December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Resend: Incorrect sbits for snd-usb-audio
by Sander Jansen 08 Dec '13

08 Dec '13

Hi, I had send this out earlier back in September, but never got a response. I have a Focusrite Scarlett 2i4 USB Audio Interface. The audio format exposed by ALSA for this device is S32_LE. From the hardware specs and lsusb. it supposed to only have a 24 bit resolution (with the LSB bits set to 0): AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 When I query the sbits for this device, I expect it to return 24, but instead it returns 32. Looking through the usb audio driver code, it doesn't look like this information is actually exposed to userspace. The 'parse_audio_format_i_type' in usb/format.c merely returns the supported formats, but doesn't include any msbits info. (and it doesn't look like it can be easily changed to return the msbits info as well). Was this intentionally left out, or is this a missing feature in the driver? Thanks, Sander

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Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by Raymond Yau 08 Dec '13

08 Dec '13

> I just changed my RadeonHD 4670 for a 7870, and now when I type > "aplay -l" I get many HDMI codecs instead of just one like for the > previous you have to specify CONFIG_SND_DYNAMIC_MINORS=y when building the driver [ 4.637413] ALSA hda_codec.c:4506 Too many HDMI devices [ 4.637414] ALSA hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y [ 4.637644] ALSA control.c:352 control 0:0:0:HDMI/DP Jack:0 is already present [ 4.639220] hda_codec: cannot build controls for #0 (error -16) > can you post the system log if the graphic driver know hdmi, vga and two >> mini display ports are connected or not ? >> >> I cannot see anything in dmesg related, so attaching Xorg log. > >> - >> > 5.268590] [drm] Radeon Display Connectors [ 5.268592] [drm] Connector 0: [ 5.268593] [drm] DP-1 [ 5.268594] [drm] HPD4 [ 5.268595] [drm] DDC: 0x6530 0x6530 0x6534 0x6534 0x6538 0x6538 0x653c 0x653c [ 5.268596] [drm] Encoders: [ 5.268597] [drm] DFP1: INTERNAL_UNIPHY2 [ 5.268598] [drm] Connector 1: [ 5.268599] [drm] DP-2 [ 5.268600] [drm] HPD5 [ 5.268601] [drm] DDC: 0x6540 0x6540 0x6544 0x6544 0x6548 0x6548 0x654c 0x654c [ 5.268602] [drm] Encoders: [ 5.268603] [drm] DFP2: INTERNAL_UNIPHY2 [ 5.268603] [drm] Connector 2: [ 5.268604] [drm] HDMI-A-1 [ 5.268605] [drm] HPD1 [ 5.268606] [drm] DDC: 0x6550 0x6550 0x6554 0x6554 0x6558 0x6558 0x655c 0x655c [ 5.268607] [drm] Encoders: [ 5.268608] [drm] DFP3: INTERNAL_UNIPHY1 [ 5.268609] [drm] Connector 3: [ 5.268610] [drm] DVI-I-1 [ 5.268610] [drm] HPD6 [ 5.268612] [drm] DDC: 0x6580 0x6580 0x6584 0x6584 0x6588 0x6588 0x658c 0x658c [ 5.268612] [drm] Encoders: [ 5.268613] [drm] DFP4: INTERNAL_UNIPHY [ 5.268614] [drm] CRT1: INTERNAL_KLDSCP_DAC1 [ 7.613] (II) RADEON(0): Output DisplayPort-0 has no monitor section [ 7.626] (II) RADEON(0): Output DisplayPort-1 has no monitor section [ 7.693] (II) RADEON(0): Output HDMI-0 has no monitor section [ 7.713] (II) RADEON(0): Output DVI-0 has no monitor section [ 7.726] (II) RADEON(0): EDID for output DisplayPort-0 [ 7.740] (II) RADEON(0): EDID for output DisplayPort-1 [ 7.801] (II) RADEON(0): EDID for output HDMI-0 [ 7.801] (II) RADEON(0): Manufacturer: DON Model: 26 Serial#: 16843009 [ 7.801] (II) RADEON(0): Year: 2011 Week: 0 [ 7.801] (II) RADEON(0): EDID Version: 1.3 [ 7.823] (II) RADEON(0): EDID for output DVI-0 [ 7.823] (II) RADEON(0): Output DisplayPort-0 disconnected [ 7.823] (II) RADEON(0): Output DisplayPort-1 disconnected [ 7.823] (II) RADEON(0): Output HDMI-0 connected [ 7.823] (II) RADEON(0): Output DVI-0 disconnected the graphic driver knew only HDMI-0 is connected and the others are not connected

