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

[alsa-devel] ASUS Phoebus (CMI8888) HDA Support
by Geoffrey McRae 04 Dec '13

04 Dec '13

Hi, I have spent the better part of the last two days trying to add support to the hda-intel driver for this card and I have finally got it functional, not through any extra code though :( To get it going I added the extra device info into the device list and specified it as a generic device with no msi and LPIB position fix. Initially this did not work, probing for codec info failed, the interrupt was getting received, but the DMA address contained no data (all 0x0). After messing around a heap trying to find out why there was no DMA data, I discovered that if I forced the device into single_cmd mode it would operate, with very nasty scratchy audio, seems the IRQs were not getting through and the circular buffer was under running. If I made did something involving IRQs such as cat /proc/interrupts, it would briefly fix the problem... if I wrote a while loop to do just that, I would get mostly clean audio. There is another thread in this mailing list where someone else has tried all of this and ended up with the same output. I decided to try to identify why the device was not returning RIRB data in the DMA responses, and out of desperation setup a KVM with windows to try to perform mmio tracing to identify where the hda-intel driver was going wrong. This did not work, windows would not detect the codecs attached to the device, I think it is due to a lack of MSI support for passthrough devices. A positive thing did come out of this though, to do this I enabled MMIO in my BIOS, which instantly fixed the missing RIRB DMA data issue... I do not know enough about MMIO to understand why this fixed this. The device now detects the codec without single_cmd mode, but alas, still no audio, but also no errors... further experimentation I found that if I started the device in single_cmd mode and then went back to normal, I would get perfect audio now. Since I did this, I get perfect audio even without entering single_cmd mode. I have dumped the registers under windows and can confirm that during its initialization of the HDA device it does indeed use single_cmd mode before switching over to IRQ/DMA mode, there might be some standard init the driver is missing here. So, to make this card work, very little extra code will be required, just need to understand why enabling MMIO fixed the DMA issue as this is not a viable fix for the problem, not everyone has MMIO. Here is playback of a stereo stream while the device is working: [ 66.697114] hda_intel: Disabling MSI [ 66.697116] ALSA hda_intel.c:2553 Using LPIB position fix [ 66.697146] snd_hda_intel 0000:06:00.0: setting latency timer to 64 [ 66.697151] ALSA hda_intel.c:2796 chipset global capabilities = 0x4300 [ 66.697159] ALSA hda_intel.c:1175 Clearing TCSEL [ 66.724049] ALSA hda_intel.c:1031 codec_mask = 0x2 [ 66.724181] ALSA hda_intel.c:1484 codec #1 probed OK [ 66.734419] ALSA hda_generic.c:586 AUD_IN = 8 [ 66.734422] ALSA hda_generic.c:239 UNMUTE IN: NID=0x8 IDX=0x0 [ 66.734879] ALSA hda_generic.c:239 UNMUTE IN: NID=0x12 IDX=0x0 [ 66.735565] ALSA hda_generic.c:619 [Capture Source] NID=0x8, #SRC=1 [ 66.735567] ALSA hda_generic.c:622 [Front Mic] IDX=0x0 [ 66.735568] ALSA hda_generic.c:291 AUD_OUT found 2 [ 66.735570] ALSA hda_generic.c:239 UNMUTE IN: NID=0xc IDX=0x0 [ 66.735903] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736136] ALSA hda_generic.c:224 UNMUTE OUT: NID=0xc [ 66.736139] ALSA hda_generic.c:291 AUD_OUT found 6 [ 66.736569] ALSA hda_generic.c:254 CONNECT: NID=0x8 IDX=0x0 [ 66.736572] ALSA hda_generic.c:699 [Capture Switch] NID=0x8, DIR=IN, IDX=0x0 [ 66.736575] ALSA hda_generic.c:725 [Capture Volume] NID=0x8, DIR=IN, IDX=0x0 [ 66.736578] ALSA hda_generic.c:710 [Master Playback Switch] NID=0x2, DIR=OUT [ 66.736580] ALSA hda_generic.c:734 [Master Playback Volume] NID=0x2, DIR=OUT [ 76.125606] ALSA hda_intel.c:1848 azx_pcm_prepare: bufsize=0x10000, format=0x31 [ 76.125615] ALSA hda_codec.c:1414 hda_codec_setup_stream: NID=0x2, stream=0x7, channel=0, format=0x31 [ 93.308666] ALSA hda_codec.c:1477 hda_codec_cleanup_stream: NID=0x2 - Geoffrey McRae

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[alsa-devel] [PATCH] ASoC: rcar: use devm_clk_get() instead of clk_get()
by Kuninori Morimoto 04 Dec '13

