diff options
Diffstat (limited to 'sound')
38 files changed, 252 insertions, 99 deletions
diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c index e6beb92..b4590df 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; } diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb7..75a0d74 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = diff --git a/sound/core/control.c b/sound/core/control.c index 6d71f9a..b0bf426 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -225,8 +225,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.id.iface = ncontrol->iface; kctl.id.device = ncontrol->device; kctl.id.subdevice = ncontrol->subdevice; - if (ncontrol->name) + if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); + if (strcmp(ncontrol->name, kctl.id.name) != 0) + snd_printk(KERN_WARNING + "Control name '%s' truncated to '%s'\n", + ncontrol->name, kctl.id.name); + } kctl.id.index = ncontrol->index; kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1af62b8..e178366 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2283,7 +2283,7 @@ static int snd_pcm_oss_open_file(struct file *file, int idx, err; struct snd_pcm_oss_file *pcm_oss_file; struct snd_pcm_substream *substream; - unsigned int f_mode = file->f_mode; + fmode_t f_mode = file->f_mode; if (rpcm_oss_file) *rpcm_oss_file = NULL; diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e341f3f..1f42e40 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -34,7 +34,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->thalf = 0; if (!atomic_read(&chip->timer_active)) return HRTIMER_NORESTART; - hrtimer_forward(&chip->timer, chip->timer.expires, + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), ktime_set(0, chip->ns_rem)); return HRTIMER_RESTART; } @@ -118,7 +118,8 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) chip->ns_rem = PCSP_PERIOD_NS(); ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); chip->ns_rem -= ns; - hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); + hrtimer_forward(&chip->timer, hrtimer_get_expires(&chip->timer), + ktime_set(0, ns)); return HRTIMER_RESTART; exit_nr_unlock2: diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 23018a7..81e1f44 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -93,7 +93,7 @@ static struct au1550_state { spinlock_t lock; struct mutex open_mutex; struct mutex sem; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; struct dmabuf { diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index d978b00..1cb13fe 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -129,7 +129,7 @@ typedef struct { int (*mixer_ioctl)(u_int, u_long); /* optional */ int (*write_sq_setup)(void); /* optional */ int (*read_sq_setup)(void); /* optional */ - int (*sq_open)(mode_t); /* optional */ + int (*sq_open)(fmode_t); /* optional */ int (*state_info)(char *, size_t); /* optional */ void (*abort_read)(void); /* optional */ int min_dsp_speed; @@ -235,7 +235,7 @@ struct sound_queue { */ int active; wait_queue_head_t action_queue, open_queue, sync_queue; - int open_mode; + fmode_t open_mode; int busy, syncing, xruns, died; }; diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 285239d..4d45bd6 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -143,7 +143,7 @@ static int AtaMixerIoctl(u_int cmd, u_long arg); static int TTMixerIoctl(u_int cmd, u_long arg); static int FalconMixerIoctl(u_int cmd, u_long arg); static int AtaWriteSqSetup(void); -static int AtaSqOpen(mode_t mode); +static int AtaSqOpen(fmode_t mode); static int TTStateInfo(char *buffer, size_t space); static int FalconStateInfo(char *buffer, size_t space); @@ -1461,7 +1461,7 @@ static int AtaWriteSqSetup(void) return 0 ; } -static int AtaSqOpen(mode_t mode) +static int AtaSqOpen(fmode_t mode) { write_sq_ignore_int = 1; return 0 ; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 95fc5c6..b8239f3 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -212,7 +212,7 @@ static int irq_installed; #endif /* MODULE */ /* control over who can modify resources shared between play/record */ -static mode_t shared_resource_owner; +static fmode_t shared_resource_owner; static int shared_resources_initialised; /* @@ -668,7 +668,7 @@ static inline void sq_init_waitqueue(struct sound_queue *sq) #if 0 /* blocking open() */ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, - mode_t mode) + fmode_t mode) { if (file->f_mode & mode) { sq->busy = 0; /* CHECK: IS THIS OK??? */ @@ -677,7 +677,7 @@ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, } #endif -static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, +static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, int numbufs, int bufsize) { int rc = 0; @@ -891,10 +891,10 @@ static int sq_release(struct inode *inode, struct file *file) is the owner - if we have problems. */ -static int shared_resources_are_mine(mode_t md) +static int shared_resources_are_mine(fmode_t md) { if (shared_resource_owner) - return (shared_resource_owner & md ) ; + return (shared_resource_owner & md) != 0; else { shared_resource_owner = md ; return 1 ; diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index eb9bc36..c180598 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -1,7 +1,7 @@ /* * Initialisation code for Cyrix/NatSemi VSA1 softaudio * - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index 61b3955..c8be47e 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -211,7 +211,7 @@ typedef struct multisound_dev { /* State variables */ enum { msndClassic, msndPinnacle } type; - mode_t mode; + fmode_t mode; unsigned long flags; #define F_RESETTING 0 #define F_HAVEDIGITAL 1 diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 1a00a32..55271fb 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -110,24 +110,16 @@ struct channel_info { #define OPEN_WRITE PCM_ENABLE_OUTPUT #define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) -#if OPEN_READ == FMODE_READ && OPEN_WRITE == FMODE_WRITE - -static inline int translate_mode(struct file *file) -{ - return file->f_mode; -} - -#else - static inline int