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Re: [alsa-devel] [pulseaudio-discuss] Issue with multiple hdmi codecs in radeonhd, hacked to work
by John Ettedgui 08 Dec '13

08 Dec '13

and same logs but with -with-debug=detect instead of full.

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: apply only when period size is greater than 64. sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..af86c71f27bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,15 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + if (runtime->period_size > 64) + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + else + azx_dev->delay_negative_threshold = 0; + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2458,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH] ALSA: hda - Ignore small negative LPIB delay correction
by Takashi Iwai 06 Dec '13

06 Dec '13

Sometimes the hardware reports LPIB being advanced than POSBUF. When this happens, the driver adjusts to a positive value by adding the buffer size. Then the driver detects it as an error (greater than the period size), and stops the LPIB delay account from this point on. When I took a close look at these conditions, the values shown are all very small numbers, and it'd be better to just ignore these values instead of discontinuing the LPIB delay correction. In this patch, the driver checks a negative delay value and ignores if it's a significantly small error. Currently the threshold is set to 64 frames, but could be smaller. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 80c5f14e8ecd..fd4f2288a33f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -431,6 +431,8 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + int delay_negative_threshold; + #ifdef CONFIG_SND_HDA_DSP_LOADER struct mutex dsp_mutex; #endif @@ -2195,6 +2197,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + /* when LPIB delay correction gives a small negative value, + * we ignore it; currently set the threshold statically to + * 64 frames + */ + azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); + /* wallclk has 24Mhz clock source */ azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); @@ -2447,8 +2455,12 @@ static unsigned int azx_get_position(struct azx *chip, delay = pos - lpib_pos; else delay = lpib_pos - pos; - if (delay < 0) - delay += azx_dev->bufsize; + if (delay < 0) { + if (delay >= azx_dev->delay_negative_threshold) + delay = 0; + else + delay += azx_dev->bufsize; + } if (delay >= azx_dev->period_bytes) { snd_printk(KERN_WARNING SFX "%s: Unstable LPIB (%d >= %d); " -- 1.8.5

1 0

[alsa-devel] [PATCH 0/3] Enhance imx-wm8962 machine driver
by Nicolin Chen 06 Dec '13

06 Dec '13

Although there's no direct relationship between any of these three patches, I recommend to apply them in order so as to circumvent merging conflicts. Nicolin Chen (3): ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver ASoC: fsl: imx-wm8962: Grant hw_params/free() permission to control FLL ASoC: fsl: imx-wm8962: Add headphone and microphone jack plugin/out detection .../devicetree/bindings/sound/imx-audio-wm8962.txt | 8 + Documentation/devicetree/bindings/sound/wm8962.txt | 8 + sound/soc/fsl/imx-wm8962.c | 361 ++++++++++++++++++--- 3 files changed, 326 insertions(+), 51 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH 1/2] ASoC: samsung: Provide helper for DMA init
by Mark Brown 06 Dec '13