04 Dec '13

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/ssi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index a821b14..85936ae 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -653,7 +653,7 @@ int rsnd_ssi_probe(struct platform_device *pdev, snprintf(name, RSND_SSI_NAME_SIZE, "ssi.%d", i); - clk = clk_get(dev, name); + clk = devm_clk_get(dev, name); if (IS_ERR(clk)) return PTR_ERR(clk); @@ -714,7 +714,6 @@ void rsnd_ssi_remove(struct platform_device *pdev, int i; for_each_rsnd_ssi(ssi, priv, i) { - clk_put(ssi->clk); if (rsnd_ssi_dma_available(ssi)) rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod)); } -- 1.7.9.5

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[alsa-devel] [PATCH v2 0/2] Only do snd_soc_dai_digital_mute() in hw_free()
by Nicolin Chen 04 Dec '13

04 Dec '13

This series of patches mainly fixes an issue of snd_soc_dai_digital_mute() operation after playback/capture is done. The current system has two of them in a normal PCM instance while one of them has an invalid condition. Thus we first fix the condition and then drop the redundant one. Changelog: v2: * Bisected into two patches. * Fixed indentation in PATCH-1. Nicolin Chen (2): ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() ASoC: soc-pcm: Drop the redundant snd_soc_dai_digital_mute() in soc_pcm_close() sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) -- 1.8.4

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[alsa-devel] ASoC fixes for v3.13
by Mark Brown 04 Dec '13

04 Dec '13

The following changes since commit dc1ccc48159d63eca5089e507c82c7d22ef60839: Linux 3.13-rc2 (2013-11-29 12:57:14 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc2 for you to fetch changes up to 29e248829dc7d44248c69bbd5d40eca152a50cab: Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus (2013-12-03 18:09:00 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A smattering of fixes here, some core ones for the rate combination issues for things other than simple bitmasks, for readback of byte controls and for updating the power of value muxes plus a bunch of driver fixes of varying severity. The warning fix in the i.MX FIQ driver is fixing a warning introduced by a previous fix. ---------------------------------------------------------------- Bo Shen (2): ASoC: atmel: sam9x5_wm8731: fix oops when unload module ASoC: wm8731: fix dsp mode configuration Charles Keepax (2): ASoC: wm5110: Remove output OSR and PGA volume controls ASoC: core: Use consistent byte ordering in snd_soc_bytes_get Fabio Estevam (1): ASoC: fsl: imx-pcm-fiq: Remove unused 'runtime' variable Jarkko Nikula (1): ASoC: omap: n810: Convert to clk_prepare_enable/clk_disable_unprepare Jean-Francois Moine (2): ASoC: kirkwood: Fix invalid S/PDIF format ASoC: kirkwood: Fix erroneous double output while playing Kuninori Morimoto (1): ASoC: rcar: select REGMAP Lars-Peter Clausen (2): ASoC: pcm: Fix rate_max calculation ASoC: pcm: Always honor DAI min and max sample rate constraints Mark Brown (4): ASoC: wm8990: Mark the register map as dirty when powering down Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/atmel', 'asoc/fix/fsl', 'asoc/fix/kirkwood', 'asoc/fix/omap', 'asoc/fix/rcar', 'asoc/fix/wm8731' and 'asoc/fix/wm8990' into asoc-linus Oskar Schirmer (1): ASoC: fsl: imx-pcm-fiq: remove bogus period delta calculation Shawn Guo (1): ASoC: core: fix devres parameter in devm_snd_soc_register_card() Stephen Warren (1): ASoC: dapm: Use SND_SOC_DAPM_INIT_REG_VAL in SND_SOC_DAPM_MUX Wei Yongjun (1): ASoC: fsl: set correct platform drvdata in pcm030_fabric_probe() include/sound/soc-dapm.h | 3 ++- sound/soc/atmel/sam9x5_wm8731.c | 2 ++ sound/soc/codecs/wm5110.c | 25 ------------------------- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8990.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 3 ++- sound/soc/kirkwood/kirkwood-i2s.c | 22 +++++++++++++--------- sound/soc/omap/n810.c | 4 ++-- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 4 ++-- sound/soc/soc-devres.c | 4 ++-- sound/soc/soc-pcm.c | 18 ++++++++++++------ 12 files changed, 42 insertions(+), 50 deletions(-)

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Implement symmetric_channels and symmetric_samplebits
by Nicolin Chen 04 Dec '13

