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

Re: [alsa-devel] Broken headphone and speaker amplifier suppliesforthe rt5640 driver
by Bard Liao 29 Jul '13

29 Jul '13

> -----Original Message----- > From: alsa-devel-bounces(a)alsa-project.org > [mailto:alsa-devel-bounces@alsa-project.org] On Behalf Of Stephen Warren > Sent: Saturday, July 27, 2013 12:58 AM > To: Lars-Peter Clausen > Cc: Bard Liao; Linux-ALSA > Subject: Re: [alsa-devel] Broken headphone and speaker amplifier suppliesfor > the rt5640 driver > > On 07/26/2013 09:06 AM, Lars-Peter Clausen wrote: > > Hi, > > > > The rt5640 driver sets for both the headphone and speaker amplifier > > widget the shift to SND_SOC_NOPM, which is -1. This doesn't make much > > sense and breaks compilation with some changes I've made. > > > > I guess for the heaphone amplifier the right fix is to set the shift > > to RT5640_PWR_HA_BIT ... > > If I make those changes (see below for the exact patch I tried), then both > headphones and speakers do appear to still work fine. Bard will have to > comment on whether the changes are actually correct though:-) The control bits for headphone amplifier and speaker amplifier are both correct. But rt5640 need a sequence for headphone depop. So, I would like to do something like this. - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), I will send a patch ASAP. Thanks. > > > diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c > > index ce585e3..e3ab0b2 100644 > > --- a/sound/soc/codecs/rt5640.c > > +++ b/sound/soc/codecs/rt5640.c > > @@ -899,8 +899,6 @@ static int spk_event(struct snd_soc_dapm_widget > > *w, > > > > switch (event) { > > case SND_SOC_DAPM_POST_PMU: > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0001); > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0xf000); > > break; > > @@ -908,8 +906,6 @@ static int spk_event(struct snd_soc_dapm_widget > *w, > > case SND_SOC_DAPM_PRE_PMD: > > regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, > > 0xf000, 0x0000); > > - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, > > - 0x0001, 0x0000); > > break; > > > > default: > > @@ -1159,13 +1155,13 @@ static const struct snd_soc_dapm_widget > rt5640_dapm_widgets[] = { > > SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", > RT5640_PWR_ANLG1, > > RT5640_PWR_MA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, > > - SND_SOC_NOPM, 0, NULL, 0), > > + RT5640_PWR_HA_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_L_BIT, 0, NULL, 0), > > SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, > > RT5640_PWR_HP_R_BIT, 0, NULL, 0), > > SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, > > - SND_SOC_NOPM, 0, spk_event, > > + RT5640_PWR_CLS_D_BIT, 0, spk_event, > > SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), > > /* Output Lines */ > > SND_SOC_DAPM_OUTPUT("SPOLP"), > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH 0/6] R-Car sound driver support
by Kuninori Morimoto 29 Jul '13

29 Jul '13

Hi Mark These are basic Renesas R-Car sound driver support. Renesas R-Car series sound circuit consists of SSI and its peripheral. But this peripheral circuit is different between R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2). This driver aims at supporting both Gen1/Gen2. The basic driver style must be satisfactory. But it was tested on Gen1 board only at this point. Gen2 will be supported soon. And, it supports PIO transfer only at this point, DMA transfer will be supported soon. These are based on mark/for-next branch Kuninori Morimoto (6): ASoC: add Renesas R-Car core feature ASoC: add Renesas R-Car module feature ASoC: add Renesas R-Car Generation feature ASoC: add Renesas R-Car SCU feature ASoC: add Renesas R-Car ADG feature ASoC: add Renesas R-Car SSI feature include/sound/rcar_snd.h | 75 +++++ sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 +++++++++++++++ sound/soc/sh/rcar/core.c | 699 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/gen.c | 306 +++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 253 ++++++++++++++++ sound/soc/sh/rcar/scu.c | 124 ++++++++ sound/soc/sh/rcar/ssi.c | 581 ++++++++++++++++++++++++++++++++++++ 10 files changed, 2284 insertions(+) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/2] ASoC: lm4857: Use table based setup for DAPM and controls
by Lars-Peter Clausen 28 Jul '13

28 Jul '13

Let the ASoC core take care of registering the DAPM widget and routes as well as the controls. This makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/lm4857.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f595..5ea2ed0 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -174,28 +174,9 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { static int lm4857_probe(struct snd_soc_codec *codec) { struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; codec->control_data = lm4857->i2c; - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_new_widgets(dapm); - return 0; } @@ -207,6 +188,13 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .reg_word_size = sizeof(uint8_t), .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, + + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; static int lm4857_i2c_probe(struct i2c_client *i2c, -- 1.8.0

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[alsa-devel] [PATCH] ASoC: compress: use soc_xxx handlers for metadata
by Vinod Koul 28 Jul '13

28 Jul '13

the compress metadata handlers were wrongly named sst_xxx Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 06a8000..d220150 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -334,7 +334,7 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_set_metadata(struct snd_compr_stream *cstream, +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -347,7 +347,7 @@ static int sst_compr_set_metadata(struct snd_compr_stream *cstream, return ret; } -static int sst_compr_get_metadata(struct snd_compr_stream *cstream, +static int soc_compr_get_metadata(struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -364,8 +364,8 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, - .set_metadata = sst_compr_set_metadata, - .get_metadata = sst_compr_get_metadata, + .set_metadata = soc_compr_set_metadata, + .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

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[alsa-devel] RFC: support for 12 & 24Khz
by Vinod Koul 28 Jul '13

28 Jul '13

Hey Takashi, For compressed audio we also need to support the PCM rates of 12 and 24KHz. Looking at pcm.h these are not defined so we can simply add them at the end. But am worried about wider impact of adding these rates. Can you let me know if more is required to be done or below is fine diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 84b10f9..e418d8d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -126,6 +126,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_12000 <1<<13> /* 12000Hz */ +#define SNDRV_PCM_RATE_24000 <1<<14> /* 24000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a68d4c6..42600b0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1779,12 +1779,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_24000 != 1 << 14 #error "Change this table" #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000, + 12000, 24000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), ~Vinod --

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[alsa-devel] Garbage at end of playback: ecasound or alsa broken?
by Knut Petersen 28 Jul '13

28 Jul '13

Hi everybody! ====================================================================== Summary: A very short period (about 45ms) garbage is played at the end of every playback. ====================================================================== Hardware ======== mobo: AOpen i915GMm-hfs cpu: Pentium-M ram: 2GB intel hda, realtek codec rme digi96/pad onboard intel hda, realtek alc880 Software ======= opensuse 12.3 linux kernel 3.10.1, 3.11-rc1 ecasound, git ab6a9c3be5440b05f2c9eb240ee0d42578379f8d Steps to reproduce the problem: ========================= 1: generate test file =============== Start audacity, generate a 44.1 khz stereo track, one second audio, start 400 Hz, end 4000Hz, start amplitude 0.5, end amplitude 1.0 export to testin.wav as 16bit wav 2: playback test file and record it ========================= Setup soundcards and cabling, execute "ecasound -t 2 -f:16,2,44100 -a:1 -i testin.wav -o alsahw,0,0 -a:2 -i alsahw,0,0 -o testout.wav" 3:Expected result: ============== testout.wav should hold a proper recording of testin.wav, padded with silence at the end. 4: my result ========= testout.wav 1 - starts with about 90 Samples of silence (not nice, how can I avoid that?) 2 - continues with 1 second of properly recorded audio (expected) 3 - continues with about 1982 samples of audio (0,045s) (unexpected, severly broken) The garbage audio is part of the original audio, starting at about 26xxHz, ending at about 28xx Hz 4 - continues with silence to the end of the file (expected) Discussion ========= The same problem exists if I record rme96 to rme96, hda to rme96 or hda to hda. So it´s probably not a problem of the rme96 or hda alsa drivers. Is this problem known? Can anybody confirm the problem with different hardware? Should I blame alsa or ecasound? Any ideas? cu, Knut

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[alsa-devel] [PATCH 1/3] ASoC: smdk_wm8994: Make driver name more unique
by Mark Brown 27 Jul '13

27 Jul '13

From: Mark Brown <broonie(a)linaro.org> Avoid collisions with other SMDK audio by using the CODEC name. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/smdk_wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 581ea4a..05c479c 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -200,7 +200,7 @@ MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio", + .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), }, @@ -212,4 +212,4 @@ module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:smdk-audio"); +MODULE_ALIAS("platform:smdk-audio-wm8994"); -- 1.8.3.2

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Re: [alsa-devel] [PATCH 35/50] media: usb: cx231xx: spin_lock in complete() cleanup
by Hans Verkuil 27 Jul '13

27 Jul '13

On 07/11/2013 11:05 AM, Ming Lei wrote: > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: Hans Verkuil <hans.verkuil(a)cisco.com> > Cc: linux-media(a)vger.kernel.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > drivers/media/usb/cx231xx/cx231xx-audio.c | 6 ++++++ > drivers/media/usb/cx231xx/cx231xx-core.c | 10 ++++++---- > drivers/media/usb/cx231xx/cx231xx-vbi.c | 5 +++-- > 3 files changed, 15 insertions(+), 6 deletions(-) > > diff --git a/drivers/media/usb/cx231xx/cx231xx-audio.c b/drivers/media/usb/cx231xx/cx231xx-audio.c > index 81a1d97..58c1b5c 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-audio.c > +++ b/drivers/media/usb/cx231xx/cx231xx-audio.c > @@ -136,6 +136,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > for (i = 0; i < urb->number_of_packets; i++) { > + unsigned long flags; > int length = urb->iso_frame_desc[i].actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer + > @@ -158,6 +159,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); Can't you use snd_pcm_stream_lock_irqsave here? Ditto for the other media drivers where this happens: em28xx and tlg2300. I've reviewed the media driver changes and they look OK to me, so if my comment above is fixed, then I can merge them for 3.12. Or are these changes required for 3.11? Regards, Hans > > dev->adev.hwptr_done_capture += length; > @@ -174,6 +176,7 @@ static void cx231xx_audio_isocirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > @@ -224,6 +227,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > stride = runtime->frame_bits >> 3; > > if (1) { > + unsigned long flags; > int length = urb->actual_length / > stride; > cp = (unsigned char *)urb->transfer_buffer; > @@ -242,6 +246,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > length * stride); > } > > + local_irq_save(flags); > snd_pcm_stream_lock(substream); > > dev->adev.hwptr_done_capture += length; > @@ -258,6 +263,7 @@ static void cx231xx_audio_bulkirq(struct urb *urb) > period_elapsed = 1; > } > snd_pcm_stream_unlock(substream); > + local_irq_restore(flags); > } > if (period_elapsed) > snd_pcm_period_elapsed(substream); > diff --git a/drivers/media/usb/cx231xx/cx231xx-core.c b/drivers/media/usb/cx231xx/cx231xx-core.c > index 4ba3ce0..593b397 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-core.c > +++ b/drivers/media/usb/cx231xx/cx231xx-core.c > @@ -798,6 +798,7 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > int i; > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -813,9 +814,9 @@ static void cx231xx_isoc_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.isoc_ctl.isoc_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > for (i = 0; i < urb->number_of_packets; i++) { > @@ -842,6 +843,7 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, video_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -857,9 +859,9 @@ static void cx231xx_bulk_irq_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->video_mode.slock); > + spin_lock_irqsave(&dev->video_mode.slock, flags); > dev->video_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->video_mode.slock); > + spin_unlock_irqrestore(&dev->video_mode.slock, flags); > > /* Reset urb buffers */ > urb->status = usb_submit_urb(urb, GFP_ATOMIC); > diff --git a/drivers/media/usb/cx231xx/cx231xx-vbi.c b/drivers/media/usb/cx231xx/cx231xx-vbi.c > index c027942..38e78f8 100644 > --- a/drivers/media/usb/cx231xx/cx231xx-vbi.c > +++ b/drivers/media/usb/cx231xx/cx231xx-vbi.c > @@ -306,6 +306,7 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > struct cx231xx_video_mode *vmode = > container_of(dma_q, struct cx231xx_video_mode, vidq); > struct cx231xx *dev = container_of(vmode, struct cx231xx, vbi_mode); > + unsigned long flags; > > switch (urb->status) { > case 0: /* success */ > @@ -322,9 +323,9 @@ static void cx231xx_irq_vbi_callback(struct urb *urb) > } > > /* Copy data from URB */ > - spin_lock(&dev->vbi_mode.slock); > + spin_lock_irqsave(&dev->vbi_mode.slock, flags); > dev->vbi_mode.bulk_ctl.bulk_copy(dev, urb); > - spin_unlock(&dev->vbi_mode.slock); > + spin_unlock_irqrestore(&dev->vbi_mode.slock, flags); > > /* Reset status */ > urb->status = 0; >

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[alsa-devel] [PATCH] ASoc: kirkwood: free the external clock when same as the internal one
by Jean-Francois Moine 27 Jul '13

27 Jul '13

Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/{kirkwood-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index ba72039..bbadf53 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c~ +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -495,6 +495,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (!IS_ERR(priv->extclk)) { if (priv->extclk == priv->clk) { + devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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Re: [alsa-devel] Decibel Levels for Storm Medusa 5.1 USB Headset
by Raymond Yau 27 Jul '13

27 Jul '13

> >>> I hope this Information helps. I have the Storm Medusa 5.1 USB Headset > >>> which reports wrong decibel levels to pulseaudio. My Alsa-Info are > >>> located at > >>> > >>> > > http://www.alsa-project.org/db/?f=a882f3db6769c5dfce83c266946eff52d1d55349 > >>> The Headset is card2 in the above dump. > >>> > >>> My decibel levels-fix > >>> > >>> [DecibelFix Speaker] > >>> db-values = 1:-30 37:3 > >> How did you define these values? > > http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003867.h… > > > > If your headset have windows driver, need to find out which USB chip and > > look at the datasheet of the chip > > > > e.g. cm106l ’s datasheet contain a table of dac volume cortol > Ok looks like cm6206 is only in newer versions of the headset in older > (like mine) there is cm106l , maybe there are other chips as well but i > don't know. http://mailman.alsa-project.org/pipermail/alsa-devel/2007-November/003877.h… Seem same USB audio id idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0006 Do you find out the audio chip from product spec, user manual or windows driver ? Master volume control by default; per-channel volume control by C-Media driver Refer to the function block diagram , dac with volume controls 0 to -45dB There is a table of dac volume control in linear scale

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