) + goto error; + + err = oxfw_create_mixer(oxfw); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(&unit->device, oxfw); + + return 0; + +err_connection: + cmp_connection_destroy(&oxfw->in_conn); +err_unit: + fw_unit_put(oxfw->unit); + mutex_destroy(&oxfw->mutex); +error: + snd_card_free(card); + return err; +} + +static void oxfw_bus_reset(struct fw_unit *unit) +{ + struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + + fcp_bus_reset(oxfw->unit); + + if (cmp_connection_update(&oxfw->in_conn) < 0) { + amdtp_stream_pcm_abort(&oxfw->rx_stream); + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + return; + } + + amdtp_stream_update(&oxfw->rx_stream); +} + +static void oxfw_remove(struct fw_unit *unit) +{ + struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + + amdtp_stream_pcm_abort(&oxfw->rx_stream); + snd_card_disconnect(oxfw->card); + + mutex_lock(&oxfw->mutex); + oxfw_stop_stream(oxfw); + mutex_unlock(&oxfw->mutex); + + snd_card_free_when_closed(oxfw->card); +} + +static const struct device_info griffin_firewave = { + .driver_name = "FireWave", + .short_name = "FireWave", + .long_name = "Griffin FireWave Surround", + .pcm_constraints = firewave_constraints, + .mixer_channels = 6, + .mute_fb_id = 0x01, + .volume_fb_id = 0x02, +}; + +static const struct device_info lacie_speakers = { + .driver_name = "FWSpeakers", + .short_name = "FireWire Speakers", + .long_name = "LaCie FireWire Speakers", + .pcm_constraints = lacie_speakers_constraints, + .mixer_channels = 1, + .mute_fb_id = 0x01, + .volume_fb_id = 0x01, +}; + +static const struct ieee1394_device_id oxfw_id_table[] = { + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VENDOR_GRIFFIN, + .model_id = 0x00f970, + .specifier_id = SPECIFIER_1394TA, + .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&griffin_firewave, + }, + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VENDOR_LACIE, + .model_id = 0x00f970, + .specifier_id = SPECIFIER_1394TA, + .version = VERSION_AVC, + .driver_data = (kernel_ulong_t)&lacie_speakers, + }, + { } +}; +MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); + +static struct fw_driver oxfw_driver = { + .driver = { + .owner = THIS_MODULE, + .name = KBUILD_MODNAME, + .bus = &fw_bus_type, + }, + .probe = oxfw_probe, + .update = oxfw_bus_reset, + .remove = oxfw_remove, + .id_table = oxfw_id_table, +}; + +static int __init snd_oxfw_init(void) +{ + return driver_register(&oxfw_driver.driver); +} + +static void __exit snd_oxfw_exit(void) +{ + driver_unregister(&oxfw_driver.driver); +} + +module_init(snd_oxfw_init); +module_exit(snd_oxfw_exit); -- cgit v1.1 From e2786ca648d780d106bd8abca06746eb30d15ee7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:27 +0900 Subject: ALSA: oxfw: Split stream functionality to a new file and add a header file This is a help for works in followed patches. And this commit remove 'fw_unit_get()/fw_unit_put()' because these are called by helper functions in 'snd-firewire-lib'. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-stream.c | 80 +++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 130 +++++++------------------------------- sound/firewire/oxfw/oxfw.h | 56 ++++++++++++++++ 4 files changed, 159 insertions(+), 109 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.h diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index 9ca49c0..e15c4c0 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,2 +1,2 @@ -snd-oxfw-objs := oxfw.o +snd-oxfw-objs := oxfw-stream.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c new file mode 100644 index 0000000..ebd156f --- /dev/null +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -0,0 +1,80 @@ +/* + * oxfw_stream.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "oxfw.h" + +int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw) +{ + int err; + + err = cmp_connection_init(&oxfw->in_conn, oxfw->unit, + CMP_INPUT, 0); + if (err < 0) + goto end; + + err = amdtp_stream_init(&oxfw->rx_stream, oxfw->unit, + AMDTP_OUT_STREAM, CIP_NONBLOCKING); + if (err < 0) { + amdtp_stream_destroy(&oxfw->rx_stream); + cmp_connection_destroy(&oxfw->in_conn); + } +end: + return err; +} + +static void stop_stream(struct snd_oxfw *oxfw) +{ + amdtp_stream_pcm_abort(&oxfw->rx_stream); + amdtp_stream_stop(&oxfw->rx_stream); + cmp_connection_break(&oxfw->in_conn); +} + +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw) +{ + int err = 0; + + if (amdtp_streaming_error(&oxfw->rx_stream)) + stop_stream(oxfw); + + if (amdtp_stream_running(&oxfw->rx_stream)) + goto end; + + err = cmp_connection_establish(&oxfw->in_conn, + amdtp_stream_get_max_payload(&oxfw->rx_stream)); + if (err < 0) + goto end; + + err = amdtp_stream_start(&oxfw->rx_stream, + oxfw->in_conn.resources.channel, + oxfw->in_conn.speed); + if (err < 0) + stop_stream(oxfw); +end: + return err; +} + +void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw) +{ + stop_stream(oxfw); +} + +void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw) +{ + stop_stream(oxfw); + + amdtp_stream_destroy(&oxfw->rx_stream); + cmp_connection_destroy(&oxfw->in_conn); +} + +void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw) +{ + if (cmp_connection_update(&oxfw->in_conn) < 0) + stop_stream(oxfw); + else + amdtp_stream_update(&oxfw->rx_stream); +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 55c687e..4055667 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -5,22 +5,7 @@ * Licensed under the terms of the GNU General Public License, version 2. */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../cmp.h" -#include "../fcp.h" -#include "../amdtp.h" -#include "../lib.h" +#include "oxfw.h" #define OXFORD_FIRMWARE_ID_ADDRESS (CSR_REGISTER_BASE + 0x50000) /* 0x970?vvvv or 0x971?vvvv, where vvvv = firmware version */ @@ -35,29 +20,6 @@ #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 -struct device_info { - const char *driver_name; - const char *short_name; - const char *long_name; - int (*pcm_constraints)(struct snd_pcm_runtime *runtime); - unsigned int mixer_channels; - u8 mute_fb_id; - u8 volume_fb_id; -}; - -struct snd_oxfw { - struct snd_card *card; - struct fw_unit *unit; - const struct device_info *device_info; - struct mutex mutex; - struct cmp_connection in_conn; - struct amdtp_stream rx_stream; - bool mute; - s16 volume[6]; - s16 volume_min; - s16 volume_max; -}; - MODULE_DESCRIPTION("Oxford Semiconductor FW970/971 driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); @@ -180,14 +142,6 @@ static int oxfw_close(struct snd_pcm_substream *substream) return 0; } -static void oxfw_stop_stream(struct snd_oxfw *oxfw) -{ - if (amdtp_stream_running(&oxfw->rx_stream)) { - amdtp_stream_stop(&oxfw->rx_stream); - cmp_connection_break(&oxfw->in_conn); - } -} - static int oxfw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -195,8 +149,8 @@ static int oxfw_hw_params(struct snd_pcm_substream *substream, int err; mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); + + snd_oxfw_stream_stop_simplex(oxfw); err = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); @@ -223,6 +177,7 @@ static int oxfw_hw_params(struct snd_pcm_substream *substream, err_buffer: snd_pcm_lib_free_vmalloc_buffer(substream); error: + mutex_unlock(&oxfw->mutex); return err; } @@ -231,7 +186,7 @@ static int oxfw_hw_free(struct snd_pcm_substream *substream) struct snd_oxfw *oxfw = substream->private_data; mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); + snd_oxfw_stream_stop_simplex(oxfw); mutex_unlock(&oxfw->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -244,33 +199,15 @@ static int oxfw_prepare(struct snd_pcm_substream *substream) mutex_lock(&oxfw->mutex); - if (amdtp_streaming_error(&oxfw->rx_stream)) - oxfw_stop_stream(oxfw); - - if (!amdtp_stream_running(&oxfw->rx_stream)) { - err = cmp_connection_establish(&oxfw->in_conn, - amdtp_stream_get_max_payload(&oxfw->rx_stream)); - if (err < 0) - goto err_mutex; + snd_oxfw_stream_stop_simplex(oxfw); - err = amdtp_stream_start(&oxfw->rx_stream, - oxfw->in_conn.resources.channel, - oxfw->in_conn.speed); - if (err < 0) - goto err_connection; - } - - mutex_unlock(&oxfw->mutex); + err = snd_oxfw_stream_start_simplex(oxfw); + if (err < 0) + goto end; amdtp_stream_pcm_prepare(&oxfw->rx_stream); - - return 0; - -err_connection: - cmp_connection_break(&oxfw->in_conn); -err_mutex: +end: mutex_unlock(&oxfw->mutex); - return err; } @@ -615,9 +552,6 @@ static void oxfw_card_free(struct snd_card *card) { struct snd_oxfw *oxfw = card->private_data; - amdtp_stream_destroy(&oxfw->rx_stream); - cmp_connection_destroy(&oxfw->in_conn); - fw_unit_put(oxfw->unit); mutex_destroy(&oxfw->mutex); } @@ -635,23 +569,13 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) return err; + card->private_free = oxfw_card_free; oxfw = card->private_data; oxfw->card = card; mutex_init(&oxfw->mutex); - oxfw->unit = fw_unit_get(unit); + oxfw->unit = unit; oxfw->device_info = (const struct device_info *)id->driver_data; - err = cmp_connection_init(&oxfw->in_conn, unit, CMP_INPUT, 0); - if (err < 0) - goto err_unit; - - err = amdtp_stream_init(&oxfw->rx_stream, unit, AMDTP_OUT_STREAM, - CIP_NONBLOCKING); - if (err < 0) - goto err_connection; - - card->private_free = oxfw_card_free; - strcpy(card->driver, oxfw->device_info->driver_name); strcpy(card->shortname, oxfw->device_info->short_name); firmware = oxfw_read_firmware_version(unit); @@ -671,19 +595,18 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_oxfw_stream_init_simplex(oxfw); if (err < 0) goto error; + err = snd_card_register(card); + if (err < 0) { + snd_oxfw_stream_destroy_simplex(oxfw); + goto error; + } dev_set_drvdata(&unit->device, oxfw); return 0; - -err_connection: - cmp_connection_destroy(&oxfw->in_conn); -err_unit: - fw_unit_put(oxfw->unit); - mutex_destroy(&oxfw->mutex); error: snd_card_free(card); return err; @@ -695,27 +618,18 @@ static void oxfw_bus_reset(struct fw_unit *unit) fcp_bus_reset(oxfw->unit); - if (cmp_connection_update(&oxfw->in_conn) < 0) { - amdtp_stream_pcm_abort(&oxfw->rx_stream); - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); - return; - } - - amdtp_stream_update(&oxfw->rx_stream); + mutex_lock(&oxfw->mutex); + snd_oxfw_stream_update_simplex(oxfw); + mutex_unlock(&oxfw->mutex); } static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - amdtp_stream_pcm_abort(&oxfw->rx_stream); snd_card_disconnect(oxfw->card); - mutex_lock(&oxfw->mutex); - oxfw_stop_stream(oxfw); - mutex_unlock(&oxfw->mutex); + snd_oxfw_stream_destroy_simplex(oxfw); snd_card_free_when_closed(oxfw->card); } diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h new file mode 100644 index 0000000..e5836e0 --- /dev/null +++ b/sound/firewire/oxfw/oxfw.h @@ -0,0 +1,56 @@ +/* + * oxfw.h - a part of driver for OXFW970/971 based devices + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../lib.h" +#include "../fcp.h" +#include "../packets-buffer.h" +#include "../iso-resources.h" +#include "../amdtp.h" +#include "../cmp.h" + +struct device_info { + const char *driver_name; + const char *short_name; + const char *long_name; + int (*pcm_constraints)(struct snd_pcm_runtime *runtime); + unsigned int mixer_channels; + u8 mute_fb_id; + u8 volume_fb_id; +}; + +struct snd_oxfw { + struct snd_card *card; + struct fw_unit *unit; + const struct device_info *device_info; + struct mutex mutex; + struct cmp_connection in_conn; + struct amdtp_stream rx_stream; + bool mute; + s16 volume[6]; + s16 volume_min; + s16 volume_max; +}; + +int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw); +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw); +void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); +void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw); +void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); -- cgit v1.1 From 3713d93a6a12f8629c2660bb4a30d48b98105fca Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:28 +0900 Subject: ALSA: oxfw: Split PCM functionality to a new file This is a help for works in followed patches. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 249 +++++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 240 +-------------------------------------- sound/firewire/oxfw/oxfw.h | 4 + 4 files changed, 255 insertions(+), 240 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-pcm.c diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index e15c4c0..7fb4d09 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,2 +1,2 @@ -snd-oxfw-objs := oxfw-stream.o oxfw.o +snd-oxfw-objs := oxfw-stream.o oxfw-pcm.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c new file mode 100644 index 0000000..d39f17a --- /dev/null +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -0,0 +1,249 @@ +/* + * oxfw_pcm.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "oxfw.h" + +static int firewave_rate_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static unsigned int stereo_rates[] = { 48000, 96000 }; + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + + /* two channels work only at 48/96 kHz */ + if (snd_interval_max(channels) < 6) + return snd_interval_list(rate, 2, stereo_rates, 0); + return 0; +} + +static int firewave_channels_constraint(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + static const struct snd_interval all_channels = { .min = 6, .max = 6 }; + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + /* 32/44.1 kHz work only with all six channels */ + if (snd_interval_max(rate) < 48000) + return snd_interval_refine(channels, &all_channels); + return 0; +} + +int firewave_constraints(struct snd_pcm_runtime *runtime) +{ + static unsigned int channels_list[] = { 2, 6 }; + static struct snd_pcm_hw_constraint_list channels_list_constraint = { + .count = 2, + .list = channels_list, + }; + int err; + + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000; + runtime->hw.channels_max = 6; + + err = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &channels_list_constraint); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + firewave_rate_constraint, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + firewave_channels_constraint, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + return err; + + return 0; +} + +int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) +{ + runtime->hw.rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + + return 0; +} + +static int pcm_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = AMDTP_OUT_PCM_FORMAT_BITS, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 4 * 1024 * 1024, + .period_bytes_min = 1, + .period_bytes_max = UINT_MAX, + .periods_min = 1, + .periods_max = UINT_MAX, + }; + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + bool used; + int err; + + err = cmp_connection_check_used(&oxfw->in_conn, &used); + if ((err < 0) || used) + goto end; + + runtime->hw = hardware; + + err = oxfw->device_info->pcm_constraints(runtime); + if (err < 0) + goto end; + err = snd_pcm_limit_hw_rates(runtime); + if (err < 0) + goto end; + + err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); +end: + return err; +} + +static int pcm_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static int pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + snd_oxfw_stream_stop_simplex(oxfw); + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + goto error; + + amdtp_stream_set_parameters(&oxfw->rx_stream, + params_rate(hw_params), + params_channels(hw_params), + 0); + + amdtp_stream_set_pcm_format(&oxfw->rx_stream, + params_format(hw_params)); + + err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), + AVC_GENERAL_PLUG_DIR_IN, 0); + if (err < 0) { + dev_err(&oxfw->unit->device, "failed to set sample rate\n"); + goto err_buffer; + } + + return 0; + +err_buffer: + snd_pcm_lib_free_vmalloc_buffer(substream); +error: + mutex_unlock(&oxfw->mutex); + return err; +} + +static int pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + mutex_lock(&oxfw->mutex); + snd_oxfw_stream_stop_simplex(oxfw); + mutex_unlock(&oxfw->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + snd_oxfw_stream_stop_simplex(oxfw); + + err = snd_oxfw_stream_start_simplex(oxfw); + if (err < 0) + goto end; + + amdtp_stream_pcm_prepare(&oxfw->rx_stream); +end: + mutex_unlock(&oxfw->mutex); + return err; +} + +static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_stream_pcm_trigger(&oxfw->rx_stream, pcm); + return 0; +} + +static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + return amdtp_stream_pcm_pointer(&oxfw->rx_stream); +} + +int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) +{ + static struct snd_pcm_ops ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_hw_params, + .hw_free = pcm_hw_free, + .prepare = pcm_prepare, + .trigger = pcm_trigger, + .pointer = pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(oxfw->card, oxfw->card->driver, 0, 1, 0, &pcm); + if (err < 0) + return err; + pcm->private_data = oxfw; + strcpy(pcm->name, oxfw->card->shortname); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); + return 0; +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 4055667..24bb6df 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,244 +25,6 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("snd-firewire-speakers"); -static int firewave_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static unsigned int stereo_rates[] = { 48000, 96000 }; - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - - /* two channels work only at 48/96 kHz */ - if (snd_interval_max(channels) < 6) - return snd_interval_list(rate, 2, stereo_rates, 0); - return 0; -} - -static int firewave_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - static const struct snd_interval all_channels = { .min = 6, .max = 6 }; - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - - /* 32/44.1 kHz work only with all six channels */ - if (snd_interval_max(rate) < 48000) - return snd_interval_refine(channels, &all_channels); - return 0; -} - -static int firewave_constraints(struct snd_pcm_runtime *runtime) -{ - static unsigned int channels_list[] = { 2, 6 }; - static struct snd_pcm_hw_constraint_list channels_list_constraint = { - .count = 2, - .list = channels_list, - }; - int err; - - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000; - runtime->hw.channels_max = 6; - - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &channels_list_constraint); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - firewave_rate_constraint, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - firewave_channels_constraint, NULL, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; - - return 0; -} - -static int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) -{ - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000; - - return 0; -} - -static int oxfw_open(struct snd_pcm_substream *substream) -{ - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 4 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; - struct snd_oxfw *oxfw = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - runtime->hw = hardware; - - err = oxfw->device_info->pcm_constraints(runtime); - if (err < 0) - return err; - err = snd_pcm_limit_hw_rates(runtime); - if (err < 0) - return err; - - err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); - if (err < 0) - return err; - - return 0; -} - -static int oxfw_close(struct snd_pcm_substream *substream) -{ - return 0; -} - -static int oxfw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_oxfw *oxfw = substream->private_data; - int err; - - mutex_lock(&oxfw->mutex); - - snd_oxfw_stream_stop_simplex(oxfw); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - goto error; - - amdtp_stream_set_parameters(&oxfw->rx_stream, - params_rate(hw_params), - params_channels(hw_params), - 0); - - amdtp_stream_set_pcm_format(&oxfw->rx_stream, - params_format(hw_params)); - - err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), - AVC_GENERAL_PLUG_DIR_IN, 0); - if (err < 0) { - dev_err(&oxfw->unit->device, "failed to set sample rate\n"); - goto err_buffer; - } - - return 0; - -err_buffer: - snd_pcm_lib_free_vmalloc_buffer(substream); -error: - mutex_unlock(&oxfw->mutex); - return err; -} - -static int oxfw_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - - mutex_lock(&oxfw->mutex); - snd_oxfw_stream_stop_simplex(oxfw); - mutex_unlock(&oxfw->mutex); - - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -static int oxfw_prepare(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - int err; - - mutex_lock(&oxfw->mutex); - - snd_oxfw_stream_stop_simplex(oxfw); - - err = snd_oxfw_stream_start_simplex(oxfw); - if (err < 0) - goto end; - - amdtp_stream_pcm_prepare(&oxfw->rx_stream); -end: - mutex_unlock(&oxfw->mutex); - return err; -} - -static int oxfw_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_oxfw *oxfw = substream->private_data; - struct snd_pcm_substream *pcm; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - pcm = substream; - break; - case SNDRV_PCM_TRIGGER_STOP: - pcm = NULL; - break; - default: - return -EINVAL; - } - amdtp_stream_pcm_trigger(&oxfw->rx_stream, pcm); - return 0; -} - -static snd_pcm_uframes_t oxfw_pointer(struct snd_pcm_substream *substream) -{ - struct snd_oxfw *oxfw = substream->private_data; - - return amdtp_stream_pcm_pointer(&oxfw->rx_stream); -} - -static int oxfw_create_pcm(struct snd_oxfw *oxfw) -{ - static struct snd_pcm_ops ops = { - .open = oxfw_open, - .close = oxfw_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = oxfw_hw_params, - .hw_free = oxfw_hw_free, - .prepare = oxfw_prepare, - .trigger = oxfw_trigger, - .pointer = oxfw_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, - }; - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(oxfw->card, "OXFW", 0, 1, 0, &pcm); - if (err < 0) - return err; - pcm->private_data = oxfw; - strcpy(pcm->name, oxfw->device_info->short_name); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); - return 0; -} - enum control_action { CTL_READ, CTL_WRITE }; enum control_attribute { CTL_MIN = 0x02, @@ -587,7 +349,7 @@ static int oxfw_probe(struct fw_unit *unit, dev_name(&unit->device), 100 << fw_dev->max_speed); strcpy(card->mixername, "OXFW"); - err = oxfw_create_pcm(oxfw); + err = snd_oxfw_create_pcm(oxfw); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index e5836e0..1196db8 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -54,3 +54,7 @@ int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); + +int firewave_constraints(struct snd_pcm_runtime *runtime); +int lacie_speakers_constraints(struct snd_pcm_runtime *runtime); +int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); -- cgit v1.1 From 31514bfb4ab8ba6f93b5ce5fcc543cb2ac4f96e5 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 29 Nov 2014 00:59:29 +0900 Subject: ALSA: oxfw: Split control functionality to a new file This is a help for works in followed patches. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-control.c | 283 +++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 277 +----------------------------------- sound/firewire/oxfw/oxfw.h | 2 + 4 files changed, 287 insertions(+), 277 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-control.c diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index 7fb4d09..0cf48fd 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,2 +1,2 @@ -snd-oxfw-objs := oxfw-stream.o oxfw-pcm.o oxfw.o +snd-oxfw-objs := oxfw-stream.o oxfw-control.o oxfw-pcm.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-control.c b/sound/firewire/oxfw/oxfw-control.c new file mode 100644 index 0000000..02a1cb9 --- /dev/null +++ b/sound/firewire/oxfw/oxfw-control.c @@ -0,0 +1,283 @@ +/* + * oxfw_stream.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "oxfw.h" + +enum control_action { CTL_READ, CTL_WRITE }; +enum control_attribute { + CTL_MIN = 0x02, + CTL_MAX = 0x03, + CTL_CURRENT = 0x10, +}; + +static int oxfw_mute_command(struct snd_oxfw *oxfw, bool *value, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(11, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->mute_fb_id; /* function block ID */ + buf[5] = 0x10; /* control attribute: current */ + buf[6] = 0x02; /* selector length */ + buf[7] = 0x00; /* audio channel number */ + buf[8] = 0x01; /* control selector: mute */ + buf[9] = 0x01; /* control data length */ + if (action == CTL_READ) + buf[10] = 0xff; + else + buf[10] = *value ? 0x70 : 0x60; + + err = fcp_avc_transaction(oxfw->unit, buf, 11, buf, 11, 0x3fe); + if (err < 0) + goto error; + if (err < 11) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "mute command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = buf[10] == 0x70; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_volume_command(struct snd_oxfw *oxfw, s16 *value, + unsigned int channel, + enum control_attribute attribute, + enum control_action action) +{ + u8 *buf; + u8 response_ok; + int err; + + buf = kmalloc(12, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + if (action == CTL_READ) { + buf[0] = 0x01; /* AV/C, STATUS */ + response_ok = 0x0c; /* STABLE */ + } else { + buf[0] = 0x00; /* AV/C, CONTROL */ + response_ok = 0x09; /* ACCEPTED */ + } + buf[1] = 0x08; /* audio unit 0 */ + buf[2] = 0xb8; /* FUNCTION BLOCK */ + buf[3] = 0x81; /* function block type: feature */ + buf[4] = oxfw->device_info->volume_fb_id; /* function block ID */ + buf[5] = attribute; /* control attribute */ + buf[6] = 0x02; /* selector length */ + buf[7] = channel; /* audio channel number */ + buf[8] = 0x02; /* control selector: volume */ + buf[9] = 0x02; /* control data length */ + if (action == CTL_READ) { + buf[10] = 0xff; + buf[11] = 0xff; + } else { + buf[10] = *value >> 8; + buf[11] = *value; + } + + err = fcp_avc_transaction(oxfw->unit, buf, 12, buf, 12, 0x3fe); + if (err < 0) + goto error; + if (err < 12) { + dev_err(&oxfw->unit->device, "short FCP response\n"); + err = -EIO; + goto error; + } + if (buf[0] != response_ok) { + dev_err(&oxfw->unit->device, "volume command failed\n"); + err = -EIO; + goto error; + } + if (action == CTL_READ) + *value = (buf[10] << 8) | buf[11]; + + err = 0; + +error: + kfree(buf); + + return err; +} + +static int oxfw_mute_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + + value->value.integer.value[0] = !oxfw->mute; + + return 0; +} + +static int oxfw_mute_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + bool mute; + int err; + + mute = !value->value.integer.value[0]; + + if (mute == oxfw->mute) + return 0; + + err = oxfw_mute_command(oxfw, &mute, CTL_WRITE); + if (err < 0) + return err; + oxfw->mute = mute; + + return 1; +} + +static int oxfw_volume_info(struct snd_kcontrol *control, + struct snd_ctl_elem_info *info) +{ + struct snd_oxfw *oxfw = control->private_data; + + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = oxfw->device_info->mixer_channels; + info->value.integer.min = oxfw->volume_min; + info->value.integer.max = oxfw->volume_max; + + return 0; +} + +static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; + +static int oxfw_volume_get(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + value->value.integer.value[channel_map[i]] = oxfw->volume[i]; + + return 0; +} + +static int oxfw_volume_put(struct snd_kcontrol *control, + struct snd_ctl_elem_value *value) +{ + struct snd_oxfw *oxfw = control->private_data; + unsigned int i, changed_channels; + bool equal_values = true; + s16 volume; + int err; + + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + if (value->value.integer.value[i] < oxfw->volume_min || + value->value.integer.value[i] > oxfw->volume_max) + return -EINVAL; + if (value->value.integer.value[i] != + value->value.integer.value[0]) + equal_values = false; + } + + changed_channels = 0; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) + if (value->value.integer.value[channel_map[i]] != + oxfw->volume[i]) + changed_channels |= 1 << (i + 1); + + if (equal_values && changed_channels != 0) + changed_channels = 1 << 0; + + for (i = 0; i <= oxfw->device_info->mixer_channels; ++i) { + volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; + if (changed_channels & (1 << i)) { + err = oxfw_volume_command(oxfw, &volume, i, + CTL_CURRENT, CTL_WRITE); + if (err < 0) + return err; + } + if (i > 0) + oxfw->volume[i - 1] = volume; + } + + return changed_channels != 0; +} + +int snd_oxfw_create_mixer(struct snd_oxfw *oxfw) +{ + static const struct snd_kcontrol_new controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = oxfw_mute_get, + .put = oxfw_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .info = oxfw_volume_info, + .get = oxfw_volume_get, + .put = oxfw_volume_put, + }, + }; + unsigned int i, first_ch; + int err; + + err = oxfw_volume_command(oxfw, &oxfw->volume_min, + 0, CTL_MIN, CTL_READ); + if (err < 0) + return err; + err = oxfw_volume_command(oxfw, &oxfw->volume_max, + 0, CTL_MAX, CTL_READ); + if (err < 0) + return err; + + err = oxfw_mute_command(oxfw, &oxfw->mute, CTL_READ); + if (err < 0) + return err; + + first_ch = oxfw->device_info->mixer_channels == 1 ? 0 : 1; + for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { + err = oxfw_volume_command(oxfw, &oxfw->volume[i], + first_ch + i, CTL_CURRENT, CTL_READ); + if (err < 0) + return err; + } + + for (i = 0; i < ARRAY_SIZE(controls); ++i) { + err = snd_ctl_add(oxfw->card, + snd_ctl_new1(&controls[i], oxfw)); + if (err < 0) + return err; + } + + return 0; +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 24bb6df..951d9a4 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,281 +25,6 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("snd-firewire-speakers"); -enum control_action { CTL_READ, CTL_WRITE }; -enum control_attribute { - CTL_MIN = 0x02, - CTL_MAX = 0x03, - CTL_CURRENT = 0x10, -}; - -static int oxfw_mute_command(struct snd_oxfw *oxfw, bool *value, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(11, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = oxfw->device_info->mute_fb_id; /* function block ID */ - buf[5] = 0x10; /* control attribute: current */ - buf[6] = 0x02; /* selector length */ - buf[7] = 0x00; /* audio channel number */ - buf[8] = 0x01; /* control selector: mute */ - buf[9] = 0x01; /* control data length */ - if (action == CTL_READ) - buf[10] = 0xff; - else - buf[10] = *value ? 0x70 : 0x60; - - err = fcp_avc_transaction(oxfw->unit, buf, 11, buf, 11, 0x3fe); - if (err < 0) - goto error; - if (err < 11) { - dev_err(&oxfw->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&oxfw->unit->device, "mute command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = buf[10] == 0x70; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int oxfw_volume_command(struct snd_oxfw *oxfw, s16 *value, - unsigned int channel, - enum control_attribute attribute, - enum control_action action) -{ - u8 *buf; - u8 response_ok; - int err; - - buf = kmalloc(12, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - if (action == CTL_READ) { - buf[0] = 0x01; /* AV/C, STATUS */ - response_ok = 0x0c; /* STABLE */ - } else { - buf[0] = 0x00; /* AV/C, CONTROL */ - response_ok = 0x09; /* ACCEPTED */ - } - buf[1] = 0x08; /* audio unit 0 */ - buf[2] = 0xb8; /* FUNCTION BLOCK */ - buf[3] = 0x81; /* function block type: feature */ - buf[4] = oxfw->device_info->volume_fb_id; /* function block ID */ - buf[5] = attribute; /* control attribute */ - buf[6] = 0x02; /* selector length */ - buf[7] = channel; /* audio channel number */ - buf[8] = 0x02; /* control selector: volume */ - buf[9] = 0x02; /* control data length */ - if (action == CTL_READ) { - buf[10] = 0xff; - buf[11] = 0xff; - } else { - buf[10] = *value >> 8; - buf[11] = *value; - } - - err = fcp_avc_transaction(oxfw->unit, buf, 12, buf, 12, 0x3fe); - if (err < 0) - goto error; - if (err < 12) { - dev_err(&oxfw->unit->device, "short FCP response\n"); - err = -EIO; - goto error; - } - if (buf[0] != response_ok) { - dev_err(&oxfw->unit->device, "volume command failed\n"); - err = -EIO; - goto error; - } - if (action == CTL_READ) - *value = (buf[10] << 8) | buf[11]; - - err = 0; - -error: - kfree(buf); - - return err; -} - -static int oxfw_mute_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - - value->value.integer.value[0] = !oxfw->mute; - - return 0; -} - -static int oxfw_mute_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - bool mute; - int err; - - mute = !value->value.integer.value[0]; - - if (mute == oxfw->mute) - return 0; - - err = oxfw_mute_command(oxfw, &mute, CTL_WRITE); - if (err < 0) - return err; - oxfw->mute = mute; - - return 1; -} - -static int oxfw_volume_info(struct snd_kcontrol *control, - struct snd_ctl_elem_info *info) -{ - struct snd_oxfw *oxfw = control->private_data; - - info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = oxfw->device_info->mixer_channels; - info->value.integer.min = oxfw->volume_min; - info->value.integer.max = oxfw->volume_max; - - return 0; -} - -static const u8 channel_map[6] = { 0, 1, 4, 5, 2, 3 }; - -static int oxfw_volume_get(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - unsigned int i; - - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) - value->value.integer.value[channel_map[i]] = oxfw->volume[i]; - - return 0; -} - -static int oxfw_volume_put(struct snd_kcontrol *control, - struct snd_ctl_elem_value *value) -{ - struct snd_oxfw *oxfw = control->private_data; - unsigned int i, changed_channels; - bool equal_values = true; - s16 volume; - int err; - - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { - if (value->value.integer.value[i] < oxfw->volume_min || - value->value.integer.value[i] > oxfw->volume_max) - return -EINVAL; - if (value->value.integer.value[i] != - value->value.integer.value[0]) - equal_values = false; - } - - changed_channels = 0; - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) - if (value->value.integer.value[channel_map[i]] != - oxfw->volume[i]) - changed_channels |= 1 << (i + 1); - - if (equal_values && changed_channels != 0) - changed_channels = 1 << 0; - - for (i = 0; i <= oxfw->device_info->mixer_channels; ++i) { - volume = value->value.integer.value[channel_map[i ? i - 1 : 0]]; - if (changed_channels & (1 << i)) { - err = oxfw_volume_command(oxfw, &volume, i, - CTL_CURRENT, CTL_WRITE); - if (err < 0) - return err; - } - if (i > 0) - oxfw->volume[i - 1] = volume; - } - - return changed_channels != 0; -} - -static int oxfw_create_mixer(struct snd_oxfw *oxfw) -{ - static const struct snd_kcontrol_new controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = oxfw_mute_get, - .put = oxfw_mute_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Volume", - .info = oxfw_volume_info, - .get = oxfw_volume_get, - .put = oxfw_volume_put, - }, - }; - unsigned int i, first_ch; - int err; - - err = oxfw_volume_command(oxfw, &oxfw->volume_min, - 0, CTL_MIN, CTL_READ); - if (err < 0) - return err; - err = oxfw_volume_command(oxfw, &oxfw->volume_max, - 0, CTL_MAX, CTL_READ); - if (err < 0) - return err; - - err = oxfw_mute_command(oxfw, &oxfw->mute, CTL_READ); - if (err < 0) - return err; - - first_ch = oxfw->device_info->mixer_channels == 1 ? 0 : 1; - for (i = 0; i < oxfw->device_info->mixer_channels; ++i) { - err = oxfw_volume_command(oxfw, &oxfw->volume[i], - first_ch + i, CTL_CURRENT, CTL_READ); - if (err < 0) - return err; - } - - for (i = 0; i < ARRAY_SIZE(controls); ++i) { - err = snd_ctl_add(oxfw->card, - snd_ctl_new1(&controls[i], oxfw)); - if (err < 0) - return err; - } - - return 0; -} - static u32 oxfw_read_firmware_version(struct fw_unit *unit) { __be32 data; @@ -353,7 +78,7 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = oxfw_create_mixer(oxfw); + err = snd_oxfw_create_mixer(oxfw); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 1196db8..6164bf3 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -58,3 +58,5 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); int firewave_constraints(struct snd_pcm_runtime *runtime); int lacie_speakers_constraints(struct snd_pcm_runtime *runtime); int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); + +int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); -- cgit v1.1 From 316638a5030a04bb3259dcbca0632281001a4b24 Mon Sep 17 00:00:00 2001 From: Kyle Chamberlin Date: Fri, 28 Nov 2014 13:59:56 -0500 Subject: ALSA: virmidi: fixed code style issues Fixed some minor code style issues and also removed some assignments inside of if conditionals. Signed-off-by: Kyle Chamberlin Signed-off-by: Takashi Iwai --- sound/drivers/virmidi.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index b17872429..0af88d9 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -99,30 +99,33 @@ static int snd_virmidi_probe(struct platform_device *devptr) if (midi_devs[dev] > MAX_MIDI_DEVICES) { snd_printk(KERN_WARNING - "too much midi devices for virmidi %d: " - "force to use %d\n", dev, MAX_MIDI_DEVICES); + "too much midi devices for virmidi %d: force to use %d\n", + dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { struct snd_rawmidi *rmidi; struct snd_virmidi_dev *rdev; - if ((err = snd_virmidi_new(card, idx, &rmidi)) < 0) + + err = snd_virmidi_new(card, idx, &rmidi); + if (err < 0) goto __nodev; rdev = rmidi->private_data; vmidi->midi[idx] = rmidi; strcpy(rmidi->name, "Virtual Raw MIDI"); rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH; } - + strcpy(card->driver, "VirMIDI"); strcpy(card->shortname, "VirMIDI"); sprintf(card->longname, "Virtual MIDI Card %i", dev + 1); - if ((err = snd_card_register(card)) == 0) { + err = snd_card_register(card); + if (err) { platform_set_drvdata(devptr, card); return 0; } - __nodev: +__nodev: snd_card_free(card); return err; } @@ -157,13 +160,15 @@ static int __init alsa_card_virmidi_init(void) { int i, cards, err; - if ((err = platform_driver_register(&snd_virmidi_driver)) < 0) + err = platform_driver_register(&snd_virmidi_driver); + if (err < 0) return err; cards = 0; for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (! enable[i]) + + if (!enable[i]) continue; device = platform_device_register_simple(SND_VIRMIDI_DRIVER, i, NULL, 0); -- cgit v1.1 From fc5d3c70f97ba54bf3d3b585912b0a76a78fb718 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 30 Nov 2014 20:13:16 +0100 Subject: ALSA: virmidi: Fix wrong error check While rewriting the code in the previous commit [316638a5030a: ALSA: virmidi: fixed code style issues], the error check was wrongly converted. This resulted an Oops. Fixes: 316638a5030a ('ALSA: virmidi: fixed code style issues') Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/drivers/virmidi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 0af88d9..d28d870 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -121,7 +121,7 @@ static int snd_virmidi_probe(struct platform_device *devptr) sprintf(card->longname, "Virtual MIDI Card %i", dev + 1); err = snd_card_register(card); - if (err) { + if (!err) { platform_set_drvdata(devptr, card); return 0; } -- cgit v1.1 From dacacb0aa0cb6fdeb69313db6acfc82456945d7e Mon Sep 17 00:00:00 2001 From: Panu Matilainen Date: Sun, 30 Nov 2014 18:45:40 +0200 Subject: ALSA: usb-audio: Add support for Zoom R16/24 capture and midi interfaces This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Mixer descriptors are garbage and will require further quirks to enable functionality, also disabled here. Signed-off-by: Panu Matilainen Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 48 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 013cba8..73d2ba4 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3175,6 +3175,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * ZOOM R16/24 in audio interface mode. + * Mixer descriptors are garbage, further quirks will be needed + * to make any of it functional, thus disabled for now. + * Playback stream appears to start and run fine but no sound + * is produced, so also disabled for now. + */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Mixer */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e45cc3a..4dbfb3d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1207,6 +1207,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- cgit v1.1 From 1689092da0815997eeded3f4c7b35be3df444df5 Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Mon, 1 Dec 2014 16:57:52 +0100 Subject: ASoC: snd-soc-afeb9260: delete driver as board has just been removed During the removal of all AT91 !DT boards, we removed the AFEB9260. This driver is !DT and was only used by this board, so we delete it as well. Reported-by: Paul Bolle Signed-off-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/Makefile | 1 - sound/soc/atmel/snd-soc-afeb9260.c | 151 ------------------------------------- 2 files changed, 152 deletions(-) delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 5baabc8..466a821 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c deleted file mode 100644 index 9579799..0000000 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * afeb9260.c -- SoC audio for AFEB9260 - * - * Copyright (C) 2009 Sergey Lapin - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include - -#include "../codecs/tlv320aic23.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - -#define CODEC_CLOCK 12000000 - -static int afeb9260_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int err; - - /* Set the codec system clock for DAC and ADC */ - err = - snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - - if (err < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return err; - } - - return err; -} - -static struct snd_soc_ops afeb9260_ops = { - .hw_params = afeb9260_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route afeb9260_audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link afeb9260_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "atmel-ssc-dai.0", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &afeb9260_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_machine_afeb9260 = { - .name = "AFEB9260", - .owner = THIS_MODULE, - .dai_link = &afeb9260_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = afeb9260_audio_map, - .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), -}; - -static struct platform_device *afeb9260_snd_device; - -static int __init afeb9260_soc_init(void) -{ - int err; - struct device *dev; - - if (!(machine_is_afeb9260())) - return -ENODEV; - - - afeb9260_snd_device = platform_device_alloc("soc-audio", -1); - if (!afeb9260_snd_device) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); - err = platform_device_add(afeb9260_snd_device); - if (err) - goto err1; - - dev = &afeb9260_snd_device->dev; - - return 0; -err1: - platform_device_put(afeb9260_snd_device); - return err; -} - -static void __exit afeb9260_soc_exit(void) -{ - platform_device_unregister(afeb9260_snd_device); -} - -module_init(afeb9260_soc_init); -module_exit(afeb9260_soc_exit); - -MODULE_AUTHOR("Sergey Lapin "); -MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); -MODULE_LICENSE("GPL"); - -- cgit v1.1 From ea09dd3b006b26002802b7b534bdb6531429c8ff Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Tue, 2 Dec 2014 04:03:16 +0800 Subject: ALSA: dice: fix semicolon.cocci warnings sound/firewire/dice/dice-transaction.c:34:2-3: Unneeded semicolon Removes unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci CC: Takashi Sakamoto Signed-off-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-transaction.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index a9b98e0..1fe304c 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -31,7 +31,7 @@ static u64 get_subaddr(struct snd_dice *dice, enum snd_dice_addr_type type, default: offset += dice->global_offset; break; - }; + } offset += DICE_PRIVATE_SPACE; return offset; } -- cgit v1.1 From de82bf6c055e1f426845e7e140c7d48dd70b6d16 Mon Sep 17 00:00:00 2001 From: Nicolas Ferre Date: Tue, 2 Dec 2014 10:04:35 +0100 Subject: ASoC: Kconfig: remove not used SND_AT91_SOC_AFEB9260 option Now that the driver snd-soc-afeb9260.c is deleted, remove its Kconfig option. Reported-by: Paul Bolle Signed-off-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 27e3fc4..fb38783 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. - -config SND_AT91_SOC_AFEB9260 - tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC - select SND_ATMEL_SOC_PDC - select SND_ATMEL_SOC_SSC - select SND_SOC_TLV320AIC23_I2C - help - Say Y here to support sound on AFEB9260 board. -- cgit v1.1 From 87164cc5723329565089a999b6671bd214caf0a0 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 18:05:32 +0100 Subject: ALSA: asihpi: Deletion of an unnecessary check before the function call "vfree" The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpioctl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index e457eb8..6aa677e 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -547,8 +547,7 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) if (pa->irq) free_irq(pa->irq, pa); - if (pa->p_buffer) - vfree(pa->p_buffer); + vfree(pa->p_buffer); if (1) dev_info(&pci_dev->dev, -- cgit v1.1 From 057a4a55e703038d22bc9f2bcf8b02dc35850e16 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 18:34:45 +0100 Subject: ALSA: echoaudio: Deletion of a check before release_and_free_resource() The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 60e4003..21228ad 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1875,8 +1875,7 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->dsp_registers) iounmap(chip->dsp_registers); - if (chip->iores) - release_and_free_resource(chip->iores); + release_and_free_resource(chip->iores); pci_disable_device(chip->pci); -- cgit v1.1 From 5c34fdf48b9522ca87372b1fae19de8f93ffd130 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 18:52:21 +0100 Subject: ALSA: trident: Deletion of a check before snd_util_memhdr_free() The snd_util_memhdr_free() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/trident/trident_main.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index da875dc..57cd757 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3702,8 +3702,7 @@ static int snd_trident_free(struct snd_trident *trident) free_irq(trident->irq, trident); if (trident->tlb.buffer.area) { outl(0, TRID_REG(trident, NX_TLBC)); - if (trident->tlb.memhdr) - snd_util_memhdr_free(trident->tlb.memhdr); + snd_util_memhdr_free(trident->tlb.memhdr); if (trident->tlb.silent_page.area) snd_dma_free_pages(&trident->tlb.silent_page); vfree(trident->tlb.shadow_entries); -- cgit v1.1 From 492a7ea0a6d3cc9c16ad5f8af832ed4278225b2e Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 22:50:24 +0100 Subject: ALSA: i2sbus: Deletion of unnecessary checks before the function call "release_and_free_resource" The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index a80d5ea..4e2b4fb 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -79,8 +79,7 @@ static void i2sbus_release_dev(struct device *dev) if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma); if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) - if (i2sdev->allocated_resource[i]) - release_and_free_resource(i2sdev->allocated_resource[i]); + release_and_free_resource(i2sdev->allocated_resource[i]); free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring); free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) @@ -323,8 +322,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (dev->out.dbdma) iounmap(dev->out.dbdma); if (dev->in.dbdma) iounmap(dev->in.dbdma); for (i=0;i<3;i++) - if (dev->allocated_resource[i]) - release_and_free_resource(dev->allocated_resource[i]); + release_and_free_resource(dev->allocated_resource[i]); mutex_destroy(&dev->lock); kfree(dev); return 0; -- cgit v1.1 From b42b4afb7482f1c079c82af824a7fe750590f438 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Dec 2014 09:47:20 +0100 Subject: ALSA: hda - Define the DCAPS preset for the old Intel chipsets Just for improving readability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fc7aff0..53e43d1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -276,6 +276,10 @@ enum { (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) #define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) +/* quirks for old Intel chipsets */ +#define AZX_DCAPS_INTEL_ICH \ + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ @@ -2054,31 +2058,30 @@ static const struct pci_device_id azx_ids[] = { /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, - /* ICH */ + /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH6 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH7 */ { PCI_DEVICE(0x8086, 0x27d8), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH7 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ESB2 */ { PCI_DEVICE(0x8086, 0x269a), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ESB2 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH8 */ { PCI_DEVICE(0x8086, 0x284b), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH8 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293f), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a3e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a6e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, /* Generic Intel */ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.1 From 103884a351a221553095c509a1dbbbf7d4fd9b05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Dec 2014 09:56:20 +0100 Subject: ALSA: hda - Drop AZX_DCAPS_ALIGN_BUFSIZE We introduced AZX_DCAPS_ALIGN_BUFSIZE to explicity show that the controller needs the alignment, with a slight hope that the buffer size alignment will be disabled as default in future. But the reality tells that most chips need the buffer size alignment, and it'll be likely enabled in future, too. This patch drops AZX_DCAPS_ALIGN_BUFSIZE to give back one more precious DCAPS bit for future use. At the same time, rename AZX_DCAPS_BUFSIZE with AZX_DCAPS_NO_ALIGN_BUFSIZE for avoiding confusion. AZX_DCAPS_ALIGN_BUFSIZE are still kept (but commented out) in each DCAPS presets for a purpose as markers. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 +++++++--------- sound/pci/hda/hda_priv.h | 4 ++-- 2 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 53e43d1..5ac0d39 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -278,24 +278,24 @@ enum { /* quirks for old Intel chipsets */ #define AZX_DCAPS_INTEL_ICH \ - (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE) + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ - (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) #define AZX_DCAPS_INTEL_HASWELL \ - (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ - (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\ + (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) @@ -315,7 +315,7 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\ + (AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | /*AZX_DCAPS_ALIGN_BUFSIZE |*/ \ AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\ AZX_DCAPS_SNOOP_TYPE(NVIDIA)) @@ -1568,10 +1568,8 @@ static int azx_first_init(struct azx *chip) if (align_buffer_size >= 0) chip->align_buffer_size = !!align_buffer_size; else { - if (chip->driver_caps & AZX_DCAPS_BUFSIZE) + if (chip->driver_caps & AZX_DCAPS_NO_ALIGN_BUFSIZE) chip->align_buffer_size = 0; - else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE) - chip->align_buffer_size = 1; else chip->align_buffer_size = 1; } @@ -2086,7 +2084,7 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_BUFSIZE }, + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_NO_ALIGN_BUFSIZE }, /* ATI SB 450/600/700/800/900 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI | AZX_DCAPS_PRESET_ATI_SB }, diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index a09703a..aa484fd 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -162,8 +162,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ -#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ -#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ +#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ +/* 22 unused */ #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ #define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ -- cgit v1.1 From b163be4cf4a3bb673038a21e368954f7f88347b7 Mon Sep 17 00:00:00 2001 From: Andrew Jackson Date: Wed, 3 Dec 2014 16:38:46 +0000 Subject: ASoC: dwc: Allocate resources with devm_ioremap_resource Prepare for the introduction of device-tree support by re-ordering some of the allocations and using devm_iomap_resource to simplify IO mapping. Signed-off-by: Andrew Jackson Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 46 +++++++++++++++++------------------------- 1 file changed, 19 insertions(+), 27 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e961388..08f0229 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev) return -EINVAL; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(&pdev->dev, "i2s region already claimed\n"); - return -EBUSY; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - dev->i2s_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dev->i2s_base) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); return -ENOMEM; } + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dev->i2s_base)) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return PTR_ERR(dev->i2s_base); + } + cap = pdata->cap; dev->capability = cap; dev->i2s_clk_cfg = pdata->i2s_clk_cfg; @@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_clk_put; - dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); - ret = -ENOMEM; - goto err_clk_disable; - } - if (cap & DWC_I2S_PLAY) { dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; @@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->capture.rates = pdata->snd_rates; } - dw_i2s_dai->ops = &dw_i2s_dai_ops; - dw_i2s_dai->suspend = dw_i2s_suspend; - dw_i2s_dai->resume = dw_i2s_resume; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, -- cgit v1.1 From e98c89e05e8b33ad40efff2012c1404e39d12ad8 Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 14:34:30 +0100 Subject: ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable" The clk_disable() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 66fddec..eefb161 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -842,12 +842,9 @@ static int fsi_clk_disable(struct device *dev, return -EINVAL; if (1 == clock->count--) { - if (clock->xck) - clk_disable(clock->xck); - if (clock->ick) - clk_disable(clock->ick); - if (clock->div) - clk_disable(clock->div); + clk_disable(clock->xck); + clk_disable(clock->ick); + clk_disable(clock->div); } return 0; -- cgit v1.1 From 1679b532870f565bc184434f545cdbd3fdeff6cf Mon Sep 17 00:00:00 2001 From: Markus Elfring Date: Tue, 2 Dec 2014 17:15:11 +0100 Subject: ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put" The of_node_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index ea9ba284..9f2d045 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,8 @@ static void mop500_of_node_put(void) int i; for (i = 0; i < 2; i++) { - if (mop500_dai_links[i].cpu_of_node) - of_node_put(mop500_dai_links[i].cpu_of_node); - if (mop500_dai_links[i].codec_of_node) - of_node_put(mop500_dai_links[i].codec_of_node); + of_node_put(mop500_dai_links[i].cpu_of_node); + of_node_put(mop500_dai_links[i].codec_of_node); } } -- cgit v1.1 From 97463e193654574e1533f71359d91d9d2fdb3571 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:02:43 +0000 Subject: ASoC: rsnd: add .fallback callback Current R-Car sound has PIO fallback support if it couldn't use DMA. This fallback is done in .remove callback, but, it should have .fallback callback. Otherwise, normal .remove callback will have strange behavior. This patch adds .fallback callback. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 9 +++++++-- sound/soc/sh/rcar/rsnd.h | 2 ++ sound/soc/sh/rcar/ssi.c | 11 +++++++++-- 3 files changed, 18 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 110b99d..0785f84 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -368,7 +368,7 @@ rsnd_dma_channel_err: /* * DMA failed. try to PIO mode * see - * rsnd_ssi_dma_remove() + * rsnd_ssi_fallback() * rsnd_rdai_continuance_probe() */ return -EAGAIN; @@ -1023,7 +1023,7 @@ static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, * SSI will be switch to PIO mode if it was DMA mode * see * rsnd_dma_init() - * rsnd_ssi_dma_remove() + * rsnd_ssi_fallback() */ rsnd_dai_call(remove, io, rdai); @@ -1034,6 +1034,11 @@ static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, rsnd_path_break(priv, io, dvc); /* + * fallback + */ + rsnd_dai_call(fallback, io, rdai); + + /* * retry to "probe". * DAI has SSI which is PIO mode only now. */ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ed44ca8..83e1066 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -206,6 +206,8 @@ struct rsnd_mod_ops { int (*pcm_new)(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd); + int (*fallback)(struct rsnd_mod *mod, + struct rsnd_dai *rdai); }; struct rsnd_dai_stream; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 346d3dc..e03e70b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -465,11 +465,17 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + + return 0; +} + +static int rsnd_ssi_fallback(struct rsnd_mod *mod, + struct rsnd_dai *rdai) +{ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); - /* * fallback to PIO * @@ -541,6 +547,7 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .quit = rsnd_ssi_quit, .start = rsnd_ssi_dma_start, .stop = rsnd_ssi_dma_stop, + .fallback = rsnd_ssi_fallback, }; /* -- cgit v1.1 From 417f96420a5823485b90ad7ee9fddb67996bbd7f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:02:55 +0000 Subject: ASoC: rsnd: add callback status check method R-Car sound can use SSI/SRC/DVC modules, and these are controlled as rsnd_mod in rsnd driver. These rsnd_mod has each own function as callback. Basically these callback function has pair like probe/remove, start/stop, etc. And, these functions are called by order to each stage like below. 1. src->probe 2. ssi->probe 3. dvc->probe 4. src->start 5. ssi->start 6. dvc->start 7. src->stop 8. ssi->stop 9. dvc->stop 10. src->remove 11. ssi->remove 12. dvc->remove But, current rsnd driver doesn't care about its status which indicates which function is called. For example, if 5) returns error, 6) is not called. In such case, 9) should not be called. This patch care about each modules status. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 13 ++++++++++--- sound/soc/sh/rcar/rsnd.h | 28 ++++++++++++++++++++++++++++ 2 files changed, 38 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0785f84..fce61a0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -416,9 +416,16 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod) ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ - dev_dbg(dev, "%s[%d] %s\n", \ - rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \ - (mod)->ops->func(mod, rdai); \ + u32 mask = 1 << __rsnd_mod_shift_##func; \ + u32 call = __rsnd_mod_call_##func << __rsnd_mod_shift_##func; \ + int ret = 0; \ + if ((mod->status & mask) == call) { \ + dev_dbg(dev, "%s[%d] %s\n", \ + rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \ + ret = (mod)->ops->func(mod, rdai); \ + mod->status = (mod->status & ~mask) | (~call & mask); \ + } \ + ret; \ }) #define rsnd_mod_call(mod, func, rdai...) \ diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 83e1066..c74c239 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -218,7 +218,35 @@ struct rsnd_mod { struct rsnd_mod_ops *ops; struct rsnd_dma dma; struct rsnd_dai_stream *io; + u32 status; }; +/* + * status + * + * bit + * 0 0: probe 1: remove + * 1 0: init 1: quit + * 2 0: start 1: stop + * 3 0: pcm_new + * 4 0: fallback + */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 +#define __rsnd_mod_shift_init 1 +#define __rsnd_mod_shift_quit 1 +#define __rsnd_mod_shift_start 2 +#define __rsnd_mod_shift_stop 2 +#define __rsnd_mod_shift_pcm_new 3 +#define __rsnd_mod_shift_fallback 4 + +#define __rsnd_mod_call_probe 0 +#define __rsnd_mod_call_remove 1 +#define __rsnd_mod_call_init 0 +#define __rsnd_mod_call_quit 1 +#define __rsnd_mod_call_start 0 +#define __rsnd_mod_call_stop 1 +#define __rsnd_mod_call_pcm_new 0 +#define __rsnd_mod_call_fallback 0 #define rsnd_mod_to_priv(mod) ((mod)->priv) #define rsnd_mod_to_dma(mod) (&(mod)->dma) -- cgit v1.1 From 660cdce2fbdcbe48eb143cc394fdb24316232dba Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:03:39 +0000 Subject: ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily rsnd_src_ssiu_stop() is used to stop SSIU, but it shouldn't depend on whether it is using SSIU. This patch stops SSIU compulsorily. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 6 ++---- sound/soc/sh/rcar/ssi.c | 4 ++-- 3 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index c74c239..1344c3a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -431,8 +431,7 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai, int use_busif); int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai, - int use_busif); + struct rsnd_dai *rdai); int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 4679501..384af90 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -175,14 +175,12 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, } int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, - struct rsnd_dai *rdai, - int use_busif) + struct rsnd_dai *rdai) { /* * DMA settings for SSIU */ - if (use_busif) - rsnd_mod_write(ssi_mod, SSI_CTRL, 0); + rsnd_mod_write(ssi_mod, SSI_CTRL, 0); return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index e03e70b..a200452 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -424,7 +424,7 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, rsnd_ssi_hw_stop(ssi, rdai); - rsnd_src_ssiu_stop(mod, rdai, 0); + rsnd_src_ssiu_stop(mod, rdai); return 0; } @@ -528,7 +528,7 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, rsnd_dma_stop(dma); - rsnd_src_ssiu_stop(mod, rdai, 1); + rsnd_src_ssiu_stop(mod, rdai); return 0; } -- cgit v1.1 From 05795411aeda8a86865b595e09f22273d85fce83 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:05:01 +0000 Subject: ASoC: rsnd: tidyup PIO/DMA mode settings method Current ssi.c has .cr_etc which is used for SSICR's etc settings. but, it is used as PIO/DMA switching purpose now. This patch tidyup this method. This is prepare for under/over run issue handling Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 25 +++++++++++++------------ 2 files changed, 14 insertions(+), 12 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 1344c3a..12e8945 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -445,6 +445,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id); int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod); +int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod); /* * R-Car DVC diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a200452..8928913 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -68,7 +68,6 @@ struct rsnd_ssi { struct rsnd_dai *rdai; u32 cr_own; u32 cr_clk; - u32 cr_etc; int err; unsigned int usrcnt; unsigned int rate; @@ -185,6 +184,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, { struct rsnd_priv *priv = rsnd_mod_to_priv(&ssi->mod); struct device *dev = rsnd_priv_to_dev(priv); + u32 cr_mode; u32 cr; if (0 == ssi->usrcnt) { @@ -198,9 +198,14 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, } } + cr_mode = rsnd_ssi_is_dma_mode(&ssi->mod) ? + DMEN : /* DMA : use DMA */ + UIEN | OIEN | DIEN; /* PIO : enable interrupt */ + + cr = ssi->cr_own | ssi->cr_clk | - ssi->cr_etc | + cr_mode | EN; rsnd_mod_write(&ssi->mod, SSICR, cr); @@ -403,9 +408,6 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - /* enable PIO IRQ */ - ssi->cr_etc = UIEN | OIEN | DIEN; - rsnd_src_ssiu_start(mod, rdai, 0); rsnd_src_enable_ssi_irq(mod, rdai); @@ -420,8 +422,6 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - ssi->cr_etc = 0; - rsnd_ssi_hw_stop(ssi, rdai); rsnd_src_ssiu_stop(mod, rdai); @@ -498,9 +498,6 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - /* enable DMA transfer */ - ssi->cr_etc = DMEN; - rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); rsnd_dma_start(dma); @@ -520,8 +517,6 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); - ssi->cr_etc = 0; - rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); rsnd_ssi_hw_stop(ssi, rdai); @@ -550,6 +545,12 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .fallback = rsnd_ssi_fallback, }; +int rsnd_ssi_is_dma_mode(struct rsnd_mod *mod) +{ + return mod->ops == &rsnd_ssi_dma_ops; +} + + /* * Non SSI */ -- cgit v1.1 From c17dba8b8e198758a4f61fe89e16106b82af18a9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:05:09 +0000 Subject: ASoC: rsnd: tidyup SSI interrupt enable/disable method Current SSI doesn't care interrupt "disable" method. And, it is used when PIO mode only at this point. SSI interrupt will be used for sound R/L issue workaround when DMA mode too. This patch tidyup SSI interrupt enable/disable method. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 4 +++- sound/soc/sh/rcar/src.c | 25 ++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 6 ++++-- 3 files changed, 29 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 12e8945..48999b1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -432,8 +432,10 @@ int rsnd_src_ssiu_start(struct rsnd_mod *ssi_mod, int use_busif); int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); -int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, +int rsnd_src_ssi_irq_enable(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai); +int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai); #define rsnd_src_nr(priv) ((priv)->src_nr) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 384af90..c301195 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -185,18 +185,37 @@ int rsnd_src_ssiu_stop(struct rsnd_mod *ssi_mod, return 0; } -int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod, +int rsnd_src_ssi_irq_enable(struct rsnd_mod *ssi_mod, struct rsnd_dai *rdai) { struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod); - /* enable PIO interrupt if Gen2 */ - if (rsnd_is_gen2(priv)) + if (rsnd_is_gen1(priv)) + return 0; + + /* enable SSI interrupt if Gen2 */ + if (rsnd_ssi_is_dma_mode(ssi_mod)) + rsnd_mod_write(ssi_mod, INT_ENABLE, 0x0e000000); + else rsnd_mod_write(ssi_mod, INT_ENABLE, 0x0f000000); return 0; } +int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod, + struct rsnd_dai *rdai) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod); + + if (rsnd_is_gen1(priv)) + return 0; + + /* disable SSI interrupt if Gen2 */ + rsnd_mod_write(ssi_mod, INT_ENABLE, 0x00000000); + + return 0; +} + unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8928913..42d7c7e 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -410,10 +410,10 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, rsnd_src_ssiu_start(mod, rdai, 0); - rsnd_src_enable_ssi_irq(mod, rdai); - rsnd_ssi_hw_start(ssi, rdai, io); + rsnd_src_ssi_irq_enable(mod, rdai); + return 0; } @@ -422,6 +422,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + rsnd_src_ssi_irq_disable(mod, rdai); + rsnd_ssi_hw_stop(ssi, rdai); rsnd_src_ssiu_stop(mod, rdai); -- cgit v1.1 From d1b64056d479a115e02c88ab63bc2406a0c46468 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:05:34 +0000 Subject: ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() Current SSI is careing SSIWSR which controls WS continue mode when DMA mode, but, it should be cared when PIO mode too. This patch fixup it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 42d7c7e..d67bca9 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -210,6 +210,10 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, rsnd_mod_write(&ssi->mod, SSICR, cr); + /* enable WS continue */ + if (rsnd_dai_is_clk_master(rdai)) + rsnd_mod_write(&ssi->mod, SSIWSR, CONT); + ssi->usrcnt++; dev_dbg(dev, "%s[%d] hw started\n", @@ -506,10 +510,6 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, rsnd_ssi_hw_start(ssi, ssi->rdai, io); - /* enable WS continue */ - if (rsnd_dai_is_clk_master(rdai)) - rsnd_mod_write(&ssi->mod, SSIWSR, CONT); - return 0; } -- cgit v1.1 From 3ba84f45231c19af2ca281273e4fca8df65de341 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:05:44 +0000 Subject: ASoC: rsnd: clear status register when HW start Let's clear SSI status when HW start Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d67bca9..3c0d31d 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -214,6 +214,9 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, if (rsnd_dai_is_clk_master(rdai)) rsnd_mod_write(&ssi->mod, SSIWSR, CONT); + /* clear error status */ + rsnd_mod_write(&ssi->mod, SSISR, 0); + ssi->usrcnt++; dev_dbg(dev, "%s[%d] hw started\n", -- cgit v1.1 From 7b466fc6130a4183b505ebbd8220b55335f69d88 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:05:54 +0000 Subject: ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode Current SSI start/stop sequence is different between PIO/DMA mode, but, almost all are same. this patch synchronize it. It will be shared in the future. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3c0d31d..292e98b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -95,6 +95,9 @@ static int rsnd_ssi_use_busif(struct rsnd_mod *mod) struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); int use_busif = 0; + if (!rsnd_ssi_is_dma_mode(mod)) + return 0; + if (!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_NO_BUSIF)) use_busif = 1; if (rsnd_io_to_mod_src(io)) @@ -415,7 +418,7 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - rsnd_src_ssiu_start(mod, rdai, 0); + rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); rsnd_ssi_hw_start(ssi, rdai, io); @@ -431,6 +434,8 @@ static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, rsnd_src_ssi_irq_disable(mod, rdai); + rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); + rsnd_ssi_hw_stop(ssi, rdai); rsnd_src_ssiu_stop(mod, rdai); @@ -509,10 +514,10 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); - rsnd_dma_start(dma); - rsnd_ssi_hw_start(ssi, ssi->rdai, io); + rsnd_dma_start(dma); + return 0; } @@ -522,12 +527,12 @@ static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + rsnd_dma_stop(dma); + rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); rsnd_ssi_hw_stop(ssi, rdai); - rsnd_dma_stop(dma); - rsnd_src_ssiu_stop(mod, rdai); return 0; -- cgit v1.1 From 4d24d44e4d368314f23159ad833fc6bd15868c1a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:06:03 +0000 Subject: ASoC: rsnd: show master clock rate when ADG probe master clock rate is useful information for debug. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index fc41a0e..14d1a71 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -430,7 +430,7 @@ int rsnd_adg_probe(struct platform_device *pdev, adg->clk[CLKI] = devm_clk_get(dev, "clk_i"); for_each_rsnd_clk(clk, adg, i) - dev_dbg(dev, "clk %d : %p\n", i, clk); + dev_dbg(dev, "clk %d : %p : %ld\n", i, clk, clk_get_rate(clk)); rsnd_adg_ssi_clk_init(priv, adg); -- cgit v1.1 From 170a2497a25688f4c6bbf011208fc5fe144ef59c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:06:14 +0000 Subject: ASoC: rsnd: move snd_kcontrol_new fucntions to core.c Current DVC is using snd_kcontrol_new functions, but, SRC will need same method. Move common functions to core.c Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 144 ++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 177 ++++------------------------------------------- sound/soc/sh/rcar/rsnd.h | 45 ++++++++++++ 3 files changed, 204 insertions(+), 162 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fce61a0..77af008 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -970,6 +970,150 @@ static struct snd_pcm_ops rsnd_pcm_ops = { }; /* + * snd_kcontrol + */ +#define kcontrol_to_cfg(kctrl) ((struct rsnd_kctrl_cfg *)kctrl->private_value) +static int rsnd_kctrl_info(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_info *uinfo) +{ + struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + + if (cfg->texts) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = cfg->size; + uinfo->value.enumerated.items = cfg->max; + if (uinfo->value.enumerated.item >= cfg->max) + uinfo->value.enumerated.item = cfg->max - 1; + strlcpy(uinfo->value.enumerated.name, + cfg->texts[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); + } else { + uinfo->count = cfg->size; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = cfg->max; + uinfo->type = (cfg->max == 1) ? + SNDRV_CTL_ELEM_TYPE_BOOLEAN : + SNDRV_CTL_ELEM_TYPE_INTEGER; + } + + return 0; +} + +static int rsnd_kctrl_get(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *uc) +{ + struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + int i; + + for (i = 0; i < cfg->size; i++) + if (cfg->texts) + uc->value.enumerated.item[i] = cfg->val[i]; + else + uc->value.integer.value[i] = cfg->val[i]; + + return 0; +} + +static int rsnd_kctrl_put(struct snd_kcontrol *kctrl, + struct snd_ctl_elem_value *uc) +{ + struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); + struct rsnd_kctrl_cfg *cfg = kcontrol_to_cfg(kctrl); + int i, change = 0; + + for (i = 0; i < cfg->size; i++) { + if (cfg->texts) { + change |= (uc->value.enumerated.item[i] != cfg->val[i]); + cfg->val[i] = uc->value.enumerated.item[i]; + } else { + change |= (uc->value.integer.value[i] != cfg->val[i]); + cfg->val[i] = uc->value.integer.value[i]; + } + } + + if (change) + cfg->update(mod); + + return change; +} + +static int __rsnd_kctrl_new(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_kctrl_cfg *cfg, + void (*update)(struct rsnd_mod *mod)) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_kcontrol *kctrl; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = name, + .info = rsnd_kctrl_info, + .get = rsnd_kctrl_get, + .put = rsnd_kctrl_put, + .private_value = (unsigned long)cfg, + }; + int ret; + + kctrl = snd_ctl_new1(&knew, mod); + if (!kctrl) + return -ENOMEM; + + ret = snd_ctl_add(card, kctrl); + if (ret < 0) + return ret; + + cfg->update = update; + + return 0; +} + +int rsnd_kctrl_new_m(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + void (*update)(struct rsnd_mod *mod), + struct rsnd_kctrl_cfg_m *_cfg, + u32 max) +{ + _cfg->cfg.max = max; + _cfg->cfg.size = RSND_DVC_CHANNELS; + _cfg->cfg.val = _cfg->val; + return __rsnd_kctrl_new(mod, rdai, rtd, name, &_cfg->cfg, update); +} + +int rsnd_kctrl_new_s(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + void (*update)(struct rsnd_mod *mod), + struct rsnd_kctrl_cfg_s *_cfg, + u32 max) +{ + _cfg->cfg.max = max; + _cfg->cfg.size = 1; + _cfg->cfg.val = &_cfg->val; + return __rsnd_kctrl_new(mod, rdai, rtd, name, &_cfg->cfg, update); +} + +int rsnd_kctrl_new_e(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_kctrl_cfg_s *_cfg, + void (*update)(struct rsnd_mod *mod), + const char * const *texts, + u32 max) +{ + _cfg->cfg.max = max; + _cfg->cfg.size = 1; + _cfg->cfg.val = &_cfg->val; + _cfg->cfg.texts = texts; + return __rsnd_kctrl_new(mod, rdai, rtd, name, &_cfg->cfg, update); +} + +/* * snd_soc_platform */ diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index e2c8473..5380a48 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -11,36 +11,18 @@ #include "rsnd.h" #define RSND_DVC_NAME_SIZE 16 -#define RSND_DVC_CHANNELS 2 #define DVC_NAME "dvc" -struct rsnd_dvc_cfg { - unsigned int max; - unsigned int size; - u32 *val; - const char * const *texts; -}; - -struct rsnd_dvc_cfg_m { - struct rsnd_dvc_cfg cfg; - u32 val[RSND_DVC_CHANNELS]; -}; - -struct rsnd_dvc_cfg_s { - struct rsnd_dvc_cfg cfg; - u32 val; -}; - struct rsnd_dvc { struct rsnd_dvc_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; struct clk *clk; - struct rsnd_dvc_cfg_m volume; - struct rsnd_dvc_cfg_m mute; - struct rsnd_dvc_cfg_s ren; /* Ramp Enable */ - struct rsnd_dvc_cfg_s rup; /* Ramp Rate Up */ - struct rsnd_dvc_cfg_s rdown; /* Ramp Rate Down */ + struct rsnd_kctrl_cfg_m volume; + struct rsnd_kctrl_cfg_m mute; + struct rsnd_kctrl_cfg_s ren; /* Ramp Enable */ + struct rsnd_kctrl_cfg_s rup; /* Ramp Rate Up */ + struct rsnd_kctrl_cfg_s rdown; /* Ramp Rate Down */ }; #define rsnd_mod_to_dvc(_mod) \ @@ -222,140 +204,6 @@ static int rsnd_dvc_stop(struct rsnd_mod *mod, return 0; } -static int rsnd_dvc_volume_info(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_info *uinfo) -{ - struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; - - if (cfg->texts) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = cfg->size; - uinfo->value.enumerated.items = cfg->max; - if (uinfo->value.enumerated.item >= cfg->max) - uinfo->value.enumerated.item = cfg->max - 1; - strlcpy(uinfo->value.enumerated.name, - cfg->texts[uinfo->value.enumerated.item], - sizeof(uinfo->value.enumerated.name)); - } else { - uinfo->count = cfg->size; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = cfg->max; - uinfo->type = (cfg->max == 1) ? - SNDRV_CTL_ELEM_TYPE_BOOLEAN : - SNDRV_CTL_ELEM_TYPE_INTEGER; - } - - return 0; -} - -static int rsnd_dvc_volume_get(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) -{ - struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; - int i; - - for (i = 0; i < cfg->size; i++) - if (cfg->texts) - ucontrol->value.enumerated.item[i] = cfg->val[i]; - else - ucontrol->value.integer.value[i] = cfg->val[i]; - - return 0; -} - -static int rsnd_dvc_volume_put(struct snd_kcontrol *kctrl, - struct snd_ctl_elem_value *ucontrol) -{ - struct rsnd_mod *mod = snd_kcontrol_chip(kctrl); - struct rsnd_dvc_cfg *cfg = (struct rsnd_dvc_cfg *)kctrl->private_value; - int i, change = 0; - - for (i = 0; i < cfg->size; i++) { - if (cfg->texts) { - change |= (ucontrol->value.enumerated.item[i] != cfg->val[i]); - cfg->val[i] = ucontrol->value.enumerated.item[i]; - } else { - change |= (ucontrol->value.integer.value[i] != cfg->val[i]); - cfg->val[i] = ucontrol->value.integer.value[i]; - } - } - - if (change) - rsnd_dvc_volume_update(mod); - - return change; -} - -static int __rsnd_dvc_pcm_new(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd, - const unsigned char *name, - struct rsnd_dvc_cfg *private) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_kcontrol *kctrl; - struct snd_kcontrol_new knew = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = name, - .info = rsnd_dvc_volume_info, - .get = rsnd_dvc_volume_get, - .put = rsnd_dvc_volume_put, - .private_value = (unsigned long)private, - }; - int ret; - - kctrl = snd_ctl_new1(&knew, mod); - if (!kctrl) - return -ENOMEM; - - ret = snd_ctl_add(card, kctrl); - if (ret < 0) - return ret; - - return 0; -} - -static int _rsnd_dvc_pcm_new_m(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd, - const unsigned char *name, - struct rsnd_dvc_cfg_m *private, - u32 max) -{ - private->cfg.max = max; - private->cfg.size = RSND_DVC_CHANNELS; - private->cfg.val = private->val; - return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); -} - -static int _rsnd_dvc_pcm_new_s(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd, - const unsigned char *name, - struct rsnd_dvc_cfg_s *private, - u32 max) -{ - private->cfg.max = max; - private->cfg.size = 1; - private->cfg.val = &private->val; - return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); -} - -static int _rsnd_dvc_pcm_new_e(struct rsnd_mod *mod, - struct rsnd_dai *rdai, - struct snd_soc_pcm_runtime *rtd, - const unsigned char *name, - struct rsnd_dvc_cfg_s *private, - const char * const *texts, - u32 max) -{ - private->cfg.max = max; - private->cfg.size = 1; - private->cfg.val = &private->val; - private->cfg.texts = texts; - return __rsnd_dvc_pcm_new(mod, rdai, rtd, name, &private->cfg); -} - static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct snd_soc_pcm_runtime *rtd) @@ -365,41 +213,46 @@ static int rsnd_dvc_pcm_new(struct rsnd_mod *mod, int ret; /* Volume */ - ret = _rsnd_dvc_pcm_new_m(mod, rdai, rtd, + ret = rsnd_kctrl_new_m(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Playback Volume" : "DVC In Capture Volume", + rsnd_dvc_volume_update, &dvc->volume, 0x00800000 - 1); if (ret < 0) return ret; /* Mute */ - ret = _rsnd_dvc_pcm_new_m(mod, rdai, rtd, + ret = rsnd_kctrl_new_m(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Mute Switch" : "DVC In Mute Switch", + rsnd_dvc_volume_update, &dvc->mute, 1); if (ret < 0) return ret; /* Ramp */ - ret = _rsnd_dvc_pcm_new_s(mod, rdai, rtd, + ret = rsnd_kctrl_new_s(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Ramp Switch" : "DVC In Ramp Switch", + rsnd_dvc_volume_update, &dvc->ren, 1); if (ret < 0) return ret; - ret = _rsnd_dvc_pcm_new_e(mod, rdai, rtd, + ret = rsnd_kctrl_new_e(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Ramp Up Rate" : "DVC In Ramp Up Rate", &dvc->rup, + rsnd_dvc_volume_update, dvc_ramp_rate, ARRAY_SIZE(dvc_ramp_rate)); if (ret < 0) return ret; - ret = _rsnd_dvc_pcm_new_e(mod, rdai, rtd, + ret = rsnd_kctrl_new_e(mod, rdai, rtd, rsnd_dai_is_play(rdai, io) ? "DVC Out Ramp Down Rate" : "DVC In Ramp Down Rate", &dvc->rdown, + rsnd_dvc_volume_update, dvc_ramp_rate, ARRAY_SIZE(dvc_ramp_rate)); if (ret < 0) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 48999b1..133ba1f 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -418,6 +418,51 @@ struct rsnd_priv { }) /* + * rsnd_kctrl + */ +struct rsnd_kctrl_cfg { + unsigned int max; + unsigned int size; + u32 *val; + const char * const *texts; + void (*update)(struct rsnd_mod *mod); +}; + +#define RSND_DVC_CHANNELS 2 +struct rsnd_kctrl_cfg_m { + struct rsnd_kctrl_cfg cfg; + u32 val[RSND_DVC_CHANNELS]; +}; + +struct rsnd_kctrl_cfg_s { + struct rsnd_kctrl_cfg cfg; + u32 val; +}; + +int rsnd_kctrl_new_m(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + void (*update)(struct rsnd_mod *mod), + struct rsnd_kctrl_cfg_m *_cfg, + u32 max); +int rsnd_kctrl_new_s(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + void (*update)(struct rsnd_mod *mod), + struct rsnd_kctrl_cfg_s *_cfg, + u32 max); +int rsnd_kctrl_new_e(struct rsnd_mod *mod, + struct rsnd_dai *rdai, + struct snd_soc_pcm_runtime *rtd, + const unsigned char *name, + struct rsnd_kctrl_cfg_s *_cfg, + void (*update)(struct rsnd_mod *mod), + const char * const *texts, + u32 max); + +/* * R-Car SRC */ int rsnd_src_probe(struct platform_device *pdev, -- cgit v1.1 From 0cf7718520dcd673d385105d92a6b1ab923ee373 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:06:25 +0000 Subject: ASoC: rsnd: tidyup rsnd_io_to_runtime() macro Avoid NULL pointer access Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 133ba1f..5826c8a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -303,7 +303,8 @@ struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id); int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io); int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai); #define rsnd_dai_get_platform_info(rdai) ((rdai)->info) -#define rsnd_io_to_runtime(io) ((io)->substream->runtime) +#define rsnd_io_to_runtime(io) ((io)->substream ? \ + (io)->substream->runtime : NULL) void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt); int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional); -- cgit v1.1 From b167a5780cacb602dfbd3d6f853d7ce916df3fb0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:06:34 +0000 Subject: ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() using many rsnd_src_convert_rate() is not readable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index c301195..0a56ccd 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -525,16 +525,17 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 convert_rate = rsnd_src_convert_rate(src); uint ratio; int ret; /* 6 - 1/6 are very enough ratio for SRC_BSDSR */ - if (!rsnd_src_convert_rate(src)) + if (!convert_rate) ratio = 0; - else if (rsnd_src_convert_rate(src) > runtime->rate) - ratio = 100 * rsnd_src_convert_rate(src) / runtime->rate; + else if (convert_rate > runtime->rate) + ratio = 100 * convert_rate / runtime->rate; else - ratio = 100 * runtime->rate / rsnd_src_convert_rate(src); + ratio = 100 * runtime->rate / convert_rate; if (ratio > 600) { dev_err(dev, "FSO/FSI ratio error\n"); -- cgit v1.1 From 603cefa59b1aed460f5fdcb1b6af32efc7e6d82f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:06:43 +0000 Subject: ASoC: rsnd: initialize SRC on rsnd_src_init() Current src initialize SRC on rsnd_src_set_convert_rate() but, it should be done on rsnd_src_init(). Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 0a56ccd..6a63f8f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -256,12 +256,6 @@ static int rsnd_src_set_convert_rate(struct rsnd_mod *mod, rsnd_mod_write(mod, SRC_SWRSR, 0); rsnd_mod_write(mod, SRC_SWRSR, 1); - /* - * Initialize the operation of the SRC internal circuits - * see rsnd_src_start() - */ - rsnd_mod_write(mod, SRC_SRCIR, 1); - /* Set channel number and output bit length */ rsnd_mod_write(mod, SRC_ADINR, rsnd_get_adinr(mod)); @@ -286,6 +280,12 @@ static int rsnd_src_init(struct rsnd_mod *mod, clk_prepare_enable(src->clk); + /* + * Initialize the operation of the SRC internal circuits + * see rsnd_src_start() + */ + rsnd_mod_write(mod, SRC_SRCIR, 1); + return 0; } @@ -306,7 +306,7 @@ static int rsnd_src_start(struct rsnd_mod *mod, /* * Cancel the initialization and operate the SRC function - * see rsnd_src_set_convert_rate() + * see rsnd_src_init() */ rsnd_mod_write(mod, SRC_SRCIR, 0); -- cgit v1.1 From 933cc8cb08d867459689662cf67017ea6f0c6b53 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:07:07 +0000 Subject: ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() Current src.c sets SRC_ROUTE_MODE0 on rsnd_src_start(), but, set it in rsnd_src_set_convert_rate() is natural. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6a63f8f..0b6b230 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -302,17 +302,12 @@ static int rsnd_src_quit(struct rsnd_mod *mod, static int rsnd_src_start(struct rsnd_mod *mod, struct rsnd_dai *rdai) { - struct rsnd_src *src = rsnd_mod_to_src(mod); - /* * Cancel the initialization and operate the SRC function * see rsnd_src_init() */ rsnd_mod_write(mod, SRC_SRCIR, 0); - if (rsnd_src_convert_rate(src)) - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); - return 0; } @@ -320,11 +315,7 @@ static int rsnd_src_start(struct rsnd_mod *mod, static int rsnd_src_stop(struct rsnd_mod *mod, struct rsnd_dai *rdai) { - struct rsnd_src *src = rsnd_mod_to_src(mod); - - if (rsnd_src_convert_rate(src)) - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0); - + /* nothing to do */ return 0; } @@ -431,6 +422,7 @@ static int rsnd_src_set_convert_timing_gen1(struct rsnd_mod *mod, static int rsnd_src_set_convert_rate_gen1(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_src *src = rsnd_mod_to_src(mod); int ret; ret = rsnd_src_set_convert_rate(mod, rdai); @@ -444,6 +436,10 @@ static int rsnd_src_set_convert_rate_gen1(struct rsnd_mod *mod, rsnd_mod_write(mod, SRC_MNFSR, rsnd_mod_read(mod, SRC_IFSVR) / 100 * 98); + /* Gen1/Gen2 are not compatible */ + if (rsnd_src_convert_rate(src)) + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */ return 0; @@ -548,6 +544,11 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod, rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); + if (convert_rate) { + /* Gen1/Gen2 are not compatible */ + rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + } + switch (rsnd_mod_id(mod)) { case 5: case 6: -- cgit v1.1 From 49229850bee539e94f0c085a0f89cbbda7f6074b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:07:17 +0000 Subject: ASoC: rsnd: share SSI starting method between PIO/DMA mode Basically, SSI starting method is same between PIO/DMA mode. Let's share it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 292e98b..5af016e7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -412,8 +412,8 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, return ret; } -static int rsnd_ssi_pio_start(struct rsnd_mod *mod, - struct rsnd_dai *rdai) +static int rsnd_ssi_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); @@ -427,8 +427,8 @@ static int rsnd_ssi_pio_start(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_pio_stop(struct rsnd_mod *mod, - struct rsnd_dai *rdai) +static int rsnd_ssi_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); @@ -448,8 +448,8 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .probe = rsnd_ssi_pio_probe, .init = rsnd_ssi_init, .quit = rsnd_ssi_quit, - .start = rsnd_ssi_pio_start, - .stop = rsnd_ssi_pio_stop, + .start = rsnd_ssi_start, + .stop = rsnd_ssi_stop, }; static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, @@ -508,13 +508,9 @@ static int rsnd_ssi_fallback(struct rsnd_mod *mod, static int rsnd_ssi_dma_start(struct rsnd_mod *mod, struct rsnd_dai *rdai) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - - rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); + struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - rsnd_ssi_hw_start(ssi, ssi->rdai, io); + rsnd_ssi_start(mod, rdai); rsnd_dma_start(dma); @@ -524,16 +520,11 @@ static int rsnd_ssi_dma_start(struct rsnd_mod *mod, static int rsnd_ssi_dma_stop(struct rsnd_mod *mod, struct rsnd_dai *rdai) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(&ssi->mod); + struct rsnd_dma *dma = rsnd_mod_to_dma(mod); rsnd_dma_stop(dma); - rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); - - rsnd_ssi_hw_stop(ssi, rdai); - - rsnd_src_ssiu_stop(mod, rdai); + rsnd_ssi_stop(mod, rdai); return 0; } -- cgit v1.1 From f0ef0cb84b4351601e696705c0be9cd54e264d81 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:07:28 +0000 Subject: ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() rsnd_src_start/stop() requests struct rsnd_dai as parameter. but, it is not used, and become more complex in L/R error handling. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 0b6b230..eede3ac 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -299,8 +299,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod, return 0; } -static int rsnd_src_start(struct rsnd_mod *mod, - struct rsnd_dai *rdai) +static int rsnd_src_start(struct rsnd_mod *mod) { /* * Cancel the initialization and operate the SRC function @@ -311,9 +310,7 @@ static int rsnd_src_start(struct rsnd_mod *mod, return 0; } - -static int rsnd_src_stop(struct rsnd_mod *mod, - struct rsnd_dai *rdai) +static int rsnd_src_stop(struct rsnd_mod *mod) { /* nothing to do */ return 0; @@ -488,7 +485,7 @@ static int rsnd_src_start_gen1(struct rsnd_mod *mod, rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), (1 << id)); - return rsnd_src_start(mod, rdai); + return rsnd_src_start(mod); } static int rsnd_src_stop_gen1(struct rsnd_mod *mod, @@ -498,7 +495,7 @@ static int rsnd_src_stop_gen1(struct rsnd_mod *mod, rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), 0); - return rsnd_src_stop(mod, rdai); + return rsnd_src_stop(mod); } static struct rsnd_mod_ops rsnd_src_gen1_ops = { @@ -646,7 +643,7 @@ static int rsnd_src_start_gen2(struct rsnd_mod *mod, rsnd_mod_write(mod, SRC_CTRL, val); - return rsnd_src_start(mod, rdai); + return rsnd_src_start(mod); } static int rsnd_src_stop_gen2(struct rsnd_mod *mod, @@ -658,7 +655,7 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, rsnd_dma_stop(rsnd_mod_to_dma(&src->mod)); - return rsnd_src_stop(mod, rdai); + return rsnd_src_stop(mod); } static struct rsnd_mod_ops rsnd_src_gen2_ops = { -- cgit v1.1 From 5395103dcc709d87f08edaecb786fc37781f3b22 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Wed, 3 Dec 2014 09:59:31 -0800 Subject: ALSA: ctxfi: Neaten get_daio_rsc Move the pointer declarations into the blocks that use them. Neaten the kfree calls when the _init functions fail. Trivially reduces object size (defconfig x86-64) $ size sound/pci/ctxfi/ctdaio.o.* text data bss dec hex filename 5287 224 0 5511 1587 sound/pci/ctxfi/ctdaio.o.new 5319 224 0 5543 15a7 sound/pci/ctxfi/ctdaio.o.old Signed-off-by: Joe Perches Noticed-by: Markus Elfring Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctdaio.c | 30 +++++++++++++----------------- 1 file changed, 13 insertions(+), 17 deletions(-) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index c1c3f88..9b87dd2 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -528,8 +528,6 @@ static int get_daio_rsc(struct daio_mgr *mgr, struct daio **rdaio) { int err; - struct dai *dai = NULL; - struct dao *dao = NULL; unsigned long flags; *rdaio = NULL; @@ -544,27 +542,30 @@ static int get_daio_rsc(struct daio_mgr *mgr, return err; } + err = -ENOMEM; /* Allocate mem for daio resource */ if (desc->type <= DAIO_OUT_MAX) { - dao = kzalloc(sizeof(*dao), GFP_KERNEL); - if (!dao) { - err = -ENOMEM; + struct dao *dao = kzalloc(sizeof(*dao), GFP_KERNEL); + if (!dao) goto error; - } + err = dao_rsc_init(dao, desc, mgr); - if (err) + if (err) { + kfree(dao); goto error; + } *rdaio = &dao->daio; } else { - dai = kzalloc(sizeof(*dai), GFP_KERNEL); - if (!dai) { - err = -ENOMEM; + struct dai *dai = kzalloc(sizeof(*dai), GFP_KERNEL); + if (!dai) goto error; - } + err = dai_rsc_init(dai, desc, mgr); - if (err) + if (err) { + kfree(dai); goto error; + } *rdaio = &dai->daio; } @@ -575,11 +576,6 @@ static int get_daio_rsc(struct daio_mgr *mgr, return 0; error: - if (dao) - kfree(dao); - else if (dai) - kfree(dai); - spin_lock_irqsave(&mgr->mgr_lock, flags); daio_mgr_put_rsc(&mgr->mgr, desc->type); spin_unlock_irqrestore(&mgr->mgr_lock, flags); -- cgit v1.1 From f8781db8aeb18d34edb191bfed9c0a68affaaa74 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Thu, 27 Nov 2014 22:02:42 +0100 Subject: ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 443be00..b6e0e31 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- cgit v1.1 From 36fba62cce8184ea6a0cecfa43f07f712716a6f8 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Wed, 3 Dec 2014 10:13:43 +0800 Subject: ASoC: soc-pcm: do not hw_free BE if it's still used Do not free BE hw if it's still used by other FE during dpcm runtime shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and won't be updated to STATE_CLOSE when shutdown ends, because BE dai shutdown function won't close pcm when detecting BE is still under use. With STATE_HW_FREE, BE can't be triggered start again. This corner case can easily appear when one BE is used by two FE, without this patch "ASoC: dpcm: Fix race between FE/BE updates and trigger"(ea9d0d771fcd32cd56070819749477d511ec9117). One FE tries to shutdown but it's raced against xrun on another FE. It improves the be dai hw_free logic. Signed-off-by: Qiao Zhou Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..8435719 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1641,6 +1641,10 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + /* do not free hw if this BE is used by other FE */ + if (be->dpcm[stream].users > 1) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && -- cgit v1.1 From 2ffa531078037a002862d4befb14bc31aff5900d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 1 Dec 2014 19:57:14 -0200 Subject: ASoC: fsl_ssi: Fix module unbound Trying to remove the snd-soc-fsl-ssi module leads to the following warning: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- The cause of problem and solution are well explained by Lars-Peter: "The driver creates the mapping by calling irq_of_parse_and_map(), so it also has to dispose the mapping. But the easy way out is to simply use platform_get_irq() instead of irq_of_parse_map(). In this case the mapping is not managed by the device but by the of core, so the device has not to dispose the mapping." Tested on a imx6q-sabresd board. Reported-by: Jiada Wang Suggested-by: Lars-Peter Clausen Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bc19849..ad12d4c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1361,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1387,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1458,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1478,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } -- cgit v1.1 From 4c9a8845f95e852a21fe6cffbd7912107d71619c Mon Sep 17 00:00:00 2001 From: Jiada Wang Date: Tue, 2 Dec 2014 14:55:06 +0900 Subject: ASoC: fsl_ssi: fix error path in probe SSI component isn't unregistered if fsl_ssi_debugfs_create() fails in probe phase. To fix it, this commit replaces label error_asoc_register with error_irq. Signed-off-by: Jiada Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ad12d4c..7dee341 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1410,7 +1410,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_asoc_register; + goto error_irq; /* * If codec-handle property is missing from SSI node, we assume -- cgit v1.1 From 40e3262e425a04743f2a579a379f2f189f084580 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Thu, 4 Dec 2014 17:00:13 -0800 Subject: ASoC: rt5677: make volume TLV closer to reality The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Only use the resolution supported, ignoring the LSB of the volume register. This results in half the steps and 0.75dB per step, but reports accurate levels through TLV. Update the masks to reflect that these are registers have the LSB ignored. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 22 +++++++++++----------- sound/soc/codecs/rt5677.h | 24 ++++++++++++------------ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 133010d..81fe146 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -763,9 +763,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); @@ -817,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), @@ -842,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, - RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), /* Sidetone Control */ diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index dbd9ffd..c0a625f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -306,10 +306,10 @@ #define RT5677_R_MUTE_SFT 7 #define RT5677_VOL_R_MUTE (0x1 << 6) #define RT5677_VOL_R_SFT 6 -#define RT5677_L_VOL_MASK (0x3f << 8) -#define RT5677_L_VOL_SFT 8 -#define RT5677_R_VOL_MASK (0x3f) -#define RT5677_R_VOL_SFT 0 +#define RT5677_L_VOL_MASK (0x7f << 9) +#define RT5677_L_VOL_SFT 9 +#define RT5677_R_VOL_MASK (0x7f << 1) +#define RT5677_R_VOL_SFT 1 /* LOUT1 Control (0x01) */ #define RT5677_LOUT1_L_MUTE (0x1 << 15) @@ -447,16 +447,16 @@ #define RT5677_SEL_DAC2_R_SRC_SFT 0 /* Stereo1 ADC Digital Volume Control (0x1c) */ -#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_STO1_ADC_L_VOL_SFT 8 -#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) -#define RT5677_STO1_ADC_R_VOL_SFT 0 +#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_STO1_ADC_L_VOL_SFT 9 +#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_STO1_ADC_R_VOL_SFT 1 /* Mono ADC Digital Volume Control (0x1d) */ -#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_MONO_ADC_L_VOL_SFT 8 -#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) -#define RT5677_MONO_ADC_R_VOL_SFT 0 +#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_MONO_ADC_L_VOL_SFT 9 +#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_MONO_ADC_R_VOL_SFT 1 /* Stereo 1/2 ADC Boost Gain Control (0x1e) */ #define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) -- cgit v1.1 From ca460cc250aa9d5255b3e2469ca0abad5d136233 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Fri, 5 Dec 2014 15:42:54 +0100 Subject: ALSA: sound/atmel/ac97c.c: Fix device index for pcm chip->pdev->id is -1 by default. This is an invalid index resulting in device file names like /dev/snd/pcmC0D-1p. Signed-off-by: Alexander Stein Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d..f1d1195 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -773,7 +773,7 @@ static int atmel_ac97c_pcm_new(struct atmel_ac97c *chip) return err; } retval = snd_pcm_new(chip->card, chip->card->shortname, - chip->pdev->id, playback, capture, &pcm); + 0, playback, capture, &pcm); if (retval) return retval; -- cgit v1.1 From fedb2245cbb8d823e449ebdd48ba9bb35c071ce0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Nov 2014 07:11:38 +0100 Subject: ALSA: hda - Fix built-in mic at resume on Lenovo Ideapad S210 The built-in mic boost volume gets almost muted after suspend/resume on Lenovo Ideapad S210. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88121 Reported-and-tested-by: Roman Kagan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b118a5b..c5ad83e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4887,6 +4887,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), -- cgit v1.1 From 1132015b16580d3f53385354ffec0f58443a1ffd Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Fri, 5 Dec 2014 20:10:06 +0100 Subject: ALSA: sound/atmel/ac97c.c: Add missing clock prepare Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. This fixes the following warning: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at drivers/clk/clk.c:895 __clk_enable+0x24/0x9c() Modules linked in: CPU: 0 PID: 1 Comm: swapper Tainted: G W 3.18.0-rc7+ #245 [] (unwind_backtrace) from [] (show_stack+0x10/0x14) [] (show_stack) from [] (warn_slowpath_common+0x60/0x80) [] (warn_slowpath_common) from [] (warn_slowpath_null+0x18/0x20) [] (warn_slowpath_null) from [] (__clk_enable+0x24/0x9c) [] (__clk_enable) from [] (clk_enable+0x18/0x2c) [] (clk_enable) from [] (atmel_ac97c_probe+0x154/0x694) [] (atmel_ac97c_probe) from [] (platform_drv_probe+0x48/0x94) [] (platform_drv_probe) from [] (driver_probe_device+0x138/0x350) [] (driver_probe_device) from [] (__driver_attach+0x68/0x8c) [] (__driver_attach) from [] (bus_for_each_dev+0x70/0x84) [] (bus_for_each_dev) from [] (bus_add_driver+0xfc/0x1f8) [] (bus_add_driver) from [] (driver_register+0x9c/0xe0) [] (driver_register) from [] (do_one_initcall+0x110/0x1c8) [] (do_one_initcall) from [] (kernel_init_freeable+0xf8/0x1b8) [] (kernel_init_freeable) from [] (kernel_init+0x8/0xe4) [] (kernel_init) from [] (ret_from_fork+0x14/0x24) ---[ end trace cb88537fdc8fa201 ]--- atmel_ac97c fffa0000.sound: AC'97 0 does not respond - RESET atmel_ac97c fffa0000.sound: AC'97 0 access is not valid [0xffffffff], removing mixer. ------------[ cut here ]------------ Signed-off-by: Alexander Stein Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index f1d1195..cb44c74 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -944,7 +944,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1122,7 +1122,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1139,7 +1139,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1149,7 +1149,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1177,7 +1177,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- cgit v1.1 From 048184540171672a724ab8f8bada7fcc0762f5c6 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Fri, 5 Dec 2014 20:10:07 +0100 Subject: ALSA: atmel_abdac: Add missing clock prepare Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. Signed-off-by: Alexander Stein Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 31061e3..0231405 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -242,7 +242,7 @@ static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - clk_enable(dac->sample_clk); + clk_prepare_enable(dac->sample_clk); retval = dw_dma_cyclic_start(dac->dma.chan); if (retval) goto out; @@ -254,7 +254,7 @@ static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd) dw_dma_cyclic_stop(dac->dma.chan); dac_writel(dac, DATA, 0); dac_writel(dac, CTRL, 0); - clk_disable(dac->sample_clk); + clk_disable_unprepare(dac->sample_clk); break; default: retval = -EINVAL; @@ -429,7 +429,7 @@ static int atmel_abdac_probe(struct platform_device *pdev) retval = PTR_ERR(sample_clk); goto out_put_pclk; } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -528,7 +528,7 @@ out_free_card: snd_card_free(card); out_put_sample_clk: clk_put(sample_clk); - clk_disable(pclk); + clk_disable_unprepare(pclk); out_put_pclk: clk_put(pclk); return retval; @@ -541,8 +541,8 @@ static int atmel_abdac_suspend(struct device *pdev) struct atmel_abdac *dac = card->private_data; dw_dma_cyclic_stop(dac->dma.chan); - clk_disable(dac->sample_clk); - clk_disable(dac->pclk); + clk_disable_unprepare(dac->sample_clk); + clk_disable_unprepare(dac->pclk); return 0; } @@ -552,8 +552,8 @@ static int atmel_abdac_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_abdac *dac = card->private_data; - clk_enable(dac->pclk); - clk_enable(dac->sample_clk); + clk_prepare_enable(dac->pclk); + clk_prepare_enable(dac->sample_clk); if (test_bit(DMA_READY, &dac->flags)) dw_dma_cyclic_start(dac->dma.chan); @@ -572,7 +572,7 @@ static int atmel_abdac_remove(struct platform_device *pdev) struct atmel_abdac *dac = get_dac(card); clk_put(dac->sample_clk); - clk_disable(dac->pclk); + clk_disable_unprepare(dac->pclk); clk_put(dac->pclk); dma_release_channel(dac->dma.chan); -- cgit v1.1 From ba56447c3586465fd6eaf9869410dafb748a4d0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Dec 2014 19:58:32 +0000 Subject: ASoC: samsung: Fix error handling for clock lookup Return the error code we got from clk_get() and check to make sure that clk_prepare_enable() worked. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0d76bc1..c60ab07 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; + int ret; if (other && other->clk) { /* If this is probe on secondary */ samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, @@ -989,9 +990,14 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); iounmap(i2s->addr); - return -ENOENT; + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->clk); + if (ret != 0) { + dev_err(&i2s->pdev->dev, "failed to enable clock: %d\n", ret); + return ret; } - clk_prepare_enable(i2s->clk); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- cgit v1.1 From 66139a48cee1530c91f37c145384b4ee7043f0b7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 6 Dec 2014 18:02:55 +0100 Subject: ALSA: usb-audio: Don't resubmit pending URBs at MIDI error recovery In snd_usbmidi_error_timer(), the driver tries to resubmit MIDI input URBs to reactivate the MIDI stream, but this causes the error when some of URBs are still pending like: WARNING: CPU: 0 PID: 0 at ../drivers/usb/core/urb.c:339 usb_submit_urb+0x5f/0x70() URB ef705c40 submitted while active CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.16.6-2-desktop #1 Hardware name: FOXCONN TPS01/TPS01, BIOS 080015 03/23/2010 c0984bfa f4009ed4 c078deaf f4009ee4 c024c884 c09a135c f4009f00 00000000 c0984bfa 00000153 c061ac4f c061ac4f 00000009 00000001 ef705c40 e854d1c0 f4009eec c024c8d3 00000009 f4009ee4 c09a135c f4009f00 f4009f04 c061ac4f Call Trace: [] try_stack_unwind+0x156/0x170 [] dump_trace+0x5a/0x1b0 [] show_trace_log_lvl+0x46/0x50 [] show_stack_log_lvl+0x51/0xe0 [] show_stack+0x27/0x50 [] dump_stack+0x45/0x65 [] warn_slowpath_common+0x84/0xa0 [] warn_slowpath_fmt+0x33/0x40 [] usb_submit_urb+0x5f/0x70 [] snd_usbmidi_submit_urb+0x14/0x60 [snd_usbmidi_lib] [] snd_usbmidi_error_timer+0x6a/0xa0 [snd_usbmidi_lib] [] call_timer_fn+0x30/0x130 [] run_timer_softirq+0x1c2/0x260 [] __do_softirq+0xc3/0x270 [] do_softirq_own_stack+0x22/0x30 [] irq_exit+0x8d/0xa0 [] smp_apic_timer_interrupt+0x38/0x50 [] apic_timer_interrupt+0x34/0x3c [] cpuidle_enter_state+0x3e/0xd0 [] cpu_idle_loop+0x29d/0x3e0 [] cpu_startup_entry+0x53/0x60 [] start_kernel+0x415/0x41a For avoiding these errors, check the pending URBs and skip resubmitting such ones. Reported-and-tested-by: Stefan Seyfried Acked-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d3d4952..5bfb695 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -365,6 +365,8 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { + if (atomic_read(&in->urbs[j]->use_count)) + continue; in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } -- cgit v1.1 From 4e7d606cd52aa8997805b44a50065a7f74cd2953 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:07:47 +0000 Subject: ASoC: rsnd: add salvage support for under/over flow error on SSI L/R channel will be switched if under/over flow error happen on Renesas R-Car sound device by the HW bugs. Then, HW restart is required for salvage. This patch add salvage support for SSI. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/ssi.c | 129 ++++++++++++++++++++++++++++++------------------ 1 file changed, 82 insertions(+), 47 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 5af016e7..6f7080b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -202,14 +202,14 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, } cr_mode = rsnd_ssi_is_dma_mode(&ssi->mod) ? - DMEN : /* DMA : use DMA */ - UIEN | OIEN | DIEN; /* PIO : enable interrupt */ + DMEN : /* DMA : enable DMA */ + DIEN; /* PIO : enable Data interrupt */ cr = ssi->cr_own | ssi->cr_clk | cr_mode | - EN; + UIEN | OIEN | EN; rsnd_mod_write(&ssi->mod, SSICR, cr); @@ -355,22 +355,54 @@ static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status) /* * SSI PIO */ +static int rsnd_ssi_start(struct rsnd_mod *mod, + struct rsnd_dai *rdai) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + + rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); + + rsnd_ssi_hw_start(ssi, rdai, io); + + rsnd_src_ssi_irq_enable(mod, rdai); + + return 0; +} + +static int rsnd_ssi_stop(struct rsnd_mod *mod, + struct rsnd_dai *rdai) +{ + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + + rsnd_src_ssi_irq_disable(mod, rdai); + + rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); + + rsnd_ssi_hw_stop(ssi, rdai); + + rsnd_src_ssiu_stop(mod, rdai); + + return 0; +} + static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) { struct rsnd_ssi *ssi = data; + struct rsnd_dai *rdai = ssi->rdai; struct rsnd_mod *mod = &ssi->mod; struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); u32 status = rsnd_mod_read(mod, SSISR); - irqreturn_t ret = IRQ_NONE; - if (io && (status & DIRQ)) { - struct rsnd_dai *rdai = ssi->rdai; + if (!io) + return IRQ_NONE; + + /* PIO only */ + if (status & DIRQ) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 *buf = (u32 *)(runtime->dma_area + rsnd_dai_pointer_offset(io, 0)); - rsnd_ssi_record_error(ssi, status); - /* * 8/16/32 data can be assesse to TDR/RDR register * directly as 32bit data @@ -382,11 +414,26 @@ static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) *buf = rsnd_mod_read(mod, SSIRDR); rsnd_dai_pointer_update(io, sizeof(*buf)); + } - ret = IRQ_HANDLED; + /* PIO / DMA */ + if (status & (UIRQ | OIRQ)) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + + /* + * restart SSI + */ + rsnd_ssi_stop(mod, rdai); + rsnd_ssi_start(mod, rdai); + + dev_dbg(dev, "%s[%d] restart\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); } - return ret; + rsnd_ssi_record_error(ssi, status); + + return IRQ_HANDLED; } static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, @@ -412,37 +459,6 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, return ret; } -static int rsnd_ssi_start(struct rsnd_mod *mod, - struct rsnd_dai *rdai) -{ - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); - - rsnd_src_ssiu_start(mod, rdai, rsnd_ssi_use_busif(mod)); - - rsnd_ssi_hw_start(ssi, rdai, io); - - rsnd_src_ssi_irq_enable(mod, rdai); - - return 0; -} - -static int rsnd_ssi_stop(struct rsnd_mod *mod, - struct rsnd_dai *rdai) -{ - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - - rsnd_src_ssi_irq_disable(mod, rdai); - - rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR)); - - rsnd_ssi_hw_stop(ssi, rdai); - - rsnd_src_ssiu_stop(mod, rdai); - - return 0; -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_pio_probe, @@ -461,17 +477,28 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, int dma_id = ssi->info->dma_id; int ret; + ret = devm_request_irq(dev, ssi->info->pio_irq, + rsnd_ssi_pio_interrupt, + IRQF_SHARED, + dev_name(dev), ssi); + if (ret) + goto rsnd_ssi_dma_probe_fail; + ret = rsnd_dma_init( priv, rsnd_mod_to_dma(mod), rsnd_info_is_playback(priv, ssi), dma_id); + if (ret) + goto rsnd_ssi_dma_probe_fail; - if (ret < 0) - dev_err(dev, "%s[%d] (DMA) is failed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); - else - dev_dbg(dev, "%s[%d] (DMA) is probed\n", - rsnd_mod_name(mod), rsnd_mod_id(mod)); + dev_dbg(dev, "%s[%d] (DMA) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + return ret; + +rsnd_ssi_dma_probe_fail: + dev_err(dev, "%s[%d] (DMA) is failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -479,8 +506,16 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + struct device *dev = rsnd_priv_to_dev(priv); + int irq = ssi->info->pio_irq; + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + /* PIO will request IRQ again */ + devm_free_irq(dev, irq, ssi); + return 0; } -- cgit v1.1 From 6cfad789610342443923737a9def8b637151dc4d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 27 Nov 2014 08:08:10 +0000 Subject: ASoC: rsnd: rename SSI function name of PIO Current R-Car sound SSI PIO/DMA mode are using interrupt. it is no longer "xxx_pio_xxx", rename it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 8 ++++---- sound/soc/sh/rcar/ssi.c | 23 +++++++++++------------ 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index d76412b..83284ca 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -36,14 +36,14 @@ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ -#define RSND_SSI(_dma_id, _pio_irq, _flags) \ -{ .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags } +#define RSND_SSI(_dma_id, _irq, _flags) \ +{ .dma_id = _dma_id, .irq = _irq, .flags = _flags } #define RSND_SSI_UNUSED \ -{ .dma_id = -1, .pio_irq = -1, .flags = 0 } +{ .dma_id = -1, .irq = -1, .flags = 0 } struct rsnd_ssi_platform_info { int dma_id; - int pio_irq; + int irq; u32 flags; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 6f7080b..3844fbe 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -82,7 +82,7 @@ struct rsnd_ssi { #define rsnd_ssi_nr(priv) ((priv)->ssi_nr) #define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod) #define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma)) -#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0) +#define rsnd_ssi_pio_available(ssi) ((ssi)->info->irq > 0) #define rsnd_ssi_dma_available(ssi) \ rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod)) #define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent) @@ -352,9 +352,6 @@ static void rsnd_ssi_record_error(struct rsnd_ssi *ssi, u32 status) } } -/* - * SSI PIO - */ static int rsnd_ssi_start(struct rsnd_mod *mod, struct rsnd_dai *rdai) { @@ -386,7 +383,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod, return 0; } -static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) +static irqreturn_t rsnd_ssi_interrupt(int irq, void *data) { struct rsnd_ssi *ssi = data; struct rsnd_dai *rdai = ssi->rdai; @@ -436,17 +433,19 @@ static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data) return IRQ_HANDLED; } +/* + * SSI PIO + */ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, struct rsnd_dai *rdai) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - int irq = ssi->info->pio_irq; int ret; - ret = devm_request_irq(dev, irq, - rsnd_ssi_pio_interrupt, + ret = devm_request_irq(dev, ssi->info->irq, + rsnd_ssi_interrupt, IRQF_SHARED, dev_name(dev), ssi); if (ret) @@ -477,8 +476,8 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, int dma_id = ssi->info->dma_id; int ret; - ret = devm_request_irq(dev, ssi->info->pio_irq, - rsnd_ssi_pio_interrupt, + ret = devm_request_irq(dev, ssi->info->irq, + rsnd_ssi_interrupt, IRQF_SHARED, dev_name(dev), ssi); if (ret) @@ -509,7 +508,7 @@ static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); - int irq = ssi->info->pio_irq; + int irq = ssi->info->irq; rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); @@ -680,7 +679,7 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, /* * irq */ - ssi_info->pio_irq = irq_of_parse_and_map(np, 0); + ssi_info->irq = irq_of_parse_and_map(np, 0); /* * DMA -- cgit v1.1 From 3f024980fbf39ef6448e53345d51fc59f1da08ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Dec 2014 19:56:17 +0000 Subject: ASoC: samsung: Fix non-DT use of I2S controller The changes in commit a5a56871f804e (ASoC: samsung: add support for exynos7 I2S controller) introduce a new variant_regs structure in the driver data which is now mandatory for accessing registers. Unfortunately this is only hooked up for DT platforms so non-DT platforms like my primary development platform for audio are broken by this change and crash on boot. Since the only non-DT user of these device is s3c64xx fix this by making the standard samsung-i2s device be of type I2Sv3 and add a new I2Sv4 name to the platform data section, currently using the I2Sv5 information which should be about right. Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index c60ab07..c7aafcd 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1424,10 +1424,13 @@ static const struct samsung_i2s_dai_data samsung_dai_type_sec = { static struct platform_device_id samsung_i2s_driver_ids[] = { { .name = "samsung-i2s", - .driver_data = (kernel_ulong_t)&samsung_dai_type_pri, + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, + }, { + .name = "samsung-i2sv4", + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, }, {}, }; -- cgit v1.1 From 47370022d2ca552ab524bb14211fefa3a2518ba8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Dec 2014 20:06:31 +0000 Subject: ASoC: wm5102: Initialize dac_comp_lock mutex Commit d74bcaaeb6682 (ASoC: wm5102: Move ultrasonic response settings lock to the driver level) created a driver local mutex for protecting the ultrasonic response settings but neglected to initialize that mutex, causing loud complaints from lockep and potential runtime failures. Fix this by initializing the mutex. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm5102.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1f75534..d78fb8d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + mutex_init(&arizona->dac_comp_lock); + wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; -- cgit v1.1 From bf35df66f1c613b46e054ca35ceb5caddacc6fa8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 8 Dec 2014 15:44:47 +0100 Subject: ALSA: jack: Add dummy snd_jack_set_key() definition MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For fixing a build error with CONFIG_SND_JACK=n sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration] Signed-off-by: Takashi Iwai --- include/sound/jack.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/jack.h b/include/sound/jack.h index 67f2bbc..2182350 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -105,6 +105,13 @@ static inline void snd_jack_set_parent(struct snd_jack *jack, { } +static inline int snd_jack_set_key(struct snd_jack *jack, + enum snd_jack_types type, + int keytype) +{ + return 0; +} + static inline void snd_jack_report(struct snd_jack *jack, int status) { } -- cgit v1.1 From 288a8d0cb04f7715c7c302c8a40bdb227142f3a6 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:35 +0900 Subject: ALSA: dice: Change the way to start stream Streaming functionality can start streams when rate is given but currently some codes are in PCM functionality. This commit changes the way to start stream and add some arrangement to make it easy to understand the way. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-pcm.c | 75 ++------------- sound/firewire/dice/dice-stream.c | 191 +++++++++++++++++++++++++++----------- sound/firewire/dice/dice.c | 4 - sound/firewire/dice/dice.h | 4 +- 4 files changed, 143 insertions(+), 131 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 2e531bd..b185391 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -169,65 +169,11 @@ static int playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; - unsigned int mode, rate, channels, i; - int err; - - mutex_lock(&dice->mutex); - snd_dice_stream_stop(dice); - mutex_unlock(&dice->mutex); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - - rate = params_rate(hw_params); - err = snd_dice_transaction_set_rate(dice, rate); - if (err < 0) - return err; - - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - return err; - - /* - * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in - * one data block of AMDTP packet. Thus sampling transfer frequency is - * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are - * transferred on AMDTP packets at 96 kHz. Two successive samples of a - * channel are stored consecutively in the packet. This quirk is called - * as 'Dual Wire'. - * For this quirk, blocking mode is required and PCM buffer size should - * be aligned to SYT_INTERVAL. - */ - channels = params_channels(hw_params); - if (mode > 1) { - if (channels > AMDTP_MAX_CHANNELS_FOR_PCM / 2) { - err = -ENOSYS; - return err; - } - - rate /= 2; - channels *= 2; - dice->rx_stream.double_pcm_frames = true; - } else { - dice->rx_stream.double_pcm_frames = false; - } - - amdtp_stream_set_parameters(&dice->rx_stream, rate, channels, - dice->rx_midi_ports[mode]); - if (mode > 1) { - channels /= 2; - - for (i = 0; i < channels; i++) { - dice->rx_stream.pcm_positions[i] = i * 2; - dice->rx_stream.pcm_positions[i + channels] = i * 2 + 1; - } - } - amdtp_stream_set_pcm_format(&dice->rx_stream, params_format(hw_params)); - return 0; + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int playback_hw_free(struct snd_pcm_substream *substream) @@ -247,21 +193,12 @@ static int playback_prepare(struct snd_pcm_substream *substream) int err; mutex_lock(&dice->mutex); - - if (amdtp_streaming_error(&dice->rx_stream)) - snd_dice_stream_stop_packets(dice); - - err = snd_dice_stream_start(dice); - if (err < 0) { - mutex_unlock(&dice->mutex); - return err; - } - + err = snd_dice_stream_start(dice, substream->runtime->rate); mutex_unlock(&dice->mutex); + if (err >= 0) + amdtp_stream_pcm_prepare(&dice->rx_stream); - amdtp_stream_pcm_prepare(&dice->rx_stream); - - return 0; + return err; } static int playback_trigger(struct snd_pcm_substream *substream, int cmd) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 4c4c4ff..b9d7a48 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -9,6 +9,8 @@ #include "dice.h" +#define CALLBACK_TIMEOUT 200 + const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { /* mode 0 */ [0] = 32000, @@ -39,83 +41,162 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, return -EINVAL; } -int snd_dice_stream_start_packets(struct snd_dice *dice) +static void release_resources(struct snd_dice *dice) { - int err; + unsigned int channel; - if (amdtp_stream_running(&dice->rx_stream)) - return 0; - - err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, - fw_parent_device(dice->unit)->max_speed); - if (err < 0) - return err; - - err = snd_dice_transaction_set_enable(dice); - if (err < 0) { - amdtp_stream_stop(&dice->rx_stream); - return err; - } + /* Reset channel number */ + channel = cpu_to_be32((u32)-1); + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); - return 0; + fw_iso_resources_free(&dice->rx_resources); } -int snd_dice_stream_start(struct snd_dice *dice) +static int keep_resources(struct snd_dice *dice, unsigned int max_payload_bytes) { - __be32 channel; + unsigned int channel; int err; - if (!dice->rx_resources.allocated) { - err = fw_iso_resources_allocate(&dice->rx_resources, - amdtp_stream_get_max_payload(&dice->rx_stream), + err = fw_iso_resources_allocate(&dice->rx_resources, max_payload_bytes, fw_parent_device(dice->unit)->max_speed); - if (err < 0) - goto error; - - channel = cpu_to_be32(dice->rx_resources.channel); - err = snd_dice_transaction_write_tx(dice, RX_ISOCHRONOUS, - &channel, 4); - if (err < 0) - goto err_resources; - } - - err = snd_dice_stream_start_packets(dice); if (err < 0) - goto err_rx_channel; - - return 0; + goto end; -err_rx_channel: - channel = cpu_to_be32((u32)-1); - snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); -err_resources: - fw_iso_resources_free(&dice->rx_resources); -error: + /* Set channel number */ + channel = cpu_to_be32(dice->rx_resources.channel); + err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, + &channel, 4); + if (err < 0) + release_resources(dice); +end: return err; } -void snd_dice_stream_stop_packets(struct snd_dice *dice) +static void stop_stream(struct snd_dice *dice) { if (!amdtp_stream_running(&dice->rx_stream)) return; - snd_dice_transaction_clear_enable(dice); + amdtp_stream_pcm_abort(&dice->rx_stream); amdtp_stream_stop(&dice->rx_stream); + release_resources(dice); } -void snd_dice_stream_stop(struct snd_dice *dice) +static int start_stream(struct snd_dice *dice, unsigned int rate) { - __be32 channel; + unsigned int i, mode, pcm_chs, midi_ports; + int err; - snd_dice_stream_stop_packets(dice); + err = snd_dice_stream_get_rate_mode(dice, rate, &mode); + if (err < 0) + goto end; - if (!dice->rx_resources.allocated) - return; + /* + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. + */ + pcm_chs = dice->rx_channels[mode]; + midi_ports = dice->rx_midi_ports[mode]; + if (mode > 1) { + rate /= 2; + pcm_chs *= 2; + dice->rx_stream.double_pcm_frames = true; + } else { + dice->rx_stream.double_pcm_frames = false; + } - channel = cpu_to_be32((u32)-1); - snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); + amdtp_stream_set_parameters(&dice->rx_stream, rate, + pcm_chs, midi_ports); + if (mode > 1) { + pcm_chs /= 2; - fw_iso_resources_free(&dice->rx_resources); + for (i = 0; i < pcm_chs; i++) { + dice->rx_stream.pcm_positions[i] = i * 2; + dice->rx_stream.pcm_positions[i + pcm_chs] = i * 2 + 1; + } + } + + err = keep_resources(dice, + amdtp_stream_get_max_payload(&dice->rx_stream)); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to keep isochronous resources\n"); + goto end; + } + + err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, + fw_parent_device(dice->unit)->max_speed); + if (err < 0) + release_resources(dice); +end: + return err; +} + +int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate) +{ + unsigned int curr_rate; + int err; + + /* Some packet queueing errors. */ + if (amdtp_streaming_error(&dice->rx_stream)) + stop_stream(dice); + + /* Stop stream if rate is different. */ + err = snd_dice_transaction_get_rate(dice, &curr_rate); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to get sampling rate\n"); + goto end; + } + if (rate != curr_rate) + stop_stream(dice); + + if (!amdtp_stream_running(&dice->rx_stream)) { + snd_dice_transaction_clear_enable(dice); + + err = snd_dice_transaction_set_rate(dice, rate); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to set sampling rate\n"); + goto end; + } + + /* Start stream. */ + err = start_stream(dice, rate); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to start AMDTP stream\n"); + goto end; + } + err = snd_dice_transaction_set_enable(dice); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to enable interface\n"); + stop_stream(dice); + goto end; + } + + if (!amdtp_stream_wait_callback(&dice->rx_stream, + CALLBACK_TIMEOUT)) { + snd_dice_transaction_clear_enable(dice); + stop_stream(dice); + err = -ETIMEDOUT; + } + } +end: + return err; +} + +void snd_dice_stream_stop(struct snd_dice *dice) +{ + snd_dice_transaction_clear_enable(dice); + stop_stream(dice); } int snd_dice_stream_init(struct snd_dice *dice) @@ -145,8 +226,8 @@ error: void snd_dice_stream_destroy(struct snd_dice *dice) { - amdtp_stream_pcm_abort(&dice->rx_stream); - snd_dice_stream_stop(dice); + snd_dice_transaction_clear_enable(dice); + stop_stream(dice); amdtp_stream_destroy(&dice->rx_stream); fw_iso_resources_destroy(&dice->rx_resources); } @@ -163,8 +244,8 @@ void snd_dice_stream_update(struct snd_dice *dice) */ dice->global_enabled = false; - amdtp_stream_pcm_abort(&dice->rx_stream); - snd_dice_stream_stop_packets(dice); + stop_stream(dice); + fw_iso_resources_update(&dice->rx_resources); } diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 8e2c172..03a7988 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -304,12 +304,8 @@ static void dice_remove(struct fw_unit *unit) snd_card_disconnect(dice->card); - mutex_lock(&dice->mutex); - snd_dice_stream_destroy(dice); - mutex_unlock(&dice->mutex); - snd_card_free_when_closed(dice->card); } diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 969189a..8be530f 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -160,9 +160,7 @@ extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, unsigned int *mode); -int snd_dice_stream_start_packets(struct snd_dice *dice); -int snd_dice_stream_start(struct snd_dice *dice); -void snd_dice_stream_stop_packets(struct snd_dice *dice); +int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate); void snd_dice_stream_stop(struct snd_dice *dice); int snd_dice_stream_init(struct snd_dice *dice); void snd_dice_stream_destroy(struct snd_dice *dice); -- cgit v1.1 From 9a02843caefbc370ef6d5895881101f9436f98da Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:36 +0900 Subject: ALSA: dice: Add support for duplex streams with synchronization This commit adds support for AMDTP in-stream. As a result, Dice driver supports full duplex streams with synchronization. AMDTP can transfer timestamps in its packets. By handling the timestamp, devices can synchronize to the other devices or drivers on the same bus. When Dice chipset is 'enabled', it starts streams with correct settings. This 'enable' register is global, thus, when a stream is started to run, an opposite stream can't start unless turning off 'enable'. Therefore a pair of streams must be running. This causes a loss of CPU usage when single stream is needed for neither playbacking or capturing. This commit assumes that playback-only models also have a functionality to transmit stream for delivering timestamps. Currently, sampling clock source is restricted to SYT-Match mode. This is improved in followed commit. I note that at SYT-Match mode, Dice can select from 4 streams for synchronization but this driver uses the 1st stream only for simplicity. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-pcm.c | 4 +- sound/firewire/dice/dice-stream.c | 230 ++++++++++++++++++++++++++++---------- sound/firewire/dice/dice.c | 29 +++-- sound/firewire/dice/dice.h | 26 +++-- 4 files changed, 202 insertions(+), 87 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index b185391..b9ce026 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -181,7 +181,7 @@ static int playback_hw_free(struct snd_pcm_substream *substream) struct snd_dice *dice = substream->private_data; mutex_lock(&dice->mutex); - snd_dice_stream_stop(dice); + snd_dice_stream_stop_duplex(dice); mutex_unlock(&dice->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -193,7 +193,7 @@ static int playback_prepare(struct snd_pcm_substream *substream) int err; mutex_lock(&dice->mutex); - err = snd_dice_stream_start(dice, substream->runtime->rate); + err = snd_dice_stream_start_duplex(dice, substream->runtime->rate); mutex_unlock(&dice->mutex); if (err >= 0) amdtp_stream_pcm_prepare(&dice->rx_stream); diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index b9d7a48..e60b84d 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -41,55 +41,79 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, return -EINVAL; } -static void release_resources(struct snd_dice *dice) +static void release_resources(struct snd_dice *dice, + struct fw_iso_resources *resources) { unsigned int channel; /* Reset channel number */ channel = cpu_to_be32((u32)-1); - snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, &channel, 4); - - fw_iso_resources_free(&dice->rx_resources); + if (resources == &dice->tx_resources) + snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, + &channel, 4); + else + snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, + &channel, 4); + + fw_iso_resources_free(resources); } -static int keep_resources(struct snd_dice *dice, unsigned int max_payload_bytes) +static int keep_resources(struct snd_dice *dice, + struct fw_iso_resources *resources, + unsigned int max_payload_bytes) { unsigned int channel; int err; - err = fw_iso_resources_allocate(&dice->rx_resources, max_payload_bytes, + err = fw_iso_resources_allocate(resources, max_payload_bytes, fw_parent_device(dice->unit)->max_speed); if (err < 0) goto end; /* Set channel number */ - channel = cpu_to_be32(dice->rx_resources.channel); - err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, - &channel, 4); + channel = cpu_to_be32(resources->channel); + if (resources == &dice->tx_resources) + err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS, + &channel, 4); + else + err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS, + &channel, 4); if (err < 0) - release_resources(dice); + release_resources(dice, resources); end: return err; } -static void stop_stream(struct snd_dice *dice) +static void stop_stream(struct snd_dice *dice, struct amdtp_stream *stream) { - if (!amdtp_stream_running(&dice->rx_stream)) - return; + amdtp_stream_pcm_abort(stream); + amdtp_stream_stop(stream); - amdtp_stream_pcm_abort(&dice->rx_stream); - amdtp_stream_stop(&dice->rx_stream); - release_resources(dice); + if (stream == &dice->tx_stream) + release_resources(dice, &dice->tx_resources); + else + release_resources(dice, &dice->rx_resources); } -static int start_stream(struct snd_dice *dice, unsigned int rate) +static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream, + unsigned int rate) { + struct fw_iso_resources *resources; unsigned int i, mode, pcm_chs, midi_ports; int err; err = snd_dice_stream_get_rate_mode(dice, rate, &mode); if (err < 0) goto end; + if (stream == &dice->tx_stream) { + resources = &dice->tx_resources; + pcm_chs = dice->tx_channels[mode]; + midi_ports = dice->tx_midi_ports[mode]; + } else { + resources = &dice->rx_resources; + pcm_chs = dice->rx_channels[mode]; + midi_ports = dice->rx_midi_ports[mode]; + } /* * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in @@ -101,51 +125,71 @@ static int start_stream(struct snd_dice *dice, unsigned int rate) * For this quirk, blocking mode is required and PCM buffer size should * be aligned to SYT_INTERVAL. */ - pcm_chs = dice->rx_channels[mode]; - midi_ports = dice->rx_midi_ports[mode]; if (mode > 1) { rate /= 2; pcm_chs *= 2; - dice->rx_stream.double_pcm_frames = true; + stream->double_pcm_frames = true; } else { - dice->rx_stream.double_pcm_frames = false; + stream->double_pcm_frames = false; } - amdtp_stream_set_parameters(&dice->rx_stream, rate, - pcm_chs, midi_ports); + amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports); if (mode > 1) { pcm_chs /= 2; for (i = 0; i < pcm_chs; i++) { - dice->rx_stream.pcm_positions[i] = i * 2; - dice->rx_stream.pcm_positions[i + pcm_chs] = i * 2 + 1; + stream->pcm_positions[i] = i * 2; + stream->pcm_positions[i + pcm_chs] = i * 2 + 1; } } - err = keep_resources(dice, - amdtp_stream_get_max_payload(&dice->rx_stream)); + err = keep_resources(dice, resources, + amdtp_stream_get_max_payload(stream)); if (err < 0) { dev_err(&dice->unit->device, "fail to keep isochronous resources\n"); goto end; } - err = amdtp_stream_start(&dice->rx_stream, dice->rx_resources.channel, + err = amdtp_stream_start(stream, resources->channel, fw_parent_device(dice->unit)->max_speed); if (err < 0) - release_resources(dice); + release_resources(dice, resources); end: return err; } -int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate) +static int get_sync_mode(struct snd_dice *dice, enum cip_flags *sync_mode) +{ + /* Currently, clock source is fixed at SYT-Match mode. */ + *sync_mode = 0; + return 0; +} + +int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { + struct amdtp_stream *master, *slave; unsigned int curr_rate; - int err; + enum cip_flags sync_mode; + int err = 0; + + if (dice->substreams_counter == 0) + goto end; + + err = get_sync_mode(dice, &sync_mode); + if (err < 0) + goto end; + if (sync_mode == CIP_SYNC_TO_DEVICE) { + master = &dice->tx_stream; + slave = &dice->rx_stream; + } else { + master = &dice->rx_stream; + slave = &dice->tx_stream; + } /* Some packet queueing errors. */ - if (amdtp_streaming_error(&dice->rx_stream)) - stop_stream(dice); + if (amdtp_streaming_error(master) || amdtp_streaming_error(slave)) + stop_stream(dice, master); /* Stop stream if rate is different. */ err = snd_dice_transaction_get_rate(dice, &curr_rate); @@ -155,11 +199,14 @@ int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate) goto end; } if (rate != curr_rate) - stop_stream(dice); + stop_stream(dice, master); - if (!amdtp_stream_running(&dice->rx_stream)) { + if (!amdtp_stream_running(master)) { + stop_stream(dice, slave); snd_dice_transaction_clear_enable(dice); + amdtp_stream_set_sync(sync_mode, master, slave); + err = snd_dice_transaction_set_rate(dice, rate); if (err < 0) { dev_err(&dice->unit->device, @@ -167,25 +214,35 @@ int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate) goto end; } - /* Start stream. */ - err = start_stream(dice, rate); + /* Start both streams. */ + err = start_stream(dice, master, rate); + if (err < 0) { + dev_err(&dice->unit->device, + "fail to start AMDTP master stream\n"); + goto end; + } + err = start_stream(dice, slave, rate); if (err < 0) { dev_err(&dice->unit->device, - "fail to start AMDTP stream\n"); + "fail to start AMDTP slave stream\n"); + stop_stream(dice, master); goto end; } err = snd_dice_transaction_set_enable(dice); if (err < 0) { dev_err(&dice->unit->device, "fail to enable interface\n"); - stop_stream(dice); + stop_stream(dice, master); + stop_stream(dice, slave); goto end; } - if (!amdtp_stream_wait_callback(&dice->rx_stream, - CALLBACK_TIMEOUT)) { + /* Wait first callbacks */ + if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT) || + !amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) { snd_dice_transaction_clear_enable(dice); - stop_stream(dice); + stop_stream(dice, master); + stop_stream(dice, slave); err = -ETIMEDOUT; } } @@ -193,46 +250,93 @@ end: return err; } -void snd_dice_stream_stop(struct snd_dice *dice) +void snd_dice_stream_stop_duplex(struct snd_dice *dice) { + if (dice->substreams_counter > 0) + return; + snd_dice_transaction_clear_enable(dice); - stop_stream(dice); + + stop_stream(dice, &dice->tx_stream); + stop_stream(dice, &dice->rx_stream); } -int snd_dice_stream_init(struct snd_dice *dice) +static int init_stream(struct snd_dice *dice, struct amdtp_stream *stream) { int err; + struct fw_iso_resources *resources; + enum amdtp_stream_direction dir; + + if (stream == &dice->tx_stream) { + resources = &dice->tx_resources; + dir = AMDTP_IN_STREAM; + } else { + resources = &dice->rx_resources; + dir = AMDTP_OUT_STREAM; + } - err = fw_iso_resources_init(&dice->rx_resources, dice->unit); + err = fw_iso_resources_init(resources, dice->unit); if (err < 0) goto end; - dice->rx_resources.channels_mask = 0x00000000ffffffffuLL; + resources->channels_mask = 0x00000000ffffffffuLL; - err = amdtp_stream_init(&dice->rx_stream, dice->unit, AMDTP_OUT_STREAM, - CIP_BLOCKING); + err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING); + if (err < 0) { + amdtp_stream_destroy(stream); + fw_iso_resources_destroy(resources); + } +end: + return err; +} + +static void destroy_stream(struct snd_dice *dice, struct amdtp_stream *stream) +{ + amdtp_stream_destroy(stream); + + if (stream == &dice->tx_stream) + fw_iso_resources_destroy(&dice->tx_resources); + else + fw_iso_resources_destroy(&dice->rx_resources); +} + +int snd_dice_stream_init_duplex(struct snd_dice *dice) +{ + int err; + + dice->substreams_counter = 0; + + err = init_stream(dice, &dice->tx_stream); if (err < 0) - goto error; + goto end; - err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); + err = init_stream(dice, &dice->rx_stream); if (err < 0) - goto error; + goto end; + + /* Currently, clock source is fixed at SYT-Match mode. */ + err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); + if (err < 0) { + destroy_stream(dice, &dice->rx_stream); + destroy_stream(dice, &dice->tx_stream); + } end: return err; -error: - amdtp_stream_destroy(&dice->rx_stream); - fw_iso_resources_destroy(&dice->rx_resources); - return err; } -void snd_dice_stream_destroy(struct snd_dice *dice) +void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { snd_dice_transaction_clear_enable(dice); - stop_stream(dice); - amdtp_stream_destroy(&dice->rx_stream); - fw_iso_resources_destroy(&dice->rx_resources); + + stop_stream(dice, &dice->tx_stream); + destroy_stream(dice, &dice->tx_stream); + + stop_stream(dice, &dice->rx_stream); + destroy_stream(dice, &dice->rx_stream); + + dice->substreams_counter = 0; } -void snd_dice_stream_update(struct snd_dice *dice) +void snd_dice_stream_update_duplex(struct snd_dice *dice) { /* * On a bus reset, the DICE firmware disables streaming and then goes @@ -244,9 +348,11 @@ void snd_dice_stream_update(struct snd_dice *dice) */ dice->global_enabled = false; - stop_stream(dice); + stop_stream(dice, &dice->rx_stream); + stop_stream(dice, &dice->tx_stream); fw_iso_resources_update(&dice->rx_resources); + fw_iso_resources_update(&dice->tx_resources); } static void dice_lock_changed(struct snd_dice *dice) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 03a7988..85bcfaf3 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -30,7 +30,6 @@ static int dice_interface_check(struct fw_unit *unit) int key, val, vendor = -1, model = -1, err; unsigned int category, i; __be32 *pointers, value; - __be32 tx_data[4]; __be32 version; pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32), @@ -85,16 +84,6 @@ static int dice_interface_check(struct fw_unit *unit) } } - /* We support playback only. Let capture devices be handled by FFADO. */ - err = snd_fw_transaction(unit, TCODE_READ_BLOCK_REQUEST, - DICE_PRIVATE_SPACE + - be32_to_cpu(pointers[2]) * 4, - tx_data, sizeof(tx_data), 0); - if (err < 0 || (tx_data[0] && tx_data[3])) { - err = -ENODEV; - goto end; - } - /* * Check that the implemented DICE driver specification major version * number matches. @@ -142,6 +131,8 @@ static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) int err; if (highest_supported_mode_rate(dice, mode, &rate) < 0) { + dice->tx_channels[mode] = 0; + dice->tx_midi_ports[mode] = 0; dice->rx_channels[mode] = 0; dice->rx_midi_ports[mode] = 0; return 0; @@ -151,6 +142,14 @@ static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) if (err < 0) return err; + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, + values, sizeof(values)); + if (err < 0) + return err; + + dice->tx_channels[mode] = be32_to_cpu(values[0]); + dice->tx_midi_ports[mode] = be32_to_cpu(values[1]); + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, values, sizeof(values)); if (err < 0) @@ -280,13 +279,13 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) snd_dice_create_proc(dice); - err = snd_dice_stream_init(dice); + err = snd_dice_stream_init_duplex(dice); if (err < 0) goto error; err = snd_card_register(card); if (err < 0) { - snd_dice_stream_destroy(dice); + snd_dice_stream_destroy_duplex(dice); goto error; } @@ -304,7 +303,7 @@ static void dice_remove(struct fw_unit *unit) snd_card_disconnect(dice->card); - snd_dice_stream_destroy(dice); + snd_dice_stream_destroy_duplex(dice); snd_card_free_when_closed(dice->card); } @@ -317,7 +316,7 @@ static void dice_bus_reset(struct fw_unit *unit) snd_dice_transaction_reinit(dice); mutex_lock(&dice->mutex); - snd_dice_stream_update(dice); + snd_dice_stream_update_duplex(dice); mutex_unlock(&dice->mutex); } diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 8be530f..a62ee22 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -52,18 +52,28 @@ struct snd_dice { unsigned int rsrv_offset; unsigned int clock_caps; + unsigned int tx_channels[3]; unsigned int rx_channels[3]; + unsigned int tx_midi_ports[3]; unsigned int rx_midi_ports[3]; + struct fw_address_handler notification_handler; int owner_generation; + u32 notification_bits; + + /* For uapi */ int dev_lock_count; /* > 0 driver, < 0 userspace */ bool dev_lock_changed; - bool global_enabled; - struct completion clock_accepted; wait_queue_head_t hwdep_wait; - u32 notification_bits; + + /* For streaming */ + struct fw_iso_resources tx_resources; struct fw_iso_resources rx_resources; + struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; + bool global_enabled; + struct completion clock_accepted; + unsigned int substreams_counter; }; enum snd_dice_addr_type { @@ -160,11 +170,11 @@ extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, unsigned int *mode); -int snd_dice_stream_start(struct snd_dice *dice, unsigned int rate); -void snd_dice_stream_stop(struct snd_dice *dice); -int snd_dice_stream_init(struct snd_dice *dice); -void snd_dice_stream_destroy(struct snd_dice *dice); -void snd_dice_stream_update(struct snd_dice *dice); +int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate); +void snd_dice_stream_stop_duplex(struct snd_dice *dice); +int snd_dice_stream_init_duplex(struct snd_dice *dice); +void snd_dice_stream_destroy_duplex(struct snd_dice *dice); +void snd_dice_stream_update_duplex(struct snd_dice *dice); int snd_dice_stream_lock_try(struct snd_dice *dice); void snd_dice_stream_lock_release(struct snd_dice *dice); -- cgit v1.1 From 8fc01fc0674e3ea7fdd13bd3d138793619227f89 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:37 +0900 Subject: ALSA: dice: Support for non SYT-Match sampling clock source mode This commit allows this driver to handle devices with non SYT-Match sampling clock source. When sampling clock source is SYT-Match mode, devices handle 'presentation timestamp' in received packets and generates sampling clock according to the information. In this case, driver is synchronization master and must transfer correct value in SYT field of each packets in outgoing stream, then the outgoing stream is a master stream. On the other hand, non SYT-Match mode, devices do this. So drivers must pick up the value in SYT field of incoming packets and use the value for outgoing stream. Currently firewire-lib module achieve this work. Furthermore, without SYT-Match and internal clock source, the sampling rate should be fixed for the other devices connected to the handled device. This commit add a restriction of sampling rate at this situation. With these implementations, this driver has no need to set clock source. This commit remove set function. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice-pcm.c | 35 ++++++++++++++++++++++++++++++++++ sound/firewire/dice/dice-stream.c | 35 ++++++++++++++++++++++------------ sound/firewire/dice/dice-transaction.c | 5 ----- sound/firewire/dice/dice.h | 2 -- 4 files changed, 58 insertions(+), 19 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index b9ce026..062b7a3 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -140,6 +140,8 @@ end: static int pcm_open(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; + unsigned int source, rate; + bool internal; int err; err = snd_dice_stream_lock_try(dice); @@ -149,6 +151,39 @@ static int pcm_open(struct snd_pcm_substream *substream) err = init_hw_info(dice, substream); if (err < 0) goto err_locked; + + err = snd_dice_transaction_get_clock_source(dice, &source); + if (err < 0) + goto err_locked; + switch (source) { + case CLOCK_SOURCE_AES1: + case CLOCK_SOURCE_AES2: + case CLOCK_SOURCE_AES3: + case CLOCK_SOURCE_AES4: + case CLOCK_SOURCE_AES_ANY: + case CLOCK_SOURCE_ADAT: + case CLOCK_SOURCE_TDIF: + case CLOCK_SOURCE_WC: + internal = false; + break; + default: + internal = true; + break; + } + + /* + * When source of clock is not internal, available sampling rate is + * limited at current sampling rate. + */ + if (!internal) { + err = snd_dice_transaction_get_rate(dice, &rate); + if (err < 0) + goto err_locked; + substream->runtime->hw.rate_min = rate; + substream->runtime->hw.rate_max = rate; + } + + snd_pcm_set_sync(substream); end: return err; err_locked: diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index e60b84d..20765a0 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -161,9 +161,29 @@ end: static int get_sync_mode(struct snd_dice *dice, enum cip_flags *sync_mode) { - /* Currently, clock source is fixed at SYT-Match mode. */ - *sync_mode = 0; - return 0; + u32 source; + int err; + + err = snd_dice_transaction_get_clock_source(dice, &source); + if (err < 0) + goto end; + + switch (source) { + /* So-called 'SYT Match' modes, sync_to_syt value of packets received */ + case CLOCK_SOURCE_ARX4: /* in 4th stream */ + case CLOCK_SOURCE_ARX3: /* in 3rd stream */ + case CLOCK_SOURCE_ARX2: /* in 2nd stream */ + err = -ENOSYS; + break; + case CLOCK_SOURCE_ARX1: /* in 1st stream, which this driver uses */ + *sync_mode = 0; + break; + default: + *sync_mode = CIP_SYNC_TO_DEVICE; + break; + } +end: + return err; } int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) @@ -310,15 +330,6 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice) goto end; err = init_stream(dice, &dice->rx_stream); - if (err < 0) - goto end; - - /* Currently, clock source is fixed at SYT-Match mode. */ - err = snd_dice_transaction_set_clock_source(dice, CLOCK_SOURCE_ARX1); - if (err < 0) { - destroy_stream(dice, &dice->rx_stream); - destroy_stream(dice, &dice->tx_stream); - } end: return err; } diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 1fe304c..aee7461 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -137,11 +137,6 @@ int snd_dice_transaction_get_clock_source(struct snd_dice *dice, return err; } -int snd_dice_transaction_set_clock_source(struct snd_dice *dice, - unsigned int source) -{ - return set_clock_info(dice, UINT_MAX, source); -} int snd_dice_transaction_get_rate(struct snd_dice *dice, unsigned int *rate) { diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index a62ee22..f30326e 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -152,8 +152,6 @@ static inline int snd_dice_transaction_read_sync(struct snd_dice *dice, buf, len); } -int snd_dice_transaction_set_clock_source(struct snd_dice *dice, - unsigned int source); int snd_dice_transaction_get_clock_source(struct snd_dice *dice, unsigned int *source); int snd_dice_transaction_set_rate(struct snd_dice *dice, unsigned int rate); -- cgit v1.1 From 69dcf3e47a39f8f42e35245289691ca8321b46f1 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:38 +0900 Subject: ALSA: dice: Add support for capturing PCM samples This commit adds a support for capturing PCM samples. When opposite PCM substream is already running, available sampling rate is limited at current one. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 3 - sound/firewire/dice/dice-pcm.c | 161 +++++++++++++++++++++++++++++++++++++---- 2 files changed, 147 insertions(+), 17 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 2a5b9a6..0932860 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -20,9 +20,6 @@ config SND_DICE Say Y here to include support for many DACs based on the DICE chip family (DICE-II/Jr/Mini) which TC Applied Technologies produces. - At the moment, this driver supports playback only. If you - want to use devices that support capturing, use FFADO instead. - To compile this driver as a module, choose M here: the module will be called snd-dice. diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 062b7a3..f7771451 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -12,7 +12,8 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_dice *dice = rule->private; + struct snd_pcm_substream *substream = rule->private; + struct snd_dice *dice = substream->private_data; const struct snd_interval *c = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); @@ -21,7 +22,12 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, struct snd_interval rates = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; + unsigned int i, rate, mode, *pcm_channels; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + pcm_channels = dice->tx_channels; + else + pcm_channels = dice->rx_channels; for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { rate = snd_dice_rates[i]; @@ -41,7 +47,8 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, static int dice_channels_constraint(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_dice *dice = rule->private; + struct snd_pcm_substream *substream = rule->private; + struct snd_dice *dice = substream->private_data; const struct snd_interval *r = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); @@ -50,7 +57,12 @@ static int dice_channels_constraint(struct snd_pcm_hw_params *params, struct snd_interval channels = { .min = UINT_MAX, .max = 0, .integer = 1 }; - unsigned int i, rate, mode, *pcm_channels = dice->rx_channels; + unsigned int i, rate, mode, *pcm_channels; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + pcm_channels = dice->tx_channels; + else + pcm_channels = dice->rx_channels; for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { rate = snd_dice_rates[i]; @@ -109,30 +121,42 @@ static int init_hw_info(struct snd_dice *dice, { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; + struct amdtp_stream *stream; + unsigned int *pcm_channels; int err; hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_BLOCK_TRANSFER; - hw->formats = AMDTP_OUT_PCM_FORMAT_BITS; - limit_channels_and_rates(dice, runtime, dice->rx_channels); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + hw->formats = AMDTP_IN_PCM_FORMAT_BITS; + stream = &dice->tx_stream; + pcm_channels = dice->tx_channels; + } else { + hw->formats = AMDTP_OUT_PCM_FORMAT_BITS; + stream = &dice->rx_stream; + pcm_channels = dice->rx_channels; + } + + limit_channels_and_rates(dice, runtime, pcm_channels); limit_period_and_buffer(hw); err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - dice_rate_constraint, dice, + dice_rate_constraint, substream, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto end; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dice_channels_constraint, dice, + dice_channels_constraint, substream, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(&dice->rx_stream, runtime); + err = amdtp_stream_add_pcm_hw_constraints(stream, runtime); end: return err; } @@ -172,10 +196,12 @@ static int pcm_open(struct snd_pcm_substream *substream) } /* - * When source of clock is not internal, available sampling rate is - * limited at current sampling rate. + * When source of clock is not internal or any PCM streams are running, + * available sampling rate is limited at current sampling rate. */ - if (!internal) { + if (!internal || + amdtp_stream_pcm_running(&dice->tx_stream) || + amdtp_stream_pcm_running(&dice->rx_stream)) { err = snd_dice_transaction_get_rate(dice, &rate); if (err < 0) goto err_locked; @@ -200,10 +226,34 @@ static int pcm_close(struct snd_pcm_substream *substream) return 0; } +static int capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_dice *dice = substream->private_data; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&dice->mutex); + dice->substreams_counter++; + mutex_unlock(&dice->mutex); + } + + amdtp_stream_set_pcm_format(&dice->tx_stream, + params_format(hw_params)); + + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} static int playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&dice->mutex); + dice->substreams_counter++; + mutex_unlock(&dice->mutex); + } + amdtp_stream_set_pcm_format(&dice->rx_stream, params_format(hw_params)); @@ -211,17 +261,51 @@ static int playback_hw_params(struct snd_pcm_substream *substream, params_buffer_bytes(hw_params)); } +static int capture_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + mutex_lock(&dice->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + dice->substreams_counter--; + + snd_dice_stream_stop_duplex(dice); + + mutex_unlock(&dice->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + static int playback_hw_free(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; mutex_lock(&dice->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + dice->substreams_counter--; + snd_dice_stream_stop_duplex(dice); + mutex_unlock(&dice->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } +static int capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + int err; + + mutex_lock(&dice->mutex); + err = snd_dice_stream_start_duplex(dice, substream->runtime->rate); + mutex_unlock(&dice->mutex); + if (err >= 0) + amdtp_stream_pcm_prepare(&dice->tx_stream); + + return 0; +} static int playback_prepare(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; @@ -236,6 +320,23 @@ static int playback_prepare(struct snd_pcm_substream *substream) return err; } +static int capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_dice *dice = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + amdtp_stream_pcm_trigger(&dice->tx_stream, substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + amdtp_stream_pcm_trigger(&dice->tx_stream, NULL); + break; + default: + return -EINVAL; + } + + return 0; +} static int playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_dice *dice = substream->private_data; @@ -254,6 +355,12 @@ static int playback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static snd_pcm_uframes_t capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_dice *dice = substream->private_data; + + return amdtp_stream_pcm_pointer(&dice->tx_stream); +} static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; @@ -263,6 +370,18 @@ static snd_pcm_uframes_t playback_pointer(struct snd_pcm_substream *substream) int snd_dice_create_pcm(struct snd_dice *dice) { + static struct snd_pcm_ops capture_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = capture_hw_params, + .hw_free = capture_hw_free, + .prepare = capture_prepare, + .trigger = capture_trigger, + .pointer = capture_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; static struct snd_pcm_ops playback_ops = { .open = pcm_open, .close = pcm_close, @@ -276,14 +395,28 @@ int snd_dice_create_pcm(struct snd_dice *dice) .