04 Dec '13

Since we introduced symmetric_channels and symmetric_samplebits, we implement these two features to fsl_ssi so as to drop some no-more-needed code and make the driver neat and clean. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 79 ++++++------------------------------------------- 1 file changed, 9 insertions(+), 70 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 90ff107..f9f4569 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -119,8 +119,6 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @first_stream: pointer to the stream that was opened first - * @second_stream: pointer to second stream * @playback: the number of playback streams opened * @capture: the number of capture streams opened * @cpu_dai: the CPU DAI for this device @@ -132,8 +130,6 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct snd_pcm_substream *first_stream; - struct snd_pcm_substream *second_stream; unsigned int fifo_depth; struct snd_soc_dai_driver cpu_dai_drv; struct device_attribute dev_attr; @@ -438,54 +434,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - int synchronous = ssi_private->cpu_dai_drv.symmetric_rates; - /* - * If this is the first stream opened, then request the IRQ - * and initialize the SSI registers. + /* First, we only do fsl_ssi_setup() when SSI is going to be active. + * Second, fsl_ssi_setup was already called by ac97_init earlier if + * the driver is in ac97 mode. */ - if (!ssi_private->first_stream) { - ssi_private->first_stream = substream; - - /* - * fsl_ssi_setup was already called by ac97_init earlier if - * the driver is in ac97 mode. - */ - if (!ssi_private->imx_ac97) - fsl_ssi_setup(ssi_private); - } else { - if (synchronous) { - struct snd_pcm_runtime *first_runtime = - ssi_private->first_stream->runtime; - /* - * This is the second stream open, and we're in - * synchronous mode, so we need to impose sample - * sample size constraints. This is because STCCR is - * used for playback and capture in synchronous mode, - * so there's no way to specify different word - * lengths. - * - * Note that this can cause a race condition if the - * second stream is opened before the first stream is - * fully initialized. We provide some protection by - * checking to make sure the first stream is - * initialized, but it's not perfect. ALSA sometimes - * re-initializes the driver with a different sample - * rate or size. If the second stream is opened - * before the first stream has received its final - * parameters, then the second stream may be - * constrained to the wrong sample rate or size. - */ - if (first_runtime->sample_bits) { - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); - } - } - - ssi_private->second_stream = substream; - } + if (!dai->active && !ssi_private->imx_ac97) + fsl_ssi_setup(ssi_private); return 0; } @@ -615,23 +570,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -/** - * fsl_ssi_shutdown: shutdown the SSI - * - * Shutdown the SSI if there are no other substreams open. - */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); - - if (ssi_private->first_stream == substream) - ssi_private->first_stream = ssi_private->second_stream; - - ssi_private->second_stream = NULL; -} - static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(dai); @@ -647,7 +585,6 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, }; @@ -722,7 +659,6 @@ static int fsl_ssi_ac97_trigger(struct snd_pcm_substream *substream, int cmd, static const struct snd_soc_dai_ops fsl_ssi_ac97_dai_ops = { .startup = fsl_ssi_startup, - .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_ac97_trigger, }; @@ -947,8 +883,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } /* Are the RX and the TX clocks locked? */ - if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) + if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } /* Determine the FIFO depth. */ iprop = of_get_property(np, "fsl,fifo-depth", NULL); -- 1.8.4

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[alsa-devel] [PATCH] Add no-mmu support for alsa-utils
by Aaron Wu 04 Dec '13

04 Dec '13

From: Aaron Wu <Aaron.wu(a)analog.com> Signed-off-by: Aaron Wu <Aaron.wu(a)analog.com> Add no-mmu support for alsa-utils mainly for Blackfin --- alsactl/init_utils_run.c | 4 ++++ alsaloop/alsaloop.c | 8 ++++++++ configure.in | 1 + 3 files changed, 13 insertions(+) diff --git a/alsactl/init_utils_run.c b/alsactl/init_utils_run.c index dde490b..ac074e6 100644 --- a/alsactl/init_utils_run.c +++ b/alsactl/init_utils_run.c @@ -89,7 +89,11 @@ int run_program0(struct space *space, argv[0] = program; } + #ifdef HAVE_FORK pid = fork(); + #else + pid = vfork(); + #endif switch(pid) { case 0: /* child closes parent ends of pipes */ diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 3b20682..1b1e540 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -861,14 +861,22 @@ int main(int argc, char *argv[]) logit(LOG_CRIT, "daemon() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } + #ifdef HAVE_FORK i = fork(); + #else + i = vfork(); + #endif if (i < 0) { logit(LOG_CRIT, "fork() failed: %s\n", strerror(errno)); exit(EXIT_FAILURE); } if (i > 0) { /* wait(&i); */ + #ifdef HAVE_FORK exit(EXIT_SUCCESS); + #else + _exit(EXIT_SUCCESS); + #endif } } diff --git a/configure.in b/configure.in index 60da1fa..f289f9b 100644 --- a/configure.in +++ b/configure.in @@ -25,6 +25,7 @@ then AC_MSG_RESULT($CC) fi +AC_CHECK_FUNC([fork]) AC_PROG_CC dnl AC_PROG_CXX AC_PROG_INSTALL -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Fix missing ELD info when using jackpoll_ms parameter
by David Henningsson 04 Dec '13

04 Dec '13

In the case of using jackpoll_ms instead of unsol events, the jack was correctly detected, but ELD info was not refreshed on plug-in. And without ELD info, no proper restriction of pcm, which can in turn break sound output on some devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) @Takashi; feel free to add stable if you think it's worth it. (After all using the jackpoll_ms parameter is not something we usually do.) It's still a bug fix so sending to Linus tree should be okay, I believe. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4098196..f0a8ea8 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hdmi_spec *spec = codec->spec; + int pin_idx = pin_nid_to_pin_index(spec, jack->nid); + if (pin_idx < 0) + return; + + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); +} + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pin_nid; - int pin_idx; struct hda_jack_tbl *jack; int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) return; - pin_nid = jack->nid; jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - pin_idx = pin_nid_to_pin_index(spec, pin_nid); - if (pin_idx < 0) - return; - - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + jack_callback(codec, jack); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -2080,7 +2082,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid, + codec->jackpoll_interval > 0 ? jack_callback : NULL); } return 0; } -- 1.7.9.5

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[alsa-devel] Independent of model for HP
by Kailang 04 Dec '13

04 Dec '13

Hi Takashi, HP Chrome Book, the headset jack was difference between other chrome book. It need to manual control Mic jack detect. Thanks. Kailang

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Re: [alsa-devel] [pulseaudio-discuss] What-did-you-plugin-dialog
by Raymond Yau 04 Dec '13

04 Dec '13

: > > > can post the output of alsa-info.sh after jack retasking ? > > Before retasking: > > http://www.alsa-project.org/db/?f=fd97b8194e205b4f91127bca74ad7c9b2cd93ab9 > > After retasking blue line in, pin ID 0x1a as Line out (Side): > > http://www.alsa-project.org/db/?f=d56f9a6d0692c9a8d5c88f43100a6cb78407c163 > > Note that retasking did not work: attempting to play a 8-channel wav > file through hw:1,0 resulted in an "Invalid argument" error. > > FWIW, here is what hdajackretask printed to the console: > > 0x11 0x99430130 > 0x12 0x4037c540 > 0x14 0x01014010 > 0x15 0x01011012 > 0x16 0x01016011 > 0x17 0x411111f0 > 0x18 0x01a19050 > 0x19 0x02a19060 > 0x1a 0x01014413 > 0x1b 0x02214020 > 0x1c 0x411111f0 > 0x1d 0x4045e601 > 0x1e 0x01452140 > 0x1f 0x411111f0 > 1 > > you need root privilege for hdajackretask to write the firmware file to /lib/firmware and modify /etc/modprobe.d/alsa-lbase.conf to use the firmware file can you try the patch which add "channel mode" control which allow you to switch from 6ch to 8ch it is not clear that the "Line jack" control is used for detect the blue jack state or reflect the line In is no longer available after it is retasked as output jack

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[alsa-devel] [PATCH - plugins] oss: avoid conflict between reclev and igain mixer ctls
by Luca Bayer 04 Dec '13

04 Dec '13

To fix the following error that happens with Intel HDA driver. $ amixer ALSA lib simple_none.c:1549:(simple_add1) helem (MIXER,'Capture Volume',0,0,0) appears twice or more amixer: Mixer default load error: Invalid argument Signed-off-by: Luca Bayer <kometto(a)web.de> diff --git a/oss/ctl_oss.c b/oss/ctl_oss.c index b8847b1..b05ba18 100644 --- a/oss/ctl_oss.c +++ b/oss/ctl_oss.c @@ -52,7 +52,7 @@ static const char *const vol_devices[SOUND_MIXER_NRDEVICES] = { [SOUND_MIXER_CD] = "CD Playback Volume", [SOUND_MIXER_IMIX] = "Monitor Mix Playback Volume", [SOUND_MIXER_ALTPCM] = "Headphone Playback Volume", - [SOUND_MIXER_RECLEV] = "Capture Volume", + [SOUND_MIXER_RECLEV] = "Master Capture Volume", [SOUND_MIXER_IGAIN] = "Capture Volume", [SOUND_MIXER_OGAIN] = "Playback Volume", [SOUND_MIXER_LINE1] = "Aux Playback Volume",

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