translate_mode(struct file *file) { - return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | - ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); + if (OPEN_READ == (__force int)FMODE_READ && + OPEN_WRITE == (__force int)FMODE_WRITE) + return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE)); + else + return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | + ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); } -#endif - - #include "sound_calls.h" #include "dev_table.h" diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 044453a..41562ec 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -295,7 +295,7 @@ struct cs4297a_state { struct mutex open_mutex; struct mutex open_sem_adc; struct mutex open_sem_dac; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; wait_queue_head_t open_wait_adc; wait_queue_head_t open_wait_dac; diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index dcbb3f7..78b8acc 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -1509,7 +1509,7 @@ typedef struct vwsnd_dev { struct mutex open_mutex; struct mutex io_mutex; struct mutex mix_mutex; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; lithium_t lith; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acb..bd510ec 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device) ac97->dev.bus = &ac97_bus_type; ac97->dev.parent = ac97->bus->card->dev; ac97->dev.release = ac97_device_release; - snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - ac97->bus->card->number, ac97->num, - snd_ac97_get_short_name(ac97)); + dev_set_name(&ac97->dev, "%d-%d:%s", + ac97->bus->card->number, ac97->num, + snd_ac97_get_short_name(ac97)); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); ac97->dev.bus = NULL; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a97..a7f38e6 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b..226fe82 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f10630..0e6e5cc 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e4..a36d4d1 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d97..1aa1c04 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f..192e784 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68..6dea5b5 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -2,7 +2,7 @@ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio * * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net> - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did * mess with it a bit. The chip seems to have to have trouble with full duplex @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6447754..ba1ab73 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -64,6 +64,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, { 0x434d, "C-Media" }, { 0x8384, "SigmaTel" }, {} /* terminator */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f316c1..35722ec 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -45,6 +45,7 @@ #include <linux/slab.h> #include <linux/pci.h> #include <linux/mutex.h> +#include <linux/reboot.h> #include <sound/core.h> #include <sound/initval.h> #include "hda_codec.h" @@ -397,6 +398,9 @@ struct azx { /* for pending irqs */ struct work_struct irq_pending_work; + + /* reboot notifier (for mysterious hangup problem at power-down) */ + struct notifier_block reboot_notifier; }; /* driver types */ @@ -1979,12 +1983,36 @@ static int azx_resume(struct pci_dev *pci) /* + * reboot notifier for hang-up problem at power-down + */ +static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void azx_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = azx_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void azx_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* * destructor */ static int azx_free(struct azx *chip) { int i; + azx_notifier_unregister(chip); + if (chip->initialized) { azx_clear_irq_pending(chip); for (i = 0; i < chip->num_streams; i++) @@ -2158,7 +2186,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; @@ -2348,6 +2376,7 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); chip->running = 1; power_down_all_codecs(chip); + azx_notifier_register(chip); dev++; return err; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2b00c4a..d3fd432 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3860,6 +3860,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e72707c..4eceab9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -307,6 +307,13 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; + +#ifdef SND_HDA_NEEDS_RESUME +#define ALC_MAX_PINS 16 + unsigned int num_pins; + hda_nid_t pin_nids[ALC_MAX_PINS]; + unsigned int pin_cfgs[ALC_MAX_PINS]; +#endif }; /* @@ -2778,6 +2785,64 @@ static void alc_free(struct hda_codec *codec) codec->spec = NULL; /* to be sure */ } +#ifdef SND_HDA_NEEDS_RESUME +static void store_pin_configs(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid, end_nid; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = + (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wid_type != AC_WID_PIN) + continue; + if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) + break; + spec->pin_nids[spec->num_pins] = nid; + spec->pin_cfgs[spec->num_pins] = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + spec->num_pins++; + } +} + +static void resume_pin_configs(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) { + hda_nid_t pin_nid = spec->pin_nids[i]; + unsigned int pin_config = spec->pin_cfgs[i]; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + } +} + +static int alc_resume(struct hda_codec *codec) +{ + resume_pin_configs(codec); + codec->patch_ops.init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + return 0; +} +#else +#define store_pin_configs(codec) +#endif + /* */ static struct hda_codec_ops alc_patch_ops = { @@ -2786,6 +2851,9 @@ static struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = alc_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = alc_resume, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = alc_check_power_status, #endif @@ -3832,6 +3900,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + store_pin_configs(codec); return 1; } @@ -4996,7 +5065,7 @@ static struct hda_verb alc260_test_init_verbs[] = { */ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, - const char *pfx) + const char *pfx, int *vol_bits) { hda_nid_t nid_vol; unsigned long vol_val, sw_val; @@ -5018,10 +5087,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, } else return 0; /* N/A */ - snprintf(name, sizeof(name), "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); - if (err < 0) - return err; + if (!