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; + unsigned int i, capture, playback; int err; - err = snd_pcm_new(dice->card, "DICE", 0, 1, 0, &pcm); + capture = playback = 0; + for (i = 0; i < 3; i++) { + if (dice->tx_channels[i] > 0) + capture = 1; + if (dice->rx_channels[i] > 0) + playback = 1; + } + + err = snd_pcm_new(dice->card, "DICE", 0, playback, capture, &pcm); if (err < 0) return err; pcm->private_data = dice; strcpy(pcm->name, dice->card->shortname); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); + + if (capture > 0) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + + if (playback > 0) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); return 0; } -- cgit v1.1 From a113ff886b9a6e892dd4107be1fd7883cf020885 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:39 +0900 Subject: ALSA: dice: Add support for MIDI capture/playback This commit adds a support for MIDI capture/playback When MIDI substrams already start streaming and PCM substreams are going to join at different sampling rate, streams are stopped once. Then sampling rate is changed and streams are restarted. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 157 ++++++++++++++++++++++++++++++++++++++ sound/firewire/dice/dice-stream.c | 2 + sound/firewire/dice/dice.c | 4 + sound/firewire/dice/dice.h | 3 + 5 files changed, 168 insertions(+), 2 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 9a48289..9ef228e 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,3 +1,3 @@ -snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-pcm.o \ - dice-hwdep.o dice.o +snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ + dice-pcm.o dice-hwdep.o dice.o obj-m += snd-dice.o diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c new file mode 100644 index 0000000..fe43ce7 --- /dev/null +++ b/sound/firewire/dice/dice-midi.c @@ -0,0 +1,157 @@ +/* + * dice_midi.c - a part of driver for Dice based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ +#include "dice.h" + +static int midi_open(struct snd_rawmidi_substream *substream) +{ + struct snd_dice *dice = substream->rmidi->private_data; + int err; + + err = snd_dice_stream_lock_try(dice); + if (err < 0) + return err; + + mutex_lock(&dice->mutex); + + dice->substreams_counter++; + err = snd_dice_stream_start_duplex(dice, 0); + + mutex_unlock(&dice->mutex); + + if (err < 0) + snd_dice_stream_lock_release(dice); + + return err; +} + +static int midi_close(struct snd_rawmidi_substream *substream) +{ + struct snd_dice *dice = substream->rmidi->private_data; + + mutex_lock(&dice->mutex); + + dice->substreams_counter--; + snd_dice_stream_stop_duplex(dice); + + mutex_unlock(&dice->mutex); + + snd_dice_stream_lock_release(dice); + return 0; +} + +static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_dice *dice = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dice->lock, flags); + + if (up) + amdtp_stream_midi_trigger(&dice->tx_stream, + substrm->number, substrm); + else + amdtp_stream_midi_trigger(&dice->tx_stream, + substrm->number, NULL); + + spin_unlock_irqrestore(&dice->lock, flags); +} + +static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_dice *dice = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&dice->lock, flags); + + if (up) + amdtp_stream_midi_trigger(&dice->rx_stream, + substrm->number, substrm); + else + amdtp_stream_midi_trigger(&dice->rx_stream, + substrm->number, NULL); + + spin_unlock_irqrestore(&dice->lock, flags); +} + +static struct snd_rawmidi_ops capture_ops = { + .open = midi_open, + .close = midi_close, + .trigger = midi_capture_trigger, +}; + +static struct snd_rawmidi_ops playback_ops = { + .open = midi_open, + .close = midi_close, + .trigger = midi_playback_trigger, +}; + +static void set_midi_substream_names(struct snd_dice *dice, + struct snd_rawmidi_str *str) +{ + struct snd_rawmidi_substream *subs; + + list_for_each_entry(subs, &str->substreams, list) { + snprintf(subs->name, sizeof(subs->name), + "%s MIDI %d", dice->card->shortname, subs->number + 1); + } +} + +int snd_dice_create_midi(struct snd_dice *dice) +{ + struct snd_rawmidi *rmidi; + struct snd_rawmidi_str *str; + unsigned int i, midi_in_ports, midi_out_ports; + int err; + + midi_in_ports = midi_out_ports = 0; + for (i = 0; i < 3; i++) { + midi_in_ports = max(dice->tx_midi_ports[i], midi_in_ports); + midi_out_ports = max(dice->rx_midi_ports[i], midi_out_ports); + } + + if (midi_in_ports + midi_out_ports == 0) + return 0; + + /* create midi ports */ + err = snd_rawmidi_new(dice->card, dice->card->driver, 0, + midi_out_ports, midi_in_ports, + &rmidi); + if (err < 0) + return err; + + snprintf(rmidi->name, sizeof(rmidi->name), + "%s MIDI", dice->card->shortname); + rmidi->private_data = dice; + + if (midi_in_ports > 0) { + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &capture_ops); + + str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]; + + set_midi_substream_names(dice, str); + } + + if (midi_out_ports > 0) { + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &playback_ops); + + str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; + + set_midi_substream_names(dice, str); + } + + if ((midi_out_ports > 0) && (midi_in_ports > 0)) + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; + + return 0; +} diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 20765a0..fa9cf76 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -218,6 +218,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) "fail to get sampling rate\n"); goto end; } + if (rate == 0) + rate = curr_rate; if (rate != curr_rate) stop_stream(dice, master); diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 85bcfaf3..90d8f40 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -279,6 +279,10 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) snd_dice_create_proc(dice); + err = snd_dice_create_midi(dice); + if (err < 0) + goto error; + err = snd_dice_stream_init_duplex(dice); if (err < 0) goto error; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index f30326e..ecf5dc8 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -32,6 +32,7 @@ #include #include #include +#include #include "../amdtp.h" #include "../iso-resources.h" @@ -183,4 +184,6 @@ int snd_dice_create_hwdep(struct snd_dice *dice); void snd_dice_create_proc(struct snd_dice *dice); +int snd_dice_create_midi(struct snd_dice *dice); + #endif -- cgit v1.1 From fec7b7536207897693e708411dcb95909d56c431 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:40 +0900 Subject: ALSA: oxfw: Change the way to name card This is a preparation for more models. In following commit, members of 'struct snd_card' related to name becomes to consists of vendor and model strings in device's config-rom. Current supported devices also has strings in their config rom, but the strings are too long to name sound card, thus this driver still keep hard-coded vendor and model names for them. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw.c | 53 ++++++++++++++++++++++++++++------------------ sound/firewire/oxfw/oxfw.h | 4 ++-- 2 files changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 951d9a4..dd576bf 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,14 +25,34 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("snd-firewire-speakers"); -static u32 oxfw_read_firmware_version(struct fw_unit *unit) +static int name_card(struct snd_oxfw *oxfw) { - __be32 data; + struct fw_device *fw_dev = fw_parent_device(oxfw->unit); + const char *d, *v, *m; + u32 firmware; int err; - err = snd_fw_transaction(unit, TCODE_READ_QUADLET_REQUEST, - OXFORD_FIRMWARE_ID_ADDRESS, &data, 4, 0); - return err >= 0 ? be32_to_cpu(data) : 0; + err = snd_fw_transaction(oxfw->unit, TCODE_READ_QUADLET_REQUEST, + OXFORD_FIRMWARE_ID_ADDRESS, &firmware, 4, 0); + if (err < 0) + goto end; + be32_to_cpus(&firmware); + + d = oxfw->device_info->driver_name; + v = oxfw->device_info->vendor_name; + m = oxfw->device_info->model_name; + + strcpy(oxfw->card->driver, d); + strcpy(oxfw->card->mixername, m); + strcpy(oxfw->card->shortname, m); + + snprintf(oxfw->card->longname, sizeof(oxfw->card->longname), + "%s %s (OXFW%x %04x), GUID %08x%08x at %s, S%d", + v, m, firmware >> 20, firmware & 0xffff, + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&oxfw->unit->device), 100 << fw_dev->max_speed); +end: + return err; } static void oxfw_card_free(struct snd_card *card) @@ -45,10 +65,8 @@ static void oxfw_card_free(struct snd_card *card) static int oxfw_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { - struct fw_device *fw_dev = fw_parent_device(unit); struct snd_card *card; struct snd_oxfw *oxfw; - u32 firmware; int err; err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, @@ -63,16 +81,9 @@ static int oxfw_probe(struct fw_unit *unit, oxfw->unit = unit; oxfw->device_info = (const struct device_info *)id->driver_data; - strcpy(card->driver, oxfw->device_info->driver_name); - strcpy(card->shortname, oxfw->device_info->short_name); - firmware = oxfw_read_firmware_version(unit); - snprintf(card->longname, sizeof(card->longname), - "%s (OXFW%x %04x), GUID %08x%08x at %s, S%d", - oxfw->device_info->long_name, - firmware >> 20, firmware & 0xffff, - fw_dev->config_rom[3], fw_dev->config_rom[4], - dev_name(&unit->device), 100 << fw_dev->max_speed); - strcpy(card->mixername, "OXFW"); + err = name_card(oxfw); + if (err < 0) + goto error; err = snd_oxfw_create_pcm(oxfw); if (err < 0) @@ -123,8 +134,8 @@ static void oxfw_remove(struct fw_unit *unit) static const struct device_info griffin_firewave = { .driver_name = "FireWave", - .short_name = "FireWave", - .long_name = "Griffin FireWave Surround", + .vendor_name = "Griffin", + .model_name = "FireWave", .pcm_constraints = firewave_constraints, .mixer_channels = 6, .mute_fb_id = 0x01, @@ -133,8 +144,8 @@ static const struct device_info griffin_firewave = { static const struct device_info lacie_speakers = { .driver_name = "FWSpeakers", - .short_name = "FireWire Speakers", - .long_name = "LaCie FireWire Speakers", + .vendor_name = "LaCie", + .model_name = "FireWire Speakers", .pcm_constraints = lacie_speakers_constraints, .mixer_channels = 1, .mute_fb_id = 0x01, diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 6164bf3..a61c75c 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -28,8 +28,8 @@ struct device_info { const char *driver_name; - const char *short_name; - const char *long_name; + const char *vendor_name; + const char *model_name; int (*pcm_constraints)(struct snd_pcm_runtime *runtime); unsigned int mixer_channels; u8 mute_fb_id; -- cgit v1.1 From 5b59d8098d2a3fa8ea4ad07b96f62c00c3b3e8d3 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:41 +0900 Subject: ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation OXFW970/971 may supports AV/C Stream Format Information Specification 1.1 Working Draft (Apr 2005, 1394TA). By using this command, drivers can get to know stream formations which device supports. This commit adds 'EXTENDED STREAM FORMAT INFORMATION' command. This command has two subfunctions, 'SINGLE' and 'LIST'. Drivers can use 'SINGLE' subfunction to know/set current formation of AMDTP stream, Drivers can use 'LIST' subfunction to know an available formation of AMDTP stream in a certain sampling rate. But some devices don't implement the 'LIST' subfunction. So this commit uses an assumption that 'if they don't implement it, they don't change stream formation depending on current each sampling rate'. With this assumption, this driver generates formations for such devices by: 1.getting current formation by SINGLE subfunction 2.getting supported sampling rates 3.applying current formation for all of supported sampling rates Followed commit implements a parser of this format information. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-command.c | 153 +++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.h | 33 ++++++++ 3 files changed, 187 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/oxfw/oxfw-command.c diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index 0cf48fd..b107134 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,2 +1,2 @@ -snd-oxfw-objs := oxfw-stream.o oxfw-control.o oxfw-pcm.o oxfw.o +snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-command.c b/sound/firewire/oxfw/oxfw-command.c new file mode 100644 index 0000000..12ef325 --- /dev/null +++ b/sound/firewire/oxfw/oxfw-command.c @@ -0,0 +1,153 @@ +/* + * oxfw_command.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "oxfw.h" + +int avc_stream_set_format(struct fw_unit *unit, enum avc_general_plug_dir dir, + unsigned int pid, u8 *format, unsigned int len) +{ + u8 *buf; + int err; + + buf = kmalloc(len + 10, GFP_KERNEL); + if (buf == NULL) + return -ENOMEM; + + buf[0] = 0x00; /* CONTROL */ + buf[1] = 0xff; /* UNIT */ + buf[2] = 0xbf; /* EXTENDED STREAM FORMAT INFORMATION */ + buf[3] = 0xc0; /* SINGLE subfunction */ + buf[4] = dir; /* Plug Direction */ + buf[5] = 0x00; /* UNIT */ + buf[6] = 0x00; /* PCR (Isochronous Plug) */ + buf[7] = 0xff & pid; /* Plug ID */ + buf[8] = 0xff; /* Padding */ + buf[9] = 0xff; /* Support status in response */ + memcpy(buf + 10, format, len); + + /* do transaction and check buf[1-8] are the same against command */ + err = fcp_avc_transaction(unit, buf, len + 10, buf, len + 10, + BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | + BIT(6) | BIT(7) | BIT(8)); + if ((err > 0) && (err < len + 10)) + err = -EIO; + else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ + err = -ENOSYS; + else if (buf[0] == 0x0a) /* REJECTED */ + err = -EINVAL; + else + err = 0; + + kfree(buf); + + return err; +} + +int avc_stream_get_format(struct fw_unit *unit, + enum avc_general_plug_dir dir, unsigned int pid, + u8 *buf, unsigned int *len, unsigned int eid) +{ + unsigned int subfunc; + int err; + + if (eid == 0xff) + subfunc = 0xc0; /* SINGLE */ + else + subfunc = 0xc1; /* LIST */ + + buf[0] = 0x01; /* STATUS */ + buf[1] = 0xff; /* UNIT */ + buf[2] = 0xbf; /* EXTENDED STREAM FORMAT INFORMATION */ + buf[3] = subfunc; /* SINGLE or LIST */ + buf[4] = dir; /* Plug Direction */ + buf[5] = 0x00; /* Unit */ + buf[6] = 0x00; /* PCR (Isochronous Plug) */ + buf[7] = 0xff & pid; /* Plug ID */ + buf[8] = 0xff; /* Padding */ + buf[9] = 0xff; /* support status in response */ + buf[10] = 0xff & eid; /* entry ID for LIST subfunction */ + buf[11] = 0xff; /* padding */ + + /* do transaction and check buf[1-7] are the same against command */ + err = fcp_avc_transaction(unit, buf, 12, buf, *len, + BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | + BIT(6) | BIT(7)); + if ((err > 0) && (err < 10)) + err = -EIO; + else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ + err = -ENOSYS; + else if (buf[0] == 0x0a) /* REJECTED */ + err = -EINVAL; + else if (buf[0] == 0x0b) /* IN TRANSITION */ + err = -EAGAIN; + /* LIST subfunction has entry ID */ + else if ((subfunc == 0xc1) && (buf[10] != eid)) + err = -EIO; + if (err < 0) + goto end; + + /* keep just stream format information */ + if (subfunc == 0xc0) { + memmove(buf, buf + 10, err - 10); + *len = err - 10; + } else { + memmove(buf, buf + 11, err - 11); + *len = err - 11; + } + + err = 0; +end: + return err; +} + +int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, + enum avc_general_plug_dir dir, + unsigned short pid) +{ + unsigned int sfc; + u8 *buf; + int err; + + for (sfc = 0; sfc < CIP_SFC_COUNT; sfc++) { + if (amdtp_rate_table[sfc] == rate) + break; + } + if (sfc == CIP_SFC_COUNT) + return -EINVAL; + + buf = kzalloc(8, GFP_KERNEL); + if (buf == NULL) + return -ENOMEM; + + buf[0] = 0x02; /* SPECIFIC INQUIRY */ + buf[1] = 0xff; /* UNIT */ + if (dir == AVC_GENERAL_PLUG_DIR_IN) + buf[2] = 0x19; /* INPUT PLUG SIGNAL FORMAT */ + else + buf[2] = 0x18; /* OUTPUT PLUG SIGNAL FORMAT */ + buf[3] = 0xff & pid; /* plug id */ + buf[4] = 0x90; /* EOH_1, Form_1, FMT. AM824 */ + buf[5] = 0x07 & sfc; /* FDF-hi. AM824, frequency */ + buf[6] = 0xff; /* FDF-mid. AM824, SYT hi (not used) */ + buf[7] = 0xff; /* FDF-low. AM824, SYT lo (not used) */ + + /* do transaction and check buf[1-5] are the same against command */ + err = fcp_avc_transaction(unit, buf, 8, buf, 8, + BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5)); + if ((err > 0) && (err < 8)) + err = -EIO; + else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ + err = -ENOSYS; + if (err < 0) + goto end; + + err = 0; +end: + kfree(buf); + return err; +} diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index a61c75c..a7031d4 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -49,6 +49,39 @@ struct snd_oxfw { s16 volume_max; }; +/* + * AV/C Stream Format Information Specification 1.1 Working Draft + * (Apr 2005, 1394TA) + */ +int avc_stream_set_format(struct fw_unit *unit, enum avc_general_plug_dir dir, + unsigned int pid, u8 *format, unsigned int len); +int avc_stream_get_format(struct fw_unit *unit, + enum avc_general_plug_dir dir, unsigned int pid, + u8 *buf, unsigned int *len, unsigned int eid); +static inline int +avc_stream_get_format_single(struct fw_unit *unit, + enum avc_general_plug_dir dir, unsigned int pid, + u8 *buf, unsigned int *len) +{ + return avc_stream_get_format(unit, dir, pid, buf, len, 0xff); +} +static inline int +avc_stream_get_format_list(struct fw_unit *unit, + enum avc_general_plug_dir dir, unsigned int pid, + u8 *buf, unsigned int *len, + unsigned int eid) +{ + return avc_stream_get_format(unit, dir, pid, buf, len, eid); +} + +/* + * AV/C Digital Interface Command Set General Specification 4.2 + * (Sep 2004, 1394TA) + */ +int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, + enum avc_general_plug_dir dir, + unsigned short pid); + int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw); int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); -- cgit v1.1 From 5cd1d3f47a6321612a51ab88ffe8ef65120fcbe0 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:42 +0900 Subject: ALSA: oxfw: Change the way to make PCM rules/constraints In previous commit, this driver can get to know stream formations at each supported sampling rates. This commit uses it to make PCM rules/constraints and obsoletes hard-coded rules/constraints. For this purpose, this commit adds 'struct snd_oxfw_stream_formation' and snd_oxfw_stream_parse_format() to parse data channel formation of data block. According to datasheet of OXFW970/971, they support 32.0kHz to 196.0kHz. As long as developers investigate, some devices are confirmed to have several formats for the same sampling rate. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 197 ++++++++++++++++------------ sound/firewire/oxfw/oxfw-stream.c | 268 ++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 10 +- sound/firewire/oxfw/oxfw.h | 21 ++- 4 files changed, 410 insertions(+), 86 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index d39f17a..0c0be98 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -7,117 +7,152 @@ #include "oxfw.h" -static int firewave_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { - static unsigned int stereo_rates[] = { 48000, 96000 }; - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - - /* two channels work only at 48/96 kHz */ - if (snd_interval_max(channels) < 6) - return snd_interval_list(rate, 2, stereo_rates, 0); - return 0; + u8 **formats = rule->private; + struct snd_interval *r = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + const struct snd_interval *c = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval t = { + .min = UINT_MAX, .max = 0, .integer = 1 + }; + struct snd_oxfw_stream_formation formation; + unsigned int i, err; + + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + if (formats[i] == NULL) + continue; + + err = snd_oxfw_stream_parse_format(formats[i], &formation); + if (err < 0) + continue; + if (!snd_interval_test(c, formation.pcm)) + continue; + + t.min = min(t.min, formation.rate); + t.max = max(t.max, formation.rate); + + } + return snd_interval_refine(r, &t); } -static int firewave_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { - static const struct snd_interval all_channels = { .min = 6, .max = 6 }; - struct snd_interval *rate = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *channels = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - - /* 32/44.1 kHz work only with all six channels */ - if (snd_interval_max(rate) < 48000) - return snd_interval_refine(channels, &all_channels); - return 0; + u8 **formats = rule->private; + struct snd_interval *c = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + const struct snd_interval *r = + hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_oxfw_stream_formation formation; + unsigned int i, j, err; + unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0}; + + count = 0; + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + if (formats[i] == NULL) + break; + + err = snd_oxfw_stream_parse_format(formats[i], &formation); + if (err < 0) + continue; + if (!snd_interval_test(r, formation.rate)) + continue; + if (list[count] == formation.pcm) + continue; + + for (j = 0; j < ARRAY_SIZE(list); j++) { + if (list[j] == formation.pcm) + break; + } + if (j == ARRAY_SIZE(list)) { + list[count] = formation.pcm; + if (++count == ARRAY_SIZE(list)) + break; + } + } + + return snd_interval_list(c, count, list, 0); } -int firewave_constraints(struct snd_pcm_runtime *runtime) +static void limit_channels_and_rates(struct snd_pcm_hardware *hw, u8 **formats) { - static unsigned int channels_list[] = { 2, 6 }; - static struct snd_pcm_hw_constraint_list channels_list_constraint = { - .count = 2, - .list = channels_list, - }; - int err; + struct snd_oxfw_stream_formation formation; + unsigned int i, err; - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000; - runtime->hw.channels_max = 6; + hw->channels_min = UINT_MAX; + hw->channels_max = 0; - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &channels_list_constraint); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - firewave_rate_constraint, NULL, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - firewave_channels_constraint, NULL, - SNDRV_PCM_HW_PARAM_RATE, -1); - if (err < 0) - return err; + hw->rate_min = UINT_MAX; + hw->rate_max = 0; + hw->rates = 0; - return 0; + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + if (formats[i] == NULL) + break; + + err = snd_oxfw_stream_parse_format(formats[i], &formation); + if (err < 0) + continue; + + hw->channels_min = min(hw->channels_min, formation.pcm); + hw->channels_max = max(hw->channels_max, formation.pcm); + + hw->rate_min = min(hw->rate_min, formation.rate); + hw->rate_max = max(hw->rate_max, formation.rate); + hw->rates |= snd_pcm_rate_to_rate_bit(formation.rate); + } } -int lacie_speakers_constraints(struct snd_pcm_runtime *runtime) +static void limit_period_and_buffer(struct snd_pcm_hardware *hw) { - runtime->hw.rates = SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000; + hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */ + hw->periods_max = UINT_MAX; - return 0; + hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */ + + /* Just to prevent from allocating much pages. */ + hw->period_bytes_max = hw->period_bytes_min * 2048; + hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; } static int pcm_open(struct snd_pcm_substream *substream) { - static const struct snd_pcm_hardware hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .formats = AMDTP_OUT_PCM_FORMAT_BITS, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 4 * 1024 * 1024, - .period_bytes_min = 1, - .period_bytes_max = UINT_MAX, - .periods_min = 1, - .periods_max = UINT_MAX, - }; struct snd_oxfw *oxfw = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - bool used; + u8 **formats; int err; - err = cmp_connection_check_used(&oxfw->in_conn, &used); - if ((err < 0) || used) - goto end; + formats = oxfw->rx_stream_formats; + + runtime->hw.info = SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID; - runtime->hw = hardware; + limit_channels_and_rates(&runtime->hw, formats); + limit_period_and_buffer(&runtime->hw); - err = oxfw->device_info->pcm_constraints(runtime); + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, formats, + SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto end; - err = snd_pcm_limit_hw_rates(runtime); + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, formats, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto end; err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); + if (err < 0) + goto end; + + snd_pcm_set_sync(substream); end: return err; } diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index ebd156f..17e3802 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -8,6 +8,35 @@ #include "oxfw.h" +#define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512 + +/* + * According to datasheet of Oxford Semiconductor: + * OXFW970: 32.0/44.1/48.0/96.0 Khz, 8 audio channels I/O + * OXFW971: 32.0/44.1/48.0/88.2/96.0/192.0 kHz, 16 audio channels I/O, MIDI I/O + */ +static const unsigned int oxfw_rate_table[] = { + [0] = 32000, + [1] = 44100, + [2] = 48000, + [3] = 88200, + [4] = 96000, + [5] = 192000, +}; + +/* + * See Table 5.7 – Sampling frequency for Multi-bit Audio + * in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA) + */ +static const unsigned int avc_stream_rate_table[] = { + [0] = 0x02, + [1] = 0x03, + [2] = 0x04, + [3] = 0x0a, + [4] = 0x05, + [5] = 0x07, +}; + int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw) { int err; @@ -78,3 +107,242 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw) else amdtp_stream_update(&oxfw->rx_stream); } + +/* + * See Table 6.16 - AM824 Stream Format + * Figure 6.19 - format_information field for AM824 Compound + * in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA) + * Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005 + */ +int snd_oxfw_stream_parse_format(u8 *format, + struct snd_oxfw_stream_formation *formation) +{ + unsigned int i, e, channels, type; + + memset(formation, 0, sizeof(struct snd_oxfw_stream_formation)); + + /* + * this module can support a hierarchy combination that: + * Root: Audio and Music (0x90) + * Level 1: AM824 Compound (0x40) + */ + if ((format[0] != 0x90) || (format[1] != 0x40)) + return -ENOSYS; + + /* check the sampling rate */ + for (i = 0; i < ARRAY_SIZE(avc_stream_rate_table); i++) { + if (format[2] == avc_stream_rate_table[i]) + break; + } + if (i == ARRAY_SIZE(avc_stream_rate_table)) + return -ENOSYS; + + formation->rate = oxfw_rate_table[i]; + + for (e = 0; e < format[4]; e++) { + channels = format[5 + e * 2]; + type = format[6 + e * 2]; + + switch (type) { + /* IEC 60958 Conformant, currently handled as MBLA */ + case 0x00: + /* Multi Bit Linear Audio (Raw) */ + case 0x06: + formation->pcm += channels; + break; + /* MIDI Conformant */ + case 0x0d: + formation->midi = channels; + break; + /* IEC 61937-3 to 7 */ + case 0x01: + case 0x02: + case 0x03: + case 0x04: + case 0x05: + /* Multi Bit Linear Audio */ + case 0x07: /* DVD-Audio */ + case 0x0c: /* High Precision */ + /* One Bit Audio */ + case 0x08: /* (Plain) Raw */ + case 0x09: /* (Plain) SACD */ + case 0x0a: /* (Encoded) Raw */ + case 0x0b: /* (Encoded) SACD */ + /* SMPTE Time-Code conformant */ + case 0x0e: + /* Sample Count */ + case 0x0f: + /* Anciliary Data */ + case 0x10: + /* Synchronization Stream (Stereo Raw audio) */ + case 0x40: + /* Don't care */ + case 0xff: + default: + return -ENOSYS; /* not supported */ + } + } + + if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM || + formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI) + return -ENOSYS; + + return 0; +} + +static int +assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, + unsigned int pid, u8 *buf, unsigned int *len, + u8 **formats) +{ + struct snd_oxfw_stream_formation formation; + unsigned int i, eid; + int err; + + /* get format at current sampling rate */ + err = avc_stream_get_format_single(oxfw->unit, dir, pid, buf, len); + if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to get current stream format for isoc %s plug %d:%d\n", + (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out", + pid, err); + goto end; + } + + /* parse and set stream format */ + eid = 0; + err = snd_oxfw_stream_parse_format(buf, &formation); + if (err < 0) + goto end; + + formats[eid] = kmalloc(*len, GFP_KERNEL); + if (formats[eid] == NULL) { + err = -ENOMEM; + goto end; + } + memcpy(formats[eid], buf, *len); + + /* apply the format for each available sampling rate */ + for (i = 0; i < ARRAY_SIZE(oxfw_rate_table); i++) { + if (formation.rate == oxfw_rate_table[i]) + continue; + + err = avc_general_inquiry_sig_fmt(oxfw->unit, + oxfw_rate_table[i], + dir, pid); + if (err < 0) + continue; + + eid++; + formats[eid] = kmalloc(*len, GFP_KERNEL); + if (formats[eid] == NULL) { + err = -ENOMEM; + goto end; + } + memcpy(formats[eid], buf, *len); + formats[eid][2] = avc_stream_rate_table[i]; + } + + err = 0; + oxfw->assumed = true; +end: + return err; +} + +static int fill_stream_formats(struct snd_oxfw *oxfw, + enum avc_general_plug_dir dir, + unsigned short pid) +{ + u8 *buf, **formats; + unsigned int len, eid = 0; + struct snd_oxfw_stream_formation dummy; + int err; + + buf = kmalloc(AVC_GENERIC_FRAME_MAXIMUM_BYTES, GFP_KERNEL); + if (buf == NULL) + return -ENOMEM; + + formats = oxfw->rx_stream_formats; + + /* get first entry */ + len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; + err = avc_stream_get_format_list(oxfw->unit, dir, 0, buf, &len, 0); + if (err == -ENOSYS) { + /* LIST subfunction is not implemented */ + len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; + err = assume_stream_formats(oxfw, dir, pid, buf, &len, + formats); + goto end; + } else if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to get stream format %d for isoc %s plug %d:%d\n", + eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out", + pid, err); + goto end; + } + + /* LIST subfunction is implemented */ + while (eid < SND_OXFW_STREAM_FORMAT_ENTRIES) { + /* The format is too short. */ + if (len < 3) { + err = -EIO; + break; + } + + /* parse and set stream format */ + err = snd_oxfw_stream_parse_format(buf, &dummy); + if (err < 0) + break; + + formats[eid] = kmalloc(len, GFP_KERNEL); + if (formats[eid] == NULL) { + err = -ENOMEM; + break; + } + memcpy(formats[eid], buf, len); + + /* get next entry */ + len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; + err = avc_stream_get_format_list(oxfw->unit, dir, 0, + buf, &len, ++eid); + /* No entries remained. */ + if (err == -EINVAL) { + err = 0; + break; + } else if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to get stream format %d for isoc %s plug %d:%d\n", + eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : + "out", + pid, err); + break; + } + } +end: + kfree(buf); + return err; +} + +int snd_oxfw_stream_discover(struct snd_oxfw *oxfw) +{ + u8 plugs[AVC_PLUG_INFO_BUF_BYTES]; + int err; + + /* the number of plugs for isoc in/out, ext in/out */ + err = avc_general_get_plug_info(oxfw->unit, 0x1f, 0x07, 0x00, plugs); + if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to get info for isoc/external in/out plugs: %d\n", + err); + goto end; + } else if (plugs[0] == 0) { + err = -ENOSYS; + goto end; + } + + /* use iPCR[0] if exists */ + if (plugs[0] > 0) + err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0); +end: + return err; +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index dd576bf..a8f9062 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -58,6 +58,10 @@ end: static void oxfw_card_free(struct snd_card *card) { struct snd_oxfw *oxfw = card->private_data; + unsigned int i; + + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) + kfree(oxfw->rx_stream_formats[i]); mutex_destroy(&oxfw->mutex); } @@ -81,6 +85,10 @@ static int oxfw_probe(struct fw_unit *unit, oxfw->unit = unit; oxfw->device_info = (const struct device_info *)id->driver_data; + err = snd_oxfw_stream_discover(oxfw); + if (err < 0) + goto error; + err = name_card(oxfw); if (err < 0) goto error; @@ -136,7 +144,6 @@ static const struct device_info griffin_firewave = { .driver_name = "FireWave", .vendor_name = "Griffin", .model_name = "FireWave", - .pcm_constraints = firewave_constraints, .mixer_channels = 6, .mute_fb_id = 0x01, .volume_fb_id = 0x02, @@ -146,7 +153,6 @@ static const struct device_info lacie_speakers = { .driver_name = "FWSpeakers", .vendor_name = "LaCie", .model_name = "FireWire Speakers", - .pcm_constraints = lacie_speakers_constraints, .mixer_channels = 1, .mute_fb_id = 0x01, .volume_fb_id = 0x01, diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index a7031d4..9c3d3e3 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -30,19 +30,24 @@ struct device_info { const char *driver_name; const char *vendor_name; const char *model_name; - int (*pcm_constraints)(struct snd_pcm_runtime *runtime); unsigned int mixer_channels; u8 mute_fb_id; u8 volume_fb_id; }; +/* This is an arbitrary number for convinience. */ +#define SND_OXFW_STREAM_FORMAT_ENTRIES 10 struct snd_oxfw { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; struct mutex mutex; + + u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; + bool assumed; struct cmp_connection in_conn; struct amdtp_stream rx_stream; + bool mute; s16 volume[6]; s16 volume_min; @@ -88,8 +93,18 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); -int firewave_constraints(struct snd_pcm_runtime *runtime); -int lacie_speakers_constraints(struct snd_pcm_runtime *runtime); +struct snd_oxfw_stream_formation { + unsigned int rate; + unsigned int pcm; + unsigned int midi; +}; +int snd_oxfw_stream_parse_format(u8 *format, + struct snd_oxfw_stream_formation *formation); +int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, + enum avc_general_plug_dir dir, + struct snd_oxfw_stream_formation *formation); +int snd_oxfw_stream_discover(struct snd_oxfw *oxfw); + int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); -- cgit v1.1 From 3c96101f190020e91d413c5835f7a722fc007923 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:43 +0900 Subject: ALSA: oxfw: Add proc interface for debugging purpose This commit adds proc interface to get information about stream formation. This commit also adds snd_oxfw_stream_get_current_formation() to get current stream formation. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-proc.c | 84 +++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw-stream.c | 27 +++++++++++++ sound/firewire/oxfw/oxfw.c | 2 + sound/firewire/oxfw/oxfw.h | 3 ++ 5 files changed, 118 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/oxfw/oxfw-proc.c diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index b107134..e9297c6 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,2 +1,3 @@ -snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o oxfw.o +snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ + oxfw-proc.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c new file mode 100644 index 0000000..18e0305 --- /dev/null +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -0,0 +1,84 @@ +/* + * oxfw_proc.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "./oxfw.h" + +static void proc_read_formation(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_oxfw *oxfw = entry->private_data; + struct snd_oxfw_stream_formation formation, curr; + u8 *format; + char flag; + unsigned int i, err; + + /* Show input. */ + err = snd_oxfw_stream_get_current_formation(oxfw, + AVC_GENERAL_PLUG_DIR_IN, + &curr); + if (err < 0) + return; + + snd_iprintf(buffer, "Input Stream to device:\n"); + snd_iprintf(buffer, "\tRate\tPCM\tMIDI\n"); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + format = oxfw->rx_stream_formats[i]; + if (format == NULL) + continue; + + err = snd_oxfw_stream_parse_format(format, &formation); + if (err < 0) + continue; + + if (memcmp(&formation, &curr, sizeof(curr)) == 0) + flag = '*'; + else + flag = ' '; + + snd_iprintf(buffer, "%c\t%d\t%d\t%d\n", flag, + formation.rate, formation.pcm, formation.midi); + } + +} + +static void add_node(struct snd_oxfw *oxfw, struct snd_info_entry *root, + const char *name, + void (*op)(struct snd_info_entry *e, + struct snd_info_buffer *b)) +{ + struct snd_info_entry *entry; + + entry = snd_info_create_card_entry(oxfw->card, name, root); + if (entry == NULL) + return; + + snd_info_set_text_ops(entry, oxfw, op); + if (snd_info_register(entry) < 0) + snd_info_free_entry(entry); +} + +void snd_oxfw_proc_init(struct snd_oxfw *oxfw) +{ + struct snd_info_entry *root; + + /* + * All nodes are automatically removed at snd_card_disconnect(), + * by following to link list. + */ + root = snd_info_create_card_entry(oxfw->card, "firewire", + oxfw->card->proc_root); + if (root == NULL) + return; + root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + if (snd_info_register(root) < 0) { + snd_info_free_entry(root); + return; + } + + add_node(oxfw, root, "formation", proc_read_formation); +} diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 17e3802..210bf5a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -108,6 +108,33 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw) amdtp_stream_update(&oxfw->rx_stream); } +int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, + enum avc_general_plug_dir dir, + struct snd_oxfw_stream_formation *formation) +{ + u8 *format; + unsigned int len; + int err; + + len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; + format = kmalloc(len, GFP_KERNEL); + if (format == NULL) + return -ENOMEM; + + err = avc_stream_get_format_single(oxfw->unit, dir, 0, format, &len); + if (err < 0) + goto end; + if (len < 3) { + err = -EIO; + goto end; + } + + err = snd_oxfw_stream_parse_format(format, formation); +end: + kfree(format); + return err; +} + /* * See Table 6.16 - AM824 Stream Format * Figure 6.19 - format_information field for AM824 Compound diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index a8f9062..a70149a 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -101,6 +101,8 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; + snd_oxfw_proc_init(oxfw); + err = snd_oxfw_stream_init_simplex(oxfw); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 9c3d3e3..8c832ea 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -18,6 +18,7 @@ #include #include #include +#include #include "../lib.h" #include "../fcp.h" @@ -108,3 +109,5 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw); int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); + +void snd_oxfw_proc_init(struct snd_oxfw *oxfw); -- cgit v1.1 From f3699e2c77455a6cccc977b391c553f2c816f639 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:44 +0900 Subject: ALSA: oxfw: Change the way to start stream In past commit, this driver can keep stream formations for each sampling rate. So its stream functionality can decide stream formations with given some parameters. This commit moves related codes from PCM functionality to stream functionality. Furthermore, to set stream format correctly, this commit uses AV/C Stream Format Information command instead of AV/C Input/Output Plug Signal Format command. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 44 ++---------- sound/firewire/oxfw/oxfw-stream.c | 139 +++++++++++++++++++++++++++++++++++--- sound/firewire/oxfw/oxfw.h | 3 +- 3 files changed, 138 insertions(+), 48 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 0c0be98..ea2b439 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -166,39 +166,10 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; - int err; - - mutex_lock(&oxfw->mutex); - - snd_oxfw_stream_stop_simplex(oxfw); - - err = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - goto error; - - amdtp_stream_set_parameters(&oxfw->rx_stream, - params_rate(hw_params), - params_channels(hw_params), - 0); - - amdtp_stream_set_pcm_format(&oxfw->rx_stream, - params_format(hw_params)); - - err = avc_general_set_sig_fmt(oxfw->unit, params_rate(hw_params), - AVC_GENERAL_PLUG_DIR_IN, 0); - if (err < 0) { - dev_err(&oxfw->unit->device, "failed to set sample rate\n"); - goto err_buffer; - } - return 0; - -err_buffer: - snd_pcm_lib_free_vmalloc_buffer(substream); -error: - mutex_unlock(&oxfw->mutex); - return err; + amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int pcm_hw_free(struct snd_pcm_substream *substream) @@ -215,19 +186,18 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) static int pcm_prepare(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; int err; mutex_lock(&oxfw->mutex); - - snd_oxfw_stream_stop_simplex(oxfw); - - err = snd_oxfw_stream_start_simplex(oxfw); + err = snd_oxfw_stream_start_simplex(oxfw, runtime->rate, + runtime->channels); + mutex_unlock(&oxfw->mutex); if (err < 0) goto end; amdtp_stream_pcm_prepare(&oxfw->rx_stream); end: - mutex_unlock(&oxfw->mutex); return err; } diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 210bf5a..1820497 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -7,8 +7,10 @@ */ #include "oxfw.h" +#include #define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512 +#define CALLBACK_TIMEOUT 200 /* * According to datasheet of Oxford Semiconductor: @@ -37,6 +39,47 @@ static const unsigned int avc_stream_rate_table[] = { [5] = 0x07, }; +static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, + unsigned int rate, unsigned int pcm_channels) +{ + u8 **formats; + struct snd_oxfw_stream_formation formation; + enum avc_general_plug_dir dir; + unsigned int i, err, len; + + formats = oxfw->rx_stream_formats; + dir = AVC_GENERAL_PLUG_DIR_IN; + + /* Seek stream format for requirements. */ + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + err = snd_oxfw_stream_parse_format(formats[i], &formation); + if (err < 0) + return err; + + if ((formation.rate == rate) && (formation.pcm == pcm_channels)) + break; + } + if (i == SND_OXFW_STREAM_FORMAT_ENTRIES) + return -EINVAL; + + /* If assumed, just change rate. */ + if (oxfw->assumed) + return avc_general_set_sig_fmt(oxfw->unit, rate, + AVC_GENERAL_PLUG_DIR_IN, 0); + + /* Calculate format length. */ + len = 5 + formats[i][4] * 2; + + err = avc_stream_set_format(oxfw->unit, dir, 0, formats[i], len); + if (err < 0) + return err; + + /* Some requests just after changing format causes freezing. */ + msleep(100); + + return 0; +} + int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw) { int err; @@ -63,30 +106,106 @@ static void stop_stream(struct snd_oxfw *oxfw) cmp_connection_break(&oxfw->in_conn); } -int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw) +static int start_stream(struct snd_oxfw *oxfw, unsigned int rate, + unsigned int pcm_channels) { - int err = 0; + u8 **formats; + struct cmp_connection *conn; + struct snd_oxfw_stream_formation formation; + unsigned int i, midi_ports; + struct amdtp_stream *stream; + int err; - if (amdtp_streaming_error(&oxfw->rx_stream)) - stop_stream(oxfw); + stream = &oxfw->rx_stream; + formats = oxfw->rx_stream_formats; + conn = &oxfw->in_conn; - if (amdtp_stream_running(&oxfw->rx_stream)) + /* Get stream formation */ + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + if (formats[i] == NULL) + break; + + err = snd_oxfw_stream_parse_format(formats[i], &formation); + if (err < 0) + goto end; + if (rate != formation.rate) + continue; + if (pcm_channels == 0 || pcm_channels == formation.pcm) + break; + } + if (i == SND_OXFW_STREAM_FORMAT_ENTRIES) { + err = -EINVAL; goto end; + } - err = cmp_connection_establish(&oxfw->in_conn, - amdtp_stream_get_max_payload(&oxfw->rx_stream)); + pcm_channels = formation.pcm; + midi_ports = DIV_ROUND_UP(formation.midi, 8); + + /* The stream should have one pcm channels at least */ + if (pcm_channels == 0) { + err = -EINVAL; + goto end; + } + amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports); + + err = cmp_connection_establish(conn, + amdtp_stream_get_max_payload(stream)); if (err < 0) goto end; - err = amdtp_stream_start(&oxfw->rx_stream, - oxfw->in_conn.resources.channel, - oxfw->in_conn.speed); + err = amdtp_stream_start(stream, + conn->resources.channel, + conn->speed); + if (err < 0) { + cmp_connection_break(conn); + goto end; + } + + /* Wait first packet */ + err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT); if (err < 0) stop_stream(oxfw); end: return err; } +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, unsigned int rate, + unsigned int pcm_channels) +{ + struct snd_oxfw_stream_formation formation; + int err = 0; + + /* packet queueing error */ + if (amdtp_streaming_error(&oxfw->rx_stream)) + stop_stream(oxfw); + + err = snd_oxfw_stream_get_current_formation(oxfw, + AVC_GENERAL_PLUG_DIR_IN, + &formation); + if (err < 0) + goto end; + + if ((formation.rate != rate) || (formation.pcm != pcm_channels)) { + stop_stream(oxfw); + + /* arrange sampling rate */ + err = set_stream_format(oxfw, &oxfw->rx_stream, rate, + pcm_channels); + if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to set stream format: %d\n", err); + goto end; + } + } + + err = start_stream(oxfw, rate, pcm_channels); + if (err < 0) + dev_err(&oxfw->unit->device, + "fail to start stream: %d\n", err); +end: + return err; +} + void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw) { stop_stream(oxfw); diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 8c832ea..c09ef38 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -89,7 +89,8 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, unsigned short pid); int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw); -int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw); +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, unsigned int rate, + unsigned int pcm_channels); void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw); void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); -- cgit v1.1 From ec4dba5053e1109368fb80d1c0b88f2a9c971122 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:45 +0900 Subject: ALSA: oxfw: Add support for Behringer/Mackie devices Some devices produced by Behringer/Mackie are based on OXFW970/971. This commit adds support for them. Additionally, this commit changes the way to name card with some information in config rom. Ids of some Mackie(Loud) models are not identified, therefore this commit applies name detection for these models. The devices support capture/playback of PCM-samples and some of them supports capture/playback of MIDI messages. These functionalities are implemented by followed commits. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 6 +++ sound/firewire/oxfw/oxfw.c | 91 +++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 91 insertions(+), 6 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 0932860..6364e5b 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -31,6 +31,12 @@ config SND_OXFW Oxford Semiconductor FW970/971 chipset. * Griffin Firewave * LaCie Firewire Speakers + * Behringer F-Control Audio 202 + * Mackie(Loud) Onyx-i series (former models) + * Mackie(Loud) Onyx Satellite + * Mackie(Loud) Tapco Link.Firewire + * Mackie(Loud) d.2 pro/d.4 pro + * Mackie(Loud) U.420/U.420d To compile this driver as a module, choose M here: the module will be called snd-oxfw. diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index a70149a..797af33 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -14,7 +14,9 @@ #define OXFORD_HARDWARE_ID_OXFW970 0x39443841 #define OXFORD_HARDWARE_ID_OXFW971 0x39373100 +#define VENDOR_LOUD 0x000ff2 #define VENDOR_GRIFFIN 0x001292 +#define VENDOR_BEHRINGER 0x001564 #define VENDOR_LACIE 0x00d04b #define SPECIFIER_1394TA 0x00a02d @@ -25,22 +27,69 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("snd-firewire-speakers"); +static bool detect_loud_models(struct fw_unit *unit) +{ + const char *const models[] = { + "Onyxi", + "Onyx-i", + "d.Pro", + "Mackie Onyx Satellite", + "Tapco LINK.firewire 4x6", + "U.420"}; + char model[32]; + unsigned int i; + int err; + + err = fw_csr_string(unit->directory, CSR_MODEL, + model, sizeof(model)); + if (err < 0) + return err; + + for (i = 0; i < ARRAY_SIZE(models); i++) { + if (strcmp(models[i], model) == 0) + break; + } + + return (i < ARRAY_SIZE(models)); +} + static int name_card(struct snd_oxfw *oxfw) { struct fw_device *fw_dev = fw_parent_device(oxfw->unit); + char vendor[24]; + char model[32]; const char *d, *v, *m; u32 firmware; int err; + /* get vendor name from root directory */ + err = fw_csr_string(fw_dev->config_rom + 5, CSR_VENDOR, + vendor, sizeof(vendor)); + if (err < 0) + goto end; + + /* get model name from unit directory */ + err = fw_csr_string(oxfw->unit->directory, CSR_MODEL, + model, sizeof(model)); + if (err < 0) + goto end; + err = snd_fw_transaction(oxfw->unit, TCODE_READ_QUADLET_REQUEST, OXFORD_FIRMWARE_ID_ADDRESS, &firmware, 4, 0); if (err < 0) goto end; be32_to_cpus(&firmware); - d = oxfw->device_info->driver_name; - v = oxfw->device_info->vendor_name; - m = oxfw->device_info->model_name; + /* to apply card definitions */ + if (oxfw->device_info) { + d = oxfw->device_info->driver_name; + v = oxfw->device_info->vendor_name; + m = oxfw->device_info->model_name; + } else { + d = "OXFW"; + v = vendor; + m = model; + } strcpy(oxfw->card->driver, d); strcpy(oxfw->card->mixername, m); @@ -73,6 +122,9 @@ static int oxfw_probe(struct fw_unit *unit, struct snd_oxfw *oxfw; int err; + if ((id->vendor_id == VENDOR_LOUD) && !detect_loud_models(unit)) + return -ENODEV; + err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, sizeof(*oxfw), &card); if (err < 0) @@ -97,9 +149,11 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_oxfw_create_mixer(oxfw); - if (err < 0) - goto error; + if (oxfw->device_info) { + err = snd_oxfw_create_mixer(oxfw); + if (err < 0) + goto error; + } snd_oxfw_proc_init(oxfw); @@ -183,6 +237,31 @@ static const struct ieee1394_device_id oxfw_id_table[] = { .version = VERSION_AVC, .driver_data = (kernel_ulong_t)&lacie_speakers, }, + /* Behringer,F-Control Audio 202 */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = VENDOR_BEHRINGER, + .model_id = 0x00fc22, + }, + /* + * Any Mackie(Loud) models (name string/model id): + * Onyx-i series (former models): 0x081216 + * Mackie Onyx Satellite: 0x00200f + * Tapco LINK.firewire 4x6: 0x000460 + * d.2 pro: Unknown + * d.4 pro: Unknown + * U.420: Unknown + * U.420d: Unknown + */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VENDOR_LOUD, + .specifier_id = SPECIFIER_1394TA, + .version = VERSION_AVC, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table); -- cgit v1.1 From b0ac00095fe1485f60bb8ea7326426d3d02a1aec Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:46 +0900 Subject: ALSA: oxfw: Add support AMDTP in-stream Previous commit adds support for some devices which can capture PCM samples. These devices transmit AMDTP stream in non-blocking mode. This commit adds functionality to handle AMDTP incoming stream. OXFW seems to have two quirks: - Transmits packets with non-zero dbc in its beginning - Transmits packets with wrong values in syt field For the first quirk, this commit adds CIP_SKIP_INIT_DBC_CHECK flag for incoming stream to skip first check of dbc. For the second quirk, this commit doesn't add duplex stream which Fireworks/BeBoB drivers use. So OXFW driver generates syt value for outgoing stream. Here are examples of a sequence of packets transmitted by Behringer F-Control Audio 202. There are differences between sequences of syt value when OXFW driver transfers outgoing stream or not. When driver gives no outgoing stream: Index Payload CIP_Header_0 CIP_Header_1 38 14 00020092 900103D1 39 12 00020098 900102FF 40 12 0002009D 9001027F 41 14 000200A2 90010396 42 14 000200A8 900102E8 43 12 000200AE 90010219 44 14 000200B3 90010331 45 12 000200B9 9001025F 46 14 000200BE 90010376 47 12 000200C4 900102A1 00 12 000200C9 9001023E 01 14 000200CE 90010358 02 12 000200D4 90010289 03 16 000200D9 900103A3 04 12 000200E0 900102DD 05 14 000200E5 900103F1 06 12 000200EB 90010335 07 12 000200F0 90010263 08 14 000200F5 9001037C 09 12 000200FB 900102AE When driver gives outgoing stream: Index Payload CIP_Header_0 CIP_Header_1 38 12 000200BD 900104A8 39 14 000200C2 900104A8 40 12 000200C8 900104AC 41 14 000200CD 900104A9 42 12 000200D3 900104B1 43 14 000200D8 900104A8 44 12 000200DE 900104AA 45 14 000200E3 900104A9 46 14 000200E9 900104AE 47 12 000200EF 900104A8 00 14 000200F4 900104AD 01 12 000200FA 900104A7 02 14 000200FF 900104A9 03 12 00020005 900104A9 04 14 0002000A 900104B1 05 12 00020010 900104AA 06 14 00020015 900104AD 07 12 0002001B 900104A7 08 14 00020020 900104AC 09 12 00020026 900104A7 Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/oxfw/oxfw-proc.c | 29 ++++ sound/firewire/oxfw/oxfw-stream.c | 270 +++++++++++++++++++++++++++++--------- sound/firewire/oxfw/oxfw.c | 25 +++- sound/firewire/oxfw/oxfw.h | 24 +++- 5 files changed, 279 insertions(+), 75 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index ea2b439..a78339c 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -177,7 +177,7 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) struct snd_oxfw *oxfw = substream->private_data; mutex_lock(&oxfw->mutex); - snd_oxfw_stream_stop_simplex(oxfw); + snd_oxfw_stream_stop_simplex(oxfw, &oxfw->rx_stream); mutex_unlock(&oxfw->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -190,8 +190,8 @@ static int pcm_prepare(struct snd_pcm_substream *substream) int err; mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_start_simplex(oxfw, runtime->rate, - runtime->channels); + err = snd_oxfw_stream_start_simplex(oxfw, &oxfw->rx_stream, + runtime->rate, runtime->channels); mutex_unlock(&oxfw->mutex); if (err < 0) goto end; diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 18e0305..604808e 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -44,6 +44,35 @@ static void proc_read_formation(struct snd_info_entry *entry, formation.rate, formation.pcm, formation.midi); } + if (!oxfw->has_output) + return; + + /* Show output. */ + err = snd_oxfw_stream_get_current_formation(oxfw, + AVC_GENERAL_PLUG_DIR_OUT, + &curr); + if (err < 0) + return; + + snd_iprintf(buffer, "Output Stream from device:\n"); + snd_iprintf(buffer, "\tRate\tPCM\tMIDI\n"); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + format = oxfw->tx_stream_formats[i]; + if (format == NULL) + continue; + + err = snd_oxfw_stream_parse_format(format, &formation); + if (err < 0) + continue; + + if (memcmp(&formation, &curr, sizeof(curr)) == 0) + flag = '*'; + else + flag = ' '; + + snd_iprintf(buffer, "%c\t%d\t%d\t%d\n", flag, + formation.rate, formation.pcm, formation.midi); + } } static void add_node(struct snd_oxfw *oxfw, struct snd_info_entry *root, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 1820497..1d15428 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -39,6 +39,22 @@ static const unsigned int avc_stream_rate_table[] = { [5] = 0x07, }; +static int set_rate(struct snd_oxfw *oxfw, unsigned int rate) +{ + int err; + + err = avc_general_set_sig_fmt(oxfw->unit, rate, + AVC_GENERAL_PLUG_DIR_IN, 0); + if (err < 0) + goto end; + + if (oxfw->has_output) + err = avc_general_set_sig_fmt(oxfw->unit, rate, + AVC_GENERAL_PLUG_DIR_OUT, 0); +end: + return err; +} + static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, unsigned int rate, unsigned int pcm_channels) { @@ -47,8 +63,13 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, enum avc_general_plug_dir dir; unsigned int i, err, len; - formats = oxfw->rx_stream_formats; - dir = AVC_GENERAL_PLUG_DIR_IN; + if (s == &oxfw->tx_stream) { + formats = oxfw->tx_stream_formats; + dir = AVC_GENERAL_PLUG_DIR_OUT; + } else { + formats = oxfw->rx_stream_formats; + dir = AVC_GENERAL_PLUG_DIR_IN; + } /* Seek stream format for requirements. */ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { @@ -64,8 +85,7 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, /* If assumed, just change rate. */ if (oxfw->assumed) - return avc_general_set_sig_fmt(oxfw->unit, rate, - AVC_GENERAL_PLUG_DIR_IN, 0); + return set_rate(oxfw, rate); /* Calculate format length. */ len = 5 + formats[i][4] * 2; @@ -80,47 +100,35 @@ static int set_stream_format(struct snd_oxfw *oxfw, struct amdtp_stream *s, return 0; } -int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw) +static void stop_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) { - int err; - - err = cmp_connection_init(&oxfw->in_conn, oxfw->unit, - CMP_INPUT, 0); - if (err < 0) - goto end; + amdtp_stream_pcm_abort(stream); + amdtp_stream_stop(stream); - err = amdtp_stream_init(&oxfw->rx_stream, oxfw->unit, - AMDTP_OUT_STREAM, CIP_NONBLOCKING); - if (err < 0) { - amdtp_stream_destroy(&oxfw->rx_stream); - cmp_connection_destroy(&oxfw->in_conn); - } -end: - return err; -} - -static void stop_stream(struct snd_oxfw *oxfw) -{ - amdtp_stream_pcm_abort(&oxfw->rx_stream); - amdtp_stream_stop(&oxfw->rx_stream); - cmp_connection_break(&oxfw->in_conn); + if (stream == &oxfw->tx_stream) + cmp_connection_break(&oxfw->out_conn); + else + cmp_connection_break(&oxfw->in_conn); } -static int start_stream(struct snd_oxfw *oxfw, unsigned int rate, - unsigned int pcm_channels) +static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream, + unsigned int rate, unsigned int pcm_channels) { u8 **formats; struct cmp_connection *conn; struct snd_oxfw_stream_formation formation; unsigned int i, midi_ports; - struct amdtp_stream *stream; int err; - stream = &oxfw->rx_stream; - formats = oxfw->rx_stream_formats; - conn = &oxfw->in_conn; + if (stream == &oxfw->rx_stream) { + formats = oxfw->rx_stream_formats; + conn = &oxfw->in_conn; + } else { + formats = oxfw->tx_stream_formats; + conn = &oxfw->out_conn; + } - /* Get stream formation */ + /* Get stream format */ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { if (formats[i] == NULL) break; @@ -164,67 +172,196 @@ static int start_stream(struct snd_oxfw *oxfw, unsigned int rate, /* Wait first packet */ err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT); if (err < 0) - stop_stream(oxfw); + stop_stream(oxfw, stream); end: return err; } -int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, unsigned int rate, - unsigned int pcm_channels) +static int check_connection_used_by_others(struct snd_oxfw *oxfw, + struct amdtp_stream *stream) { + struct cmp_connection *conn; + bool used; + int err; + + if (stream == &oxfw->tx_stream) + conn = &oxfw->out_conn; + else + conn = &oxfw->in_conn; + + err = cmp_connection_check_used(conn, &used); + if ((err >= 0) && used && !amdtp_stream_running(stream)) { + dev_err(&oxfw->unit->device, + "Connection established by others: %cPCR[%d]\n", + (conn->direction == CMP_OUTPUT) ? 'o' : 'i', + conn->pcr_index); + err = -EBUSY; + } + + return err; +} + +int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream) +{ + struct cmp_connection *conn; + enum cmp_direction c_dir; + enum amdtp_stream_direction s_dir; + int err; + + if (stream == &oxfw->tx_stream) { + conn = &oxfw->out_conn; + c_dir = CMP_OUTPUT; + s_dir = AMDTP_IN_STREAM; + } else { + conn = &oxfw->in_conn; + c_dir = CMP_INPUT; + s_dir = AMDTP_OUT_STREAM; + } + + err = cmp_connection_init(conn, oxfw->unit, c_dir, 0); + if (err < 0) + goto end; + + err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING); + if (err < 0) { + amdtp_stream_destroy(stream); + cmp_connection_destroy(conn); + goto end; + } + + /* OXFW starts to transmit packets with non-zero dbc. */ + if (stream == &oxfw->tx_stream) + oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; +end: + return err; +} + +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream, + unsigned int rate, unsigned int pcm_channels) +{ + struct amdtp_stream *opposite; struct snd_oxfw_stream_formation formation; + enum avc_general_plug_dir dir; + unsigned int substreams, opposite_substreams; int err = 0; + if (stream == &oxfw->tx_stream) { + substreams = oxfw->capture_substreams; + opposite = &oxfw->rx_stream; + opposite_substreams = oxfw->playback_substreams; + dir = AVC_GENERAL_PLUG_DIR_OUT; + } else { + substreams = oxfw->playback_substreams; + opposite_substreams = oxfw->capture_substreams; + + if (oxfw->has_output) + opposite = &oxfw->rx_stream; + else + opposite = NULL; + + dir = AVC_GENERAL_PLUG_DIR_IN; + } + + if (substreams == 0) + goto end; + + /* + * Considering JACK/FFADO streaming: + * TODO: This can be removed hwdep functionality becomes popular. + */ + err = check_connection_used_by_others(oxfw, stream); + if (err < 0) + goto end; + /* packet queueing error */ - if (amdtp_streaming_error(&oxfw->rx_stream)) - stop_stream(oxfw); + if (amdtp_streaming_error(stream)) + stop_stream(oxfw, stream); - err = snd_oxfw_stream_get_current_formation(oxfw, - AVC_GENERAL_PLUG_DIR_IN, - &formation); + err = snd_oxfw_stream_get_current_formation(oxfw, dir, &formation); if (err < 0) goto end; if ((formation.rate != rate) || (formation.pcm != pcm_channels)) { - stop_stream(oxfw); + if (opposite != NULL) { + err = check_connection_used_by_others(oxfw, opposite); + if (err < 0) + goto end; + stop_stream(oxfw, opposite); + } + stop_stream(oxfw, stream); - /* arrange sampling rate */ - err = set_stream_format(oxfw, &oxfw->rx_stream, rate, - pcm_channels); + err = set_stream_format(oxfw, stream, rate, pcm_channels); if (err < 0) { dev_err(&oxfw->unit->device, "fail to set stream format: %d\n", err); goto end; } + + /* Start opposite stream if needed. */ + if (opposite && !amdtp_stream_running(opposite) && + (opposite_substreams > 0)) { + err = start_stream(oxfw, opposite, rate, 0); + if (err < 0) { + dev_err(&oxfw->unit->device, + "fail to restart stream: %d\n", err); + goto end; + } + } } - err = start_stream(oxfw, rate, pcm_channels); - if (err < 0) - dev_err(&oxfw->unit->device, - "fail to start stream: %d\n", err); + /* Start requested stream. */ + if (!amdtp_stream_running(stream)) { + err = start_stream(oxfw, stream, rate, pcm_channels); + if (err < 0) + dev_err(&oxfw->unit->device, + "fail to start stream: %d\n", err); + } end: return err; } -void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw) +void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream) { - stop_stream(oxfw); + if (((stream == &oxfw->tx_stream) && (oxfw->capture_substreams > 0)) || + ((stream == &oxfw->rx_stream) && (oxfw->playback_substreams > 0))) + return; + + stop_stream(oxfw, stream); } -void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw) +void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream) { - stop_stream(oxfw); + struct cmp_connection *conn; + + if (stream == &oxfw->tx_stream) + conn = &oxfw->out_conn; + else + conn = &oxfw->in_conn; - amdtp_stream_destroy(&oxfw->rx_stream); - cmp_connection_destroy(&oxfw->in_conn); + stop_stream(oxfw, stream); + + amdtp_stream_destroy(stream); + cmp_connection_destroy(conn); } -void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw) +void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream) { - if (cmp_connection_update(&oxfw->in_conn) < 0) - stop_stream(oxfw); + struct cmp_connection *conn; + + if (stream == &oxfw->tx_stream) + conn = &oxfw->out_conn; + else + conn = &oxfw->in_conn; + + if (cmp_connection_update(conn) < 0) + stop_stream(oxfw, stream); else - amdtp_stream_update(&oxfw->rx_stream); + amdtp_stream_update(stream); } int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, @@ -408,7 +545,10 @@ static int fill_stream_formats(struct snd_oxfw *oxfw, if (buf == NULL) return -ENOMEM; - formats = oxfw->rx_stream_formats; + if (dir == AVC_GENERAL_PLUG_DIR_OUT) + formats = oxfw->tx_stream_formats; + else + formats = oxfw->rx_stream_formats; /* get first entry */ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; @@ -481,11 +621,19 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw) "fail to get info for isoc/external in/out plugs: %d\n", err); goto end; - } else if (plugs[0] == 0) { + } else if ((plugs[0] == 0) && (plugs[1] == 0)) { err = -ENOSYS; goto end; } + /* use oPCR[0] if exists */ + if (plugs[1] > 0) { + err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0); + if (err < 0) + goto end; + oxfw->has_output = true; + } + /* use iPCR[0] if exists */ if (plugs[0] > 0) err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0); diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 797af33..23c00a2 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -109,8 +109,10 @@ static void oxfw_card_free(struct snd_card *card) struct snd_oxfw *oxfw = card->private_data; unsigned int i; - for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + kfree(oxfw->tx_stream_formats[i]); kfree(oxfw->rx_stream_formats[i]); + } mutex_destroy(&oxfw->mutex); } @@ -157,13 +159,20 @@ static int oxfw_probe(struct fw_unit *unit, snd_oxfw_proc_init(oxfw); - err = snd_oxfw_stream_init_simplex(oxfw); + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; + if (oxfw->has_output) { + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); + if (err < 0) + goto error; + } err = snd_card_register(card); if (err < 0) { - snd_oxfw_stream_destroy_simplex(oxfw); + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); goto error; } dev_set_drvdata(&unit->device, oxfw); @@ -181,7 +190,11 @@ static void oxfw_bus_reset(struct fw_unit *unit) fcp_bus_reset(oxfw->unit); mutex_lock(&oxfw->mutex); - snd_oxfw_stream_update_simplex(oxfw); + + snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); + mutex_unlock(&oxfw->mutex); } @@ -191,7 +204,9 @@ static void oxfw_remove(struct fw_unit *unit) snd_card_disconnect(oxfw->card); - snd_oxfw_stream_destroy_simplex(oxfw); + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); snd_card_free_when_closed(oxfw->card); } diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index c09ef38..2211d11 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -44,10 +44,16 @@ struct snd_oxfw { const struct device_info *device_info; struct mutex mutex; + bool has_output; + u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; bool assumed; + struct cmp_connection out_conn; struct cmp_connection in_conn; + struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; + unsigned int capture_substreams; + unsigned int playback_substreams; bool mute; s16 volume[6]; @@ -88,12 +94,17 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, enum avc_general_plug_dir dir, unsigned short pid); -int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw); -int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, unsigned int rate, - unsigned int pcm_channels); -void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw); -void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw); -void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw); +int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream); +int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream, + unsigned int rate, unsigned int pcm_channels); +void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream); +void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream); +void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw, + struct amdtp_stream *stream); struct snd_oxfw_stream_formation { unsigned int rate; @@ -105,6 +116,7 @@ int snd_oxfw_stream_parse_format(u8 *format, int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, struct snd_oxfw_stream_formation *formation); + int snd_oxfw_stream_discover(struct snd_oxfw *oxfw); int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); -- cgit v1.1 From 216e256f7bf974ba402309d0ceb24f3500dc65c4 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:47 +0900 Subject: ALSA: oxfw: add support for capturing PCM samples In previous commit, a support for transmitted packets is added. This commit add a support for capturing PCM samples. When any streams are already started, this driver should not change sampling rate of the device, thus this commit also adds a restriction of sampling rate in this situation. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/oxfw-pcm.c | 200 ++++++++++++++++++++++++++++++++++++----- 1 file changed, 180 insertions(+), 20 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index a78339c..e84fc9c 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -118,21 +118,31 @@ static void limit_period_and_buffer(struct snd_pcm_hardware *hw) hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min; } -static int pcm_open(struct snd_pcm_substream *substream) +static int init_hw_params(struct snd_oxfw *oxfw, + struct snd_pcm_substream *substream) { - struct snd_oxfw *oxfw = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; u8 **formats; + struct amdtp_stream *stream; int err; - formats = oxfw->rx_stream_formats; - runtime->hw.info = SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS; + stream = &oxfw->tx_stream; + formats = oxfw->tx_stream_formats; + } else { + runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS; + stream = &oxfw->rx_stream; + formats = oxfw->rx_stream_formats; + } + limit_channels_and_rates(&runtime->hw, formats); limit_period_and_buffer(&runtime->hw); @@ -148,10 +158,55 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto end; - err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime); + err = amdtp_stream_add_pcm_hw_constraints(stream, runtime); +end: + return err; +} + +static int limit_to_current_params(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_oxfw_stream_formation formation; + enum avc_general_plug_dir dir; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = AVC_GENERAL_PLUG_DIR_OUT; + else + dir = AVC_GENERAL_PLUG_DIR_IN; + + err = snd_oxfw_stream_get_current_formation(oxfw, dir, &formation); + if (err < 0) + goto end; + + substream->runtime->hw.channels_min = formation.pcm; + substream->runtime->hw.channels_max = formation.pcm; + substream->runtime->hw.rate_min = formation.rate; + substream->runtime->hw.rate_max = formation.rate; +end: + return err; +} + +static int pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + int err; + + err = init_hw_params(oxfw, substream); if (err < 0) goto end; + /* + * When any PCM streams are already running, the available sampling + * rate is limited at current value. + */ + if (amdtp_stream_pcm_running(&oxfw->tx_stream) || + amdtp_stream_pcm_running(&oxfw->rx_stream)) { + err = limit_to_current_params(substream); + if (err < 0) + goto end; + } + snd_pcm_set_sync(substream); end: return err; @@ -162,28 +217,89 @@ static int pcm_close(struct snd_pcm_substream *substream) return 0; } -static int pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) +static int pcm_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&oxfw->mutex); + oxfw->capture_substreams++; + mutex_unlock(&oxfw->mutex); + } + + amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); + + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} +static int pcm_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_oxfw *oxfw = substream->private_data; + + if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { + mutex_lock(&oxfw->mutex); + oxfw->playback_substreams++; + mutex_unlock(&oxfw->mutex); + } + amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); } -static int pcm_hw_free(struct snd_pcm_substream *substream) +static int pcm_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + + mutex_lock(&oxfw->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + oxfw->capture_substreams--; + + snd_oxfw_stream_stop_simplex(oxfw, &oxfw->tx_stream); + + mutex_unlock(&oxfw->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} +static int pcm_playback_hw_free(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; mutex_lock(&oxfw->mutex); + + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + oxfw->playback_substreams--; + snd_oxfw_stream_stop_simplex(oxfw, &oxfw->rx_stream); + mutex_unlock(&oxfw->mutex); return snd_pcm_lib_free_vmalloc_buffer(substream); } -static int pcm_prepare(struct snd_pcm_substream *substream) +static int pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&oxfw->mutex); + err = snd_oxfw_stream_start_simplex(oxfw, &oxfw->tx_stream, + runtime->rate, runtime->channels); + mutex_unlock(&oxfw->mutex); + if (err < 0) + goto end; + + amdtp_stream_pcm_prepare(&oxfw->tx_stream); +end: + return err; +} +static int pcm_playback_prepare(struct snd_pcm_substream *substream) { struct snd_oxfw *oxfw = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -201,7 +317,25 @@ end: return err; } -static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_oxfw *oxfw = substream->private_data; + struct snd_pcm_substream *pcm; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pcm = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pcm = NULL; + break; + default: + return -EINVAL; + } + amdtp_stream_pcm_trigger(&oxfw->tx_stream, pcm); + return 0; +} +static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_oxfw *oxfw = substream->private_data; struct snd_pcm_substream *pcm; @@ -220,35 +354,61 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstm) { - struct snd_oxfw *oxfw = substream->private_data; + struct snd_oxfw *oxfw = sbstm->private_data; + + return amdtp_stream_pcm_pointer(&oxfw->tx_stream); +} +static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstm) +{ + struct snd_oxfw *oxfw = sbstm->private_data; return amdtp_stream_pcm_pointer(&oxfw->rx_stream); } int snd_oxfw_create_pcm(struct snd_oxfw *oxfw) { - static struct snd_pcm_ops ops = { + static struct snd_pcm_ops capture_ops = { + .open = pcm_open, + .close = pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = pcm_capture_hw_params, + .hw_free = pcm_capture_hw_free, + .prepare = pcm_capture_prepare, + .trigger = pcm_capture_trigger, + .pointer = pcm_capture_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + static struct snd_pcm_ops playback_ops = { .open = pcm_open, .close = pcm_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = pcm_hw_params, - .hw_free = pcm_hw_free, - .prepare = pcm_prepare, - .trigger = pcm_trigger, - .pointer = pcm_pointer, + .hw_params = pcm_playback_hw_params, + .hw_free = pcm_playback_hw_free, + .prepare = pcm_playback_prepare, + .trigger = pcm_playback_trigger, + .pointer = pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, .mmap = snd_pcm_lib_mmap_vmalloc, }; struct snd_pcm *pcm; + unsigned int cap = 0; int err; - err = snd_pcm_new(oxfw->card, oxfw->card->driver, 0, 1, 0, &pcm); + if (oxfw->has_output) + cap = 1; + + err = snd_pcm_new(oxfw->card, oxfw->card->driver, 0, 1, cap, &pcm); if (err < 0) return err; + pcm->private_data = oxfw; strcpy(pcm->name, oxfw->card->shortname); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops); + if (cap > 0) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops); + return 0; } -- cgit v1.1 From 05588d340a128ff5c7b768c517150e31842a78aa Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:48 +0900 Subject: ALSA: oxfw: Add support for capture/playback MIDI messages This commit adds MIDI functionality with an assumption of 'if the device has MIDI comformant data channels in its stream formation, the device has one MIDI port'. When no streams have already started, MIDI functionality starts stream with current sampling rate. When MIDI functionality has already starts some streams and PCM functionality is going to start streams at different sampling rate, this driver stops streams once and changes sampling rate, then restarts streams for both PCM/MIDI substreams. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-midi.c | 191 ++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw-stream.c | 4 + sound/firewire/oxfw/oxfw.c | 5 + sound/firewire/oxfw/oxfw.h | 7 ++ 5 files changed, 208 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/oxfw/oxfw-midi.c diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index e9297c6..3904a30 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ - oxfw-proc.o oxfw.o + oxfw-proc.o oxfw-midi.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c new file mode 100644 index 0000000..334b11d --- /dev/null +++ b/sound/firewire/oxfw/oxfw-midi.c @@ -0,0 +1,191 @@ +/* + * oxfw_midi.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "oxfw.h" + +static int midi_capture_open(struct snd_rawmidi_substream *substream) +{ + struct snd_oxfw *oxfw = substream->rmidi->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + oxfw->capture_substreams++; + err = snd_oxfw_stream_start_simplex(oxfw, &oxfw->tx_stream, 0, 0); + + mutex_unlock(&oxfw->mutex); + + return err; +} + +static int midi_playback_open(struct snd_rawmidi_substream *substream) +{ + struct snd_oxfw *oxfw = substream->rmidi->private_data; + int err; + + mutex_lock(&oxfw->mutex); + + oxfw->playback_substreams++; + err = snd_oxfw_stream_start_simplex(oxfw, &oxfw->rx_stream, 0, 0); + + mutex_unlock(&oxfw->mutex); + + return err; +} + +static int midi_capture_close(struct snd_rawmidi_substream *substream) +{ + struct snd_oxfw *oxfw = substream->rmidi->private_data; + + mutex_lock(&oxfw->mutex); + + oxfw->capture_substreams--; + snd_oxfw_stream_stop_simplex(oxfw, &oxfw->tx_stream); + + mutex_unlock(&oxfw->mutex); + + return 0; +} + +static int midi_playback_close(struct snd_rawmidi_substream *substream) +{ + struct snd_oxfw *oxfw = substream->rmidi->private_data; + + mutex_lock(&oxfw->mutex); + + oxfw->playback_substreams--; + snd_oxfw_stream_stop_simplex(oxfw, &oxfw->rx_stream); + + mutex_unlock(&oxfw->mutex); + + return 0; +} + +static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_oxfw *oxfw = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&oxfw->lock, flags); + + if (up) + amdtp_stream_midi_trigger(&oxfw->tx_stream, + substrm->number, substrm); + else + amdtp_stream_midi_trigger(&oxfw->tx_stream, + substrm->number, NULL); + + spin_unlock_irqrestore(&oxfw->lock, flags); +} + +static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up) +{ + struct snd_oxfw *oxfw = substrm->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&oxfw->lock, flags); + + if (up) + amdtp_stream_midi_trigger(&oxfw->rx_stream, + substrm->number, substrm); + else + amdtp_stream_midi_trigger(&oxfw->rx_stream, + substrm->number, NULL); + + spin_unlock_irqrestore(&oxfw->lock, flags); +} + +static struct snd_rawmidi_ops midi_capture_ops = { + .open = midi_capture_open, + .close = midi_capture_close, + .trigger = midi_capture_trigger, +}; + +static struct snd_rawmidi_ops midi_playback_ops = { + .open = midi_playback_open, + .close = midi_playback_close, + .trigger = midi_playback_trigger, +}; + +static void set_midi_substream_names(struct snd_oxfw *oxfw, + struct snd_rawmidi_str *str) +{ + struct snd_rawmidi_substream *subs; + + list_for_each_entry(subs, &str->substreams, list) { + snprintf(subs->name, sizeof(subs->name), + "%s MIDI %d", + oxfw->card->shortname, subs->number + 1); + } +} + +int snd_oxfw_create_midi(struct snd_oxfw *oxfw) +{ + struct snd_oxfw_stream_formation formation; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_str *str; + u8 *format; + int i, err; + + /* If its stream has MIDI conformant data channel, add one MIDI port */ + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) { + format = oxfw->tx_stream_formats[i]; + if (format != NULL) { + err = snd_oxfw_stream_parse_format(format, &formation); + if (err >= 0 && formation.midi > 0) + oxfw->midi_input_ports = 1; + } + + format = oxfw->rx_stream_formats[i]; + if (format != NULL) { + err = snd_oxfw_stream_parse_format(format, &formation); + if (err >= 0 && formation.midi > 0) + oxfw->midi_output_ports = 1; + } + } + if ((oxfw->midi_input_ports == 0) && (oxfw->midi_output_ports == 0)) + return 0; + + /* create midi ports */ + err = snd_rawmidi_new(oxfw->card, oxfw->card->driver, 0, + oxfw->midi_output_ports, oxfw->midi_input_ports, + &rmidi); + if (err < 0) + return err; + + snprintf(rmidi->name, sizeof(rmidi->name), + "%s MIDI", oxfw->card->shortname); + rmidi->private_data = oxfw; + + if (oxfw->midi_input_ports > 0) { + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &midi_capture_ops); + + str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]; + + set_midi_substream_names(oxfw, str); + } + + if (oxfw->midi_output_ports > 0) { + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &midi_playback_ops); + + str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]; + + set_midi_substream_names(oxfw, str); + } + + if ((oxfw->midi_output_ports > 0) && (oxfw->midi_input_ports > 0)) + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; + + return 0; +} diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 1d15428..a38b3c3 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -282,6 +282,10 @@ int snd_oxfw_stream_start_simplex(struct snd_oxfw *oxfw, err = snd_oxfw_stream_get_current_formation(oxfw, dir, &formation); if (err < 0) goto end; + if (rate == 0) + rate = formation.rate; + if (pcm_channels == 0) + pcm_channels = formation.pcm; if ((formation.rate != rate) || (formation.pcm != pcm_channels)) { if (opposite != NULL) { diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 23c00a2..9cfbfb1 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -138,6 +138,7 @@ static int oxfw_probe(struct fw_unit *unit, mutex_init(&oxfw->mutex); oxfw->unit = unit; oxfw->device_info = (const struct device_info *)id->driver_data; + spin_lock_init(&oxfw->lock); err = snd_oxfw_stream_discover(oxfw); if (err < 0) @@ -159,6 +160,10 @@ static int oxfw_probe(struct fw_unit *unit, snd_oxfw_proc_init(oxfw); + err = snd_oxfw_create_midi(oxfw); + if (err < 0) + goto error; + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 2211d11..83a54fc 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -19,6 +19,7 @@ #include #include #include +#include #include "../lib.h" #include "../fcp.h" @@ -43,6 +44,7 @@ struct snd_oxfw { struct fw_unit *unit; const struct device_info *device_info; struct mutex mutex; + spinlock_t lock; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; @@ -55,6 +57,9 @@ struct snd_oxfw { unsigned int capture_substreams; unsigned int playback_substreams; + unsigned int midi_input_ports; + unsigned int midi_output_ports; + bool mute; s16 volume[6]; s16 volume_min; @@ -124,3 +129,5 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); void snd_oxfw_proc_init(struct snd_oxfw *oxfw); + +int snd_oxfw_create_midi(struct snd_oxfw *oxfw); -- cgit v1.1 From 8985f4ac1c42bd25799f294f4e87fa73064673c7 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 9 Dec 2014 00:10:49 +0900 Subject: ALSA: oxfw: Add hwdep interface This interface is designed for mixer/control application. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 1 + sound/firewire/oxfw/Makefile | 2 +- sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw-midi.c | 16 ++++ sound/firewire/oxfw/oxfw-pcm.c | 12 ++- sound/firewire/oxfw/oxfw-stream.c | 39 ++++++++ sound/firewire/oxfw/oxfw.c | 5 + sound/firewire/oxfw/oxfw.h | 13 +++ 10 files changed, 280 insertions(+), 4 deletions(-) create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 941d32f..1f23cd6 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -96,9 +96,10 @@ enum { SNDRV_HWDEP_IFACE_FW_DICE, /* TC DICE FireWire device */ SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ + SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_BEBOB + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index af4bd13..49122df 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -55,7 +55,8 @@ union snd_firewire_event { #define SNDRV_FIREWIRE_TYPE_DICE 1 #define SNDRV_FIREWIRE_TYPE_FIREWORKS 2 #define SNDRV_FIREWIRE_TYPE_BEBOB 3 -/* AV/C, RME, MOTU, ... */ +#define SNDRV_FIREWIRE_TYPE_OXFW 4 +/* RME, MOTU, ... */ struct snd_firewire_get_info { unsigned int type; /* SNDRV_FIREWIRE_TYPE_xxx */ diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 6364e5b..ecec547 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -26,6 +26,7 @@ config SND_DICE config SND_OXFW tristate "Oxford Semiconductor FW970/971 chipset support" select SND_FIREWIRE_LIB + select SND_HWDEP help Say Y here to include support for FireWire devices based on Oxford Semiconductor FW970/971 chipset. diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index 3904a30..a926850 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,3 +1,3 @@ snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \ - oxfw-proc.o oxfw-midi.o oxfw.o + oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o obj-m += snd-oxfw.o diff --git a/sound/firewire/oxfw/oxfw-hwdep.c b/sound/firewire/oxfw/oxfw-hwdep.c new file mode 100644 index 0000000..ff2687a --- /dev/null +++ b/sound/firewire/oxfw/oxfw-hwdep.c @@ -0,0 +1,190 @@ +/* + * oxfw_hwdep.c - a part of driver for OXFW970/971 based devices + * + * Copyright (c) 2014 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +/* + * This codes give three functionality. + * + * 1.get firewire node information + * 2.get notification about starting/stopping stream + * 3.lock/unlock stream + */ + +#include "oxfw.h" + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct snd_oxfw *oxfw = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&oxfw->lock); + + while (!oxfw->dev_lock_changed) { + prepare_to_wait(&oxfw->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&oxfw->lock); + schedule(); + finish_wait(&oxfw->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&oxfw->lock); + } + + memset(&event, 0, sizeof(event)); + if (oxfw->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (oxfw->dev_lock_count > 0); + oxfw->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + } + + spin_unlock_irq(&oxfw->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_oxfw *oxfw = hwdep->private_data; + unsigned int events; + + poll_wait(file, &oxfw->hwdep_wait, wait); + + spin_lock_irq(&oxfw->lock); + if (oxfw->dev_lock_changed) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&oxfw->lock); + + return events; +} + +static int hwdep_get_info(struct snd_oxfw *oxfw, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(oxfw->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_OXFW; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_oxfw *oxfw) +{ + int err; + + spin_lock_irq(&oxfw->lock); + + if (oxfw->dev_lock_count == 0) { + oxfw->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&oxfw->lock); + + return err; +} + +static int hwdep_unlock(struct snd_oxfw *oxfw) +{ + int err; + + spin_lock_irq(&oxfw->lock); + + if (oxfw->dev_lock_count == -1) { + oxfw->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&oxfw->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_oxfw *oxfw = hwdep->private_data; + + spin_lock_irq(&oxfw->lock); + if (oxfw->dev_lock_count == -1) + oxfw->dev_lock_count = 0; + spin_unlock_irq(&oxfw->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_oxfw *oxfw = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(oxfw, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(oxfw); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(oxfw); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +int snd_oxfw_create_hwdep(struct snd_oxfw *oxfw) +{ + static const struct snd_hwdep_ops hwdep_ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, + }; + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(oxfw->card, oxfw->card->driver, 0, &hwdep); + if (err < 0) + goto end; + strcpy(hwdep->name, oxfw->card->driver); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_OXFW; + hwdep->ops = hwdep_ops; + hwdep->private_data = oxfw; + hwdep->exclusive = true; +end: + return err; +} diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c index 334b11d..540a303 100644 --- a/sound/firewire/oxfw/oxfw-midi.c +++ b/sound/firewire/oxfw/oxfw-midi.c @@ -13,6 +13,10 @@ static int midi_capture_open(struct snd_rawmidi_substream *substream) struct snd_oxfw *oxfw = substream->rmidi->private_data; int err; + err = snd_oxfw_stream_lock_try(oxfw); + if (err < 0) + return err; + mutex_lock(&oxfw->mutex); oxfw->capture_substreams++; @@ -20,6 +24,9 @@ static int midi_capture_open(struct snd_rawmidi_substream *substream) mutex_unlock(&oxfw->mutex); + if (err < 0) + snd_oxfw_stream_lock_release(oxfw); + return err; } @@ -28,6 +35,10 @@ static int midi_playback_open(struct snd_rawmidi_substream *substream) struct snd_oxfw *oxfw = substream->rmidi->private_data; int err; + err = snd_oxfw_stream_lock_try(oxfw); + if (err < 0) + return err; + mutex_lock(&oxfw->mutex); oxfw->playback_substreams++; @@ -35,6 +46,9 @@ static int midi_playback_open(struct snd_rawmidi_substream *substream) mutex_unlock(&oxfw->mutex); + if (err < 0) + snd_oxfw_stream_lock_release(oxfw); + return err; } @@ -49,6 +63,7 @@ static int midi_capture_close(struct snd_rawmidi_substream *substream) mutex_unlock(&oxfw->mutex); + snd_oxfw_stream_lock_release(oxfw); return 0; } @@ -63,6 +78,7 @@ static int midi_playback_close(struct snd_rawmidi_substream *substream) mutex_unlock(&oxfw->mutex); + snd_oxfw_stream_lock_release(oxfw); return 0; } diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index e84fc9c..9bc556b 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -192,10 +192,14 @@ static int pcm_open(struct snd_pcm_substream *substream) struct snd_oxfw *oxfw = substream->private_data; int err; - err = init_hw_params(oxfw, substream); + err = snd_oxfw_stream_lock_try(oxfw); if (err < 0) goto end; + err = init_hw_params(oxfw, substream); + if (err < 0) + goto err_locked; + /* * When any PCM streams are already running, the available sampling * rate is limited at current value. @@ -210,10 +214,16 @@ static int pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); end: return err; +err_locked: + snd_oxfw_stream_lock_release(oxfw); + return err; } static int pcm_close(struct snd_pcm_substream *substream) { + struct snd_oxfw *oxfw = substream->private_data; + + snd_oxfw_stream_lock_release(oxfw); return 0; } diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index a38b3c3..b77cf80 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -644,3 +644,42 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw) end: return err; } + +void snd_oxfw_stream_lock_changed(struct snd_oxfw *oxfw) +{ + oxfw->dev_lock_changed = true; + wake_up(&oxfw->hwdep_wait); +} + +int snd_oxfw_stream_lock_try(struct snd_oxfw *oxfw) +{ + int err; + + spin_lock_irq(&oxfw->lock); + + /* user land lock this */ + if (oxfw->dev_lock_count < 0) { + err = -EBUSY; + goto end; + } + + /* this is the first time */ + if (oxfw->dev_lock_count++ == 0) + snd_oxfw_stream_lock_changed(oxfw); + err = 0; +end: + spin_unlock_irq(&oxfw->lock); + return err; +} + +void snd_oxfw_stream_lock_release(struct snd_oxfw *oxfw) +{ + spin_lock_irq(&oxfw->lock); + + if (WARN_ON(oxfw->dev_lock_count <= 0)) + goto end; + if (--oxfw->dev_lock_count == 0) + snd_oxfw_stream_lock_changed(oxfw); +end: + spin_unlock_irq(&oxfw->lock); +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 9cfbfb1..cf1d0b5 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -139,6 +139,7 @@ static int oxfw_probe(struct fw_unit *unit, oxfw->unit = unit; oxfw->device_info = (const struct device_info *)id->driver_data; spin_lock_init(&oxfw->lock); + init_waitqueue_head(&oxfw->hwdep_wait); err = snd_oxfw_stream_discover(oxfw); if (err < 0) @@ -164,6 +165,10 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; + err = snd_oxfw_create_hwdep(oxfw); + if (err < 0) + goto error; + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 83a54fc..cace5ad 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -12,6 +12,7 @@ #include #include #include +#include #include #include @@ -20,6 +21,8 @@ #include #include #include +#include +#include #include "../lib.h" #include "../fcp.h" @@ -64,6 +67,10 @@ struct snd_oxfw { s16 volume[6]; s16 volume_min; s16 volume_max; + + int dev_lock_count; + bool dev_lock_changed; + wait_queue_head_t hwdep_wait; }; /* @@ -124,6 +131,10 @@ int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw, int snd_oxfw_stream_discover(struct snd_oxfw *oxfw); +void snd_oxfw_stream_lock_changed(struct snd_oxfw *oxfw); +int snd_oxfw_stream_lock_try(struct snd_oxfw *oxfw); +void snd_oxfw_stream_lock_release(struct snd_oxfw *oxfw); + int snd_oxfw_create_pcm(struct snd_oxfw *oxfw); int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); @@ -131,3 +142,5 @@ int snd_oxfw_create_mixer(struct snd_oxfw *oxfw); void snd_oxfw_proc_init(struct snd_oxfw *oxfw); int snd_oxfw_create_midi(struct snd_oxfw *oxfw); + +int snd_oxfw_create_hwdep(struct snd_oxfw *oxfw); -- cgit v1.1 From f62f5eff3d40a56ad1cf0d81a6cac8dd8743e8a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 9 Dec 2014 19:58:53 +0100 Subject: ALSA: hda - Add EAPD fixup for ASUS Z99He laptop The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A codec like another ASUS machine. Reported-and-tested-by: Dmitry V. Zimin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..4714ff9 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), -- cgit v1.1 From 9faa73f06ec2408cf86d20758d879b8d928ab30a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Dec 2014 13:58:37 +0100 Subject: ALSA: hda - Add "eapd" model string for AD1986A codec Also update the documentation to the latest state. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 8 ++------ sound/pci/hda/patch_analog.c | 1 + 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a5e7547..5a3163c 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -113,14 +113,10 @@ AD1984 AD1986A ======= - 6stack 6-jack, separate surrounds (default) 3stack 3-stack, shared surrounds laptop 2-channel only (FSC V2060, Samsung M50) - laptop-eapd 2-channel with EAPD (ASUS A6J) - laptop-automute 2-channel with EAPD and HP-automute (Lenovo N100) - ultra 2-channel with EAPD (Samsung Ultra tablet PC) - samsung 2-channel with EAPD (Samsung R65) - samsung-p50 2-channel with HP-automute (Samsung P50) + laptop-imic 2-channel with built-in mic + eapd Turn on EAPD constantly AD1988/AD1988B/AD1989A/AD1989B ============================== diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4714ff9..c81b715 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -352,6 +352,7 @@ static const struct hda_model_fixup ad1986a_fixup_models[] = { { .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" }, { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" }, { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */ + { .id = AD1986A_FIXUP_EAPD, .name = "eapd" }, {} }; -- cgit v1.1 From feabb67e0eb0f509b530a56756ea44364843faea Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 10 Dec 2014 16:06:46 +0300 Subject: ALSA: lola: NULL dereference on probe failure "card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index a75c8dc..4cf4be5 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -719,7 +719,7 @@ static int lola_probe(struct pci_dev *pci, err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - dev_err(card->dev, "Error creating card!\n"); + dev_err(&pci->dev, "Error creating card!\n"); return err; } -- cgit v1.1 From 6e1d7a51392f06899bd7b693f28ac60fa1e00032 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 10 Dec 2014 16:26:21 +0300 Subject: ALSA: pcxhr: NULL dereference on probe failure "card" is NULL if snd_card_new() fails. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index a602930..c6092e4 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1638,7 +1638,7 @@ static int pcxhr_probe(struct pci_dev *pci, 0, &card); if (err < 0) { - dev_err(card->dev, "cannot allocate the card %d\n", i); + dev_err(&pci->dev, "cannot allocate the card %d\n", i); pcxhr_free(mgr); return err; } -- cgit v1.1