06 Dec '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 7 +++++-- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 42 insertions(+), 45 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 350ba23a9893..4a88e36c82ec 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b494d4..ee23194f7ab8 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315a3eaf..fb09a1c5f75b 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index a5cbdb4f1655..eab0050d4579 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -946,8 +946,11 @@ 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; - if (other && other->clk) /* If this is probe on secondary */ + if (other && other->clk) { /* If this is probe on secondary */ + samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, + NULL); goto probe_exit; + } i2s->addr = ioremap(i2s->base, 0x100); if (i2s->addr == NULL) { @@ -963,7 +966,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256fc4b2c..6a5e4bf6ac96 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.5.1

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[alsa-devel] [PATCH] ASoC mc13783 codec: correct names
by Lambrecht Jürgen 06 Dec '13

06 Dec '13

Correct names to reality and for consistency: - "Speaker Amp Switch"->"Speaker Amp" - "Earpiece Amp Switch"->"Speaker Amp Switch" - "Loudspeaker enable"->"Loudspeaker Amp Switch" - "..Capture.."->"..Playback" (Line in..., CODEC..., ...Mixin Switch) - move "PCM Capture Volume" to bottom with other Capture controls Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/mc13783.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 08fb022..878e3e5 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -534,7 +534,7 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, &cdcout_ctl), - SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, @@ -590,7 +590,7 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "SP", "Speaker Amp", "Speaker Amp Source MUX"}, { "LSP", "Loudspeaker Amp", "Loudspeaker Amp Source MUX"}, { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, @@ -609,26 +609,27 @@ static const struct soc_enum mc13783_enum_3d_mixer = mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { - SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), - SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), - SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_SINGLE("Loudspeaker Amp Switch", MC13783_AUDIO_RX0, 5, 1, 0), SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), - SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), - SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), - SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + SOC_SINGLE("CODEC Playback Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + SOC_SINGLE("PCM Playback Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Playback Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Playback Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Playback Switch", MC13783_AUDIO_RX1, 0, 1, 0), - SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), - SOC_SINGLE("CODEC Capture Switch", MC13783_AUDIO_RX1, 0, 1, 0), + SOC_SINGLE("Line in Playback Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Playback Switch", MC13783_AUDIO_RX1, 10, 1, 0), - SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), - SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), -- 1.7.10.4

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Re: [alsa-devel] Intermittent cut out on the video
by Dolevo Jay 06 Dec '13

06 Dec '13

From: cmst(a)live.com To: clemens(a)ladisch.de Subject: RE: [alsa-devel] Intermittent cut out on the video Date: Mon, 2 Dec 2013 09:44:16 +0000 > Date: Thu, 28 Nov 2013 09:29:45 +0100 > From: clemens(a)ladisch.de > To: cmst(a)live.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Intermittent cut out on the video > > Dolevo Jay wrote: > > I have two sound cards on my system and I use the [multi plugin] to > > have audio from two jacks simultaneously. > > > > The audio coming from the jack on the pc is cutting out intermittently. > > It just cuts out for a fraction of a second and continues as normal > > and after some time it happens again. > > The two cards use different sample clocks, so it is not possible to feed > them reliably from the same stream. > > You would need software that measures that actual sample rate of both > devices and resamples its output appropriately. Try using PulseAudio. > > > Regards, > Clemens > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Hi, I have tried to apply pulseaudio but couldn't really manage it. Could you please tell me if I can fix it in the configuration file? I mean, for example, would it be solved if I resample the audio for the appropriate sound card in the asound.conf? What other option do I have other than pulseaudio? Thanks a lot.

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[alsa-devel] [PATCH] ALSA: hda - Allow capture-only configuration
by Takashi Iwai 06 Dec '13

06 Dec '13

We have blindly assumed that all valid configurations should have either analog or digital playback, but there can be capture-only configurations. The parser shouldn't escape in such a case. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2729e881541d..5ba3106b9712 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4334,7 +4334,8 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, spec->no_analog = 1; goto dig_only; } - return 0; /* can't find valid BIOS pin config */ + if (!cfg->num_inputs && !cfg->dig_in_pin) + return 0; /* can't find valid BIOS pin config */ } if (!spec->no_primary_hp && -- 1.8.5

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