(*vol_bits & (1 << nid_vol))) { + /* first control for the volume widget */ + snprintf(name, sizeof(name), "%s Playback Volume", pfx); + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); + if (err < 0) + return err; + *vol_bits |= (1 << nid_vol); + } snprintf(name, sizeof(name), "%s Playback Switch", pfx); err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val); if (err < 0) @@ -5035,6 +5108,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, { hda_nid_t nid; int err; + int vols = 0; spec->multiout.num_dacs = 1; spec->multiout.dac_nids = spec->private_dac_nids; @@ -5042,21 +5116,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Front"); + err = alc260_add_playback_controls(spec, nid, "Front", &vols); if (err < 0) return err; } nid = cfg->speaker_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Speaker"); + err = alc260_add_playback_controls(spec, nid, "Speaker", &vols); if (err < 0) return err; } nid = cfg->hp_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Headphone"); + err = alc260_add_playback_controls(spec, nid, "Headphone", + &vols); if (err < 0) return err; } @@ -5244,6 +5319,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) } spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -10307,6 +10383,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -11441,6 +11518,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -12224,6 +12302,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc269_capture_mixer; spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -13310,6 +13389,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc861_capture_mixer; spec->num_mixers++; + store_pin_configs(codec); return 1; } @@ -14421,6 +14501,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + store_pin_configs(codec); return 1; } @@ -16252,6 +16333,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; + + store_pin_configs(codec); return 1; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2ac720..df9b0bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -566,10 +566,8 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, nid = codec->slave_dig_outs[smux_idx - 1]; if (spec->cur_smux[smux_idx] == smux->num_items - 1) val = AMP_OUT_MUTE; - if (smux_idx == 0) - nid = spec->multiout.dig_out_nid; else - nid = codec->slave_dig_outs[smux_idx - 1]; + val = AMP_OUT_UNMUTE; /* un/mute SPDIF out */ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); @@ -1282,7 +1280,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; spec->multiout.share_spdif = 1; } - if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) { + if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); if (err < 0) return err; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5b44238..58d7cda 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2688,12 +2688,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } err = snd_ice1712_ac97_mixer(ice); if (err < 0) { @@ -2715,12 +2716,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c88d1ea..19d3391 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), + SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000), SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000), { } /* terminator */ }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce6..ae7601f 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 827587f..e020c16 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = { } }; -static u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, -}; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) @@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 05336ed..cff276e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba402b..945b32e 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -140,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), SOC_ENUM("ALC Function", wm9713_enum[6]), SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 8692329..94a02ea 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -277,7 +277,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; u16 imr; u8 psc_cmd; - long flags; + unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_i2s->capture; @@ -699,9 +699,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev, return count; } -DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); -DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -819,8 +821,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Register the SYSFS files */ rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33a..8485a8a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 462e635d..a3adbf0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1462,7 +1462,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - unsigned int shift = mc->min; + unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; if (max == 1) diff --git a/sound/sound_core.c b/sound/sound_core.c index faef87a..a75b289 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -57,7 +57,7 @@ module_exit(cleanup_soundcore); /* * OSS sound core handling. Breaks out sound functions to submodules * - * Author: Alan Cox <alan.cox@linux.org> + * Author: Alan Cox <alan@lxorguk.ukuu.org.uk> * * Fixes: * diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 69689e7..9211575 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1480,6 +1480,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Edirol UA-25EX. + * For the standard mode, UA-25EX has ID 0582:00e7, which + * offers only 16-bit PCM at 44.1 kHz and no MIDI. + */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-25EX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { |
