From e3b1e6a19e09877b91517dfe304a2b3f6b2138fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Dec 2014 11:46:38 +0000 Subject: ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe This reverts commit f8781db8aeb18d (ASoC: dapm: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing) since it is broken for deferred probing as it ends up storing data allocated with devm_ over multiple instantiations of the device. Reported-by: Russell King Tested-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9357210..d759546 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3231,7 +3231,7 @@ 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, old_routes; + int num_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -3249,9 +3249,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - old_routes = card->num_dapm_routes; - routes = devm_kzalloc(card->dev, - (old_routes + num_routes) * sizeof(*routes), + routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -3259,11 +3257,9 @@ 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[old_routes + i].sink); + 2 * i, &routes[i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3271,7 +3267,7 @@ 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[old_routes + i].source); + (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -3280,7 +3276,7 @@ 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 da042e3a6261a444868b99bece98e4e5b77a0fce Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 18 Dec 2014 16:40:35 +0200 Subject: ASoC: Intel: Add I2C dependency to two new machines MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix following build error when CONFIG_I2C is not enabled: sound/soc/codecs/rt5640.c:2252:1: warning: data definition has no type or storage class module_i2c_driver(rt5640_i2c_driver); ^ sound/soc/codecs/rt5640.c:2252:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2252:1: warning: parameter names (without types) in function declaration sound/soc/codecs/rt5640.c:2241:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] static struct i2c_driver rt5640_i2c_driver = { ^ cc1: some warnings being treated as errors Signed-off-by: Jarkko Nikula Reported-by: Randy Dunlap Acked-by: Randy Dunlap Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf..f86de12 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -89,7 +89,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 + depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -101,7 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- cgit v1.1 From 3475c3d034d7f276a474c8bd53f44b48c8bf669d Mon Sep 17 00:00:00 2001 From: Andrew Jackson Date: Fri, 19 Dec 2014 16:18:05 +0000 Subject: ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap Flush the FIFOs when the stream is prepared for use. This avoids an inadvertent swapping of the left/right channels if the FIFOs are not empty at startup. Signed-off-by: Andrew Jackson Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/dwc/designware_i2s.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index b93168d..4df19b5 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -263,6 +263,19 @@ static void dw_i2s_shutdown(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, NULL); } +static int dw_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, TXFFR, 1); + else + i2s_write_reg(dev->i2s_base, RXFFR, 1); + + return 0; +} + static int dw_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -294,6 +307,7 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, + .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, }; -- cgit v1.1 From db2c1f9e38a38e0f9294b5a8a83d744f68fbe726 Mon Sep 17 00:00:00 2001 From: Andrew Jackson Date: Fri, 19 Dec 2014 16:18:06 +0000 Subject: ASoC: dwc: Iterate over all channels The Designware core can be configured with up to four stereo channels. Each stereo channel is individually configured so, when the driver's hw_params call is made, each requested stereo channel has to be programmed. Signed-off-by: Andrew Jackson Signed-off-by: Mark Brown --- sound/soc/dwc/designware_i2s.c | 35 ++++++++++++++++------------------- 1 file changed, 16 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 4df19b5..8d18bbd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -209,16 +209,9 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: - ch_reg = 3; - break; case SIX_CHANNEL_SUPPORT: - ch_reg = 2; - break; case FOUR_CHANNEL_SUPPORT: - ch_reg = 1; - break; case TWO_CHANNEL_SUPPORT: - ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); @@ -227,18 +220,22 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + for (ch_reg = 0; ch_reg < (config->chan_nr / 2); ch_reg++) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } } i2s_write_reg(dev->i2s_base, CCR, ccr); -- cgit v1.1 From 64b9c90b86002fb33ddc5583c5be165128835913 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 22 Dec 2014 16:49:19 +0200 Subject: ASoC: Intel: Fix BYTCR firmware name BYTCR DSP firmware is in intel/ subdirectory. See linux-firmware.git commit d562a3b63632 ("linux-firmware: add sst audio firmware for baytrail platforms"). Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3abc29e..2ac72eb 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,7 +343,7 @@ int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; -- cgit v1.1 From c6905d62269cdf66e179c579c1bd4b1b8baa02fe Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 22 Dec 2014 16:49:20 +0200 Subject: ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS snd_soc_sst_bytcr_dpcm_rt5640 doesn't autoload because MODULE_ALIAS doesn't match with "bytt100_rt5640" platform device. Signed-off-by: Jarkko Nikula Cc: Subhransu S. Prusty Signed-off-by: Mark Brown --- sound/soc/intel/bytcr_dpcm_rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index f5d0fc1..eef0c56 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -227,4 +227,4 @@ 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"); +MODULE_ALIAS("platform:bytt100_rt5640"); -- cgit v1.1 From 6087fcab7bd5122e7264504854ec77d5be0286ff Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Tue, 23 Dec 2014 23:49:05 -0400 Subject: ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic snd_kcontrol_chip should return snd_soc_component instead of snd_soc_codec Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..c0fbe18 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -784,8 +784,8 @@ static unsigned int bst_tlv[] = { 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); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; @@ -795,8 +795,9 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, 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); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; -- cgit v1.1 From 25f97549b5a19a373922e07c3e0f0b0b56a49148 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 23 Dec 2014 09:12:45 +0800 Subject: ASoC: Intel: correct the fixed free block allocation For block span more than 1 section, when allocate it from a free block, we need allocate the remain buffers within the block, and then continue alloc the rest of needed size buffer. Here also make sure this free block is moved from free list to used list, and add it to block_list which may be used for power gating disabling later. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- sound/soc/intel/sst-firmware.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 4a5bde9..ef2e8b5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -763,8 +763,12 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba /* does block span more than 1 section */ if (ba->offset >= block->offset && ba->offset < block_end) { + /* add block */ + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->offset = block->offset; + ba->size -= block_end - ba->offset; + ba->offset = block_end; err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) -- cgit v1.1 From 4db9c4a9b27f2b3c7df0d75b16078322447dc87a Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Wed, 24 Dec 2014 17:37:00 +0800 Subject: ASoC: rockchip: i2s: fix error defination of transmit data level According to description about "Transmit Data Level", This bit field controls the level at which a DMA request is made by the transmit logic. It is equal to the watermark level. That is, the dma_tx_req signal is generated when the number of valid data entries in the TXFIFO (TXFIFO0 if CSR=00 TXFIFO1 if CSR=01 TXFIFO2 if CSR=10 TXFIFO3 if CSR=11) is equal to or below this field value. Different to receive data level, transmit data level does not need to "-1". Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index 89a5d8b..93f456f 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -127,7 +127,7 @@ #define I2S_DMACR_TDE_DISABLE (0 << I2S_DMACR_TDE_SHIFT) #define I2S_DMACR_TDE_ENABLE (1 << I2S_DMACR_TDE_SHIFT) #define I2S_DMACR_TDL_SHIFT 0 -#define I2S_DMACR_TDL(x) ((x - 1) << I2S_DMACR_TDL_SHIFT) +#define I2S_DMACR_TDL(x) ((x) << I2S_DMACR_TDL_SHIFT) #define I2S_DMACR_TDL_MASK (0x1f << I2S_DMACR_TDL_SHIFT) /* -- cgit v1.1 From 27fd36ab135724ed9c648c57e5c0d0299bd7f67a Mon Sep 17 00:00:00 2001 From: Jianqun Xu Date: Wed, 24 Dec 2014 17:37:02 +0800 Subject: ASoC: rockchip: i2s: fix maxburst of dma data to 4 Since RK3288 DMAC's burst length only support max to 4, here set maxburst of playback and capture dma data to 4. Signed-off-by: Jianqun Xu Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 26ec511..13d8507 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -454,11 +454,11 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->playback_dma_data.addr = res->start + I2S_TXDR; i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->playback_dma_data.maxburst = 16; + i2s->playback_dma_data.maxburst = 4; i2s->capture_dma_data.addr = res->start + I2S_RXDR; i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - i2s->capture_dma_data.maxburst = 16; + i2s->capture_dma_data.maxburst = 4; i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); -- cgit v1.1 From 93e3423e6ba4b0ddaf056ecbdf5bc46f18f41deb Mon Sep 17 00:00:00 2001 From: Rafal Redzimski Date: Fri, 19 Dec 2014 08:44:30 +0800 Subject: ALSA: hda_controller: Separate stream_tag for input and output streams. Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: Rafal Redzimski Signed-off-by: Jayachandran B Signed-off-by: Libin Yang Reviewed-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8276a74..0cfc9c8 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ 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 */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- cgit v1.1 From d6795827bd79b28fef1abdaf7e525fcca506b831 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 19 Dec 2014 08:44:31 +0800 Subject: ALSA: hda_intel: apply the Seperate stream_tag for Skylake The total stream number of Skylake's input and output stream exceeds 15, which will cause some streams do not work because of the overflow on SDxCTL.STRM field if using the legacy stream tag allocation method. This patch uses the new stream tag allocation method by add the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform. Signed-off-by: Libin Yang Reviewed-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2bf0b56..d426a0b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -299,6 +299,9 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_SKYLAKE \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ @@ -2027,7 +2030,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- cgit v1.1 From 49cdd5b641933fda6324fc901eaf856924ba6a27 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 4 Jan 2015 19:59:29 +0100 Subject: ALSA: snd-usb-caiaq: fix stream count check Commit 897c329bc ("ALSA: usb: caiaq: check for cdev->n_streams > 1") introduced a safety check to protect against bogus data provided by devices. However, the n_streams variable is already divided by CHANNELS_PER_STREAM, so the correct check is 'n_streams > 0'. Fix this to un-break support for stereo devices. Signed-off-by: Daniel Mack Cc: stable@kernel.org [v3.18+] Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2728447..327f864 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -816,7 +816,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev) return -EINVAL; } - if (cdev->n_streams < 2) { + if (cdev->n_streams < 1) { dev_err(dev, "bogus number of streams: %d\n", cdev->n_streams); return -EINVAL; } -- cgit v1.1 From c507de88f6a336bd7296c9ec0073b2d4af8b4f5e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Jan 2015 13:27:33 +0100 Subject: ALSA: hda - Fix wrong gpio_dir & gpio_mask hint setups for IDT/STAC codecs stac_store_hints() does utterly wrong for masking the values for gpio_dir and gpio_data, likely due to copy&paste errors. Fortunately, this feature is used very rarely, so the impact must be really small. Reported-by: Rasmus Villemoes Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4f6413e..605d140 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -568,9 +568,9 @@ static void stac_store_hints(struct hda_codec *codec) spec->gpio_mask; } if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_data &= spec->gpio_mask; if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) spec->eapd_mask &= spec->gpio_mask; if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) -- cgit v1.1 From 60834b73a9c2bbc2f514122ddc626f3350fb40cd Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Tue, 6 Jan 2015 13:40:14 -0800 Subject: ALSA: hda - Add new GPU codec ID 0x10de0072 to snd-hda Vendor ID 0x10de0072 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5f13d2d..b422e40 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3353,6 +3353,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, { .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3413,6 +3414,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); +MODULE_ALIAS("snd-hda-codec-id:10de0072"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); MODULE_ALIAS("snd-hda-codec-id:11069f81"); -- cgit v1.1 From 92cb46584e104e2f4b14a44959109ffe13524a26 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 8 Jan 2015 00:31:16 +0900 Subject: ALSA: fireworks: fix an endianness bug for transaction length Although the 't->length' is a big-endian value, it's used without any conversion. This means that the driver always uses 'length' parameter. Fixes: 555e8a8f7f14("ALSA: fireworks: Add command/response functionality into hwdep interface") Reported-by: Clemens Ladisch Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks_transaction.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 255dabc..2a85e42 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -124,7 +124,7 @@ copy_resp_to_buf(struct snd_efw *efw, void *data, size_t length, int *rcode) spin_lock_irq(&efw->lock); t = (struct snd_efw_transaction *)data; - length = min_t(size_t, t->length * sizeof(t->length), length); + length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length); if (efw->push_ptr < efw->pull_ptr) capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr); -- cgit v1.1