January 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ALSA: hda - Fix possible races in HDMI driver - lockup on shutdown when radeon.audio=1 after using audacity
by Arthur Marsh 21 Jan '14

21 Jan '14

I have had reproducible lock-ups on shut-down (at the shutting down ALSA stage) of my AMD64 machine (Asus M3A78Pro motherboard, BIOS 1701 01/27/2011, CPU AMD Athlon(tm) II X4 640 Processor) running the 64 bit Linux kernel more recent than 3.12 when *both* radeon.audio=1 was set and I had been running audacity 2.0.5. (iommu=noaperture is also set). The problem was reproducible with the stock Debian kernel linux-image-3.13-rc6-amd64 version 3.13~rc6-1~exp1. The machine is using an ATI/AMD 3850HD video card with a DVI cable to a DVI input on my monitor, and the default audio device is the motherboard's on-board audio device: 00:14.2 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] SBx00 Azalia (Intel HDA) 01:00.0 VGA compatible controller: Advanced Micro Devices, Inc. [AMD/ATI] RV670 [Radeon HD 3690/3850] 01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] RV670/680 HDMI Audio [Radeon HD 3690/3800 Series] $ git bisect bad cbbaa603a03cc46681e24d6b2804b62fde95a2af is the first bad commit commit cbbaa603a03cc46681e24d6b2804b62fde95a2af Author: Takashi Iwai <tiwai(a)suse.de> Date: Thu Oct 17 18:03:24 2013 +0200 ALSA: hda - Fix possible races in HDMI driver Some per_pin fields and ELD contents might be changed dynamically in multiple ways where the concurrent accesses are still opened in the current code. This patch fixes such possible races by using eld->lock in appropriate places. Reported-by: Anssi Hannula <anssi.hannula(a)iki.fi> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 0c29281f82a3ebd9a704d481114f9cfcefea07c8 d71fd101125cd29a628cb5e66c7ee4f56e28329b M sound When running audacity from the command line there was the following output: ALSA lib pcm_dsnoop.c:618:(snd_pcm_dsnoop_open) unable to open slave ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 Expression 'stream->playback.pcm' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 4611 I am happy to supply further information or run further tests to help in isolating the problem and verifying a solution. Arthur.

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[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLLgeneration
by Daniel Matuschek 20 Jan '14

20 Jan '14

WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Tested-by: Florian Meier <florian.meier(a)koalo.de> --- sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206..4163786 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.8.3.2

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[alsa-devel] ASoC updates for v3.14
by Mark Brown 20 Jan '14

20 Jan '14

The following changes since commit 258753361534a40ad7180c742da813fc659e427b: Merge branch 'for-next' into for-linus (2014-01-20 10:20:14 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.14-3 for you to fetch changes up to ac6d7c48e3a5e82e876114b9ca0b489e96997743: Merge remote-tracking branch 'asoc/topic/compress' into asoc-next (2014-01-20 11:50:41 +0000) ---------------------------------------------------------------- ASoC: Last updates for the merge window A couple more fixes plus some extensions to DPCM for use with compressed audio from Liam which arrived just after my previous pull request. ---------------------------------------------------------------- Arun Shamanna Lakshmi (1): ASoC: dapm: Fix double prefix addition Liam Girdwood (2): ASoC: DPCM: make some DPCM API calls non static for compressed usage ASoC: compress: Add suport for DPCM into compressed audio Mark Brown (3): Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Xiubo Li (1): ASoC: core: Fix possible NULL pointer dereference of pcm->config include/sound/soc-dpcm.h | 22 +++ include/sound/soc.h | 1 + sound/soc/soc-compress.c | 301 +++++++++++++++++++++++++++++++++- sound/soc/soc-dapm.c | 11 +- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/soc-pcm.c | 29 ++-- 6 files changed, 347 insertions(+), 26 deletions(-)

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[alsa-devel] [PATCH v3 01/16] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 20 Jan '14

20 Jan '14

This function performs waiting when the SPI bus completes a transaction. Timeout error checking introduced and the timeout increased to 400 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..a917f37 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 400 microseconds */ + unsigned int count; + for (count = 100; count > 0; count--) { + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + udelay(4); + --count; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

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[alsa-devel] Thinkpad T520 (Low Volume)
by Raymond Yau 20 Jan '14

20 Jan '14

> 2014/1/16 Axel Braun <axel.braun(a)gmx.de> > same problem on a ThinkPad T520, sound volume is way too low - watching movie via loudspeakers is fairly impossible. > Any hint you could give me? > > post the output of alsa-info.sh , there is a bug in alsa-driver which treat > dock headphone and headphone to support surround 4.0 you get low volume because Amp Out vals are 0x3d instead of 0x4a but speaker volumes are at 100% (0dB) Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ca2… thinkpad t520 was supported with CXT5066_AUTO which use cxt5066_update_speaker which check port a and port d Do this mean thinkpad don't support surround 4.0 by dock headphone and headphone ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda/conexant - Set up the missing docking-station pins [ 9.956636] ALSA hda_auto_parser.c:393 autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker [ 9.956639] ALSA hda_auto_parser.c:397 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 9.956641] ALSA hda_auto_parser.c:401 hp_outs=2 (0x1c/0x19/0x0/0x0/0x0) [ 9.956643] ALSA hda_auto_parser.c:402 mono: mono_out=0x0 [ 9.956644] ALSA hda_auto_parser.c:406 inputs: [ 9.956646] ALSA hda_auto_parser.c:410 Mic=0x1b [ 9.956647] ALSA hda_auto_parser.c:410 Internal Mic=0x23 [ 9.956649] ALSA hda_auto_parser.c:410 Dock Mic=0x1a the driver select this as best config ==> Best config: lo_type=1, wired=1, mio=1 multi_outs = 1f/0/0/0 : 10/0/0/0 (type SP) out path: depth=2 :10:1f hp_outs = 19/1c/0/0 : 11/11/0/0 hp path: depth=2 :11:19 hp path: depth=2 :11:1c https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… ALSA: hda - Fix multi-io channel mode management this patch change the the channel_count from 2 to 4 since there are two headphone + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + spec->const_channel_count = max(spec->const_channel_count, + cfg->hp_outs * 2); if you don't have docking station , the workaround is to remove those pin fix of node 0x1a and 0x1c

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[alsa-devel] valgrind error in snd_tlv_get_dB_range
by Ivan Sorokin 20 Jan '14

20 Jan '14

One program (qasmixer -- alsa mixer) on my system prints lots of valgrind errors. The first one is this: ==25716== Conditional jump or move depends on uninitialised value(s) ==25716== at 0x4E72307: snd_tlv_get_dB_range (tlv.c:170) ==25716== by 0x4E8A746: get_dB_range (simple_none.c:1162) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) ==25716== by 0x4DB863: Desktop_Items::tray_mixer_create() (desktop_items.cpp:504) ==25716== Uninitialised value was created by a heap allocation ==25716== at 0x4C2CD7B: malloc (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so) ==25716== by 0x4E8A572: init_db_range (simple_none.c:1114) ==25716== by 0x4E8A70D: get_dB_range (simple_none.c:1159) ==25716== by 0x4E8A7E7: get_dB_range_ops (simple_none.c:1176) ==25716== by 0x4E85590: snd_mixer_selem_get_playback_dB_range (simple.c:298) ==25716== by 0x472B2B: QSnd::Mixer_Simple_Elem::set_snd_mixer_selem_id(_snd_mixer_selem_id*) (mixer_simple_elem.cpp:161) ==25716== by 0x4724B8: QSnd::Mixer_Simple_Elem::Mixer_Simple_Elem(QObject*, _snd_mixer*, _snd_mixer_selem_id*) (mixer_simple_elem.cpp:31) ==25716== by 0x479F17: QSnd::Mixer_Simple::create_mixer_elem(_snd_mixer_elem*) (mixer_simple.cpp:318) ==25716== by 0x4797B3: QSnd::Mixer_Simple::create_mixer_elems() (mixer_simple.cpp:217) ==25716== by 0x4795DF: QSnd::Mixer_Simple::open(QString const&) (mixer_simple.cpp:178) ==25716== by 0x4D4754: Tray_Mixer::load_mixer() (tray_mixer.cpp:222) ==25716== by 0x4D4505: Tray_Mixer::set_mdev_setup(Tray_Mixer_MDev_Setup*) (tray_mixer.cpp:137) I compiled libasound2 from the latest git from git://gitorious.org/alsa/alsa-lib.git with -g -O0, so you can see line numbers. As I understand the problem is that init_db_range initialize invalid rec->db_info, because snd_hctl_elem_tlv_read failed to fully initialize tlv: > tlv = malloc(tlv_size); // line 1114 > if (! tlv) > return -ENOMEM; > if (snd_hctl_elem_tlv_read(ctl, tlv, tlv_size) < 0) > goto error; > db_size = snd_tlv_parse_dB_info(tlv, tlv_size, &dbrec); > if (db_size < 0) > goto error; > rec->db_info = malloc(db_size); > if (!rec->db_info) > goto error; > memcpy(rec->db_info, dbrec, db_size); > free(tlv); I tried to trace problem in snd_hctl_elem_tlv_read, but I failed to do so due to lack of experience with alsa-lib. Could someone more experienced with alsa-lib look at the problem?

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[alsa-devel] Realtec ALC898 supported?
by Haug Bürger 20 Jan '14

20 Jan '14

Hi, I would like to know if the new Realtec 89x chips are supported. I found drivers on the Realtec site but the seem to be older than the alsa-project.org drivers and I'm not sure if they support the 89x chips. thanks Haug

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[alsa-devel] [PATCH] ASoC: core: Use devm_kzalloc() instead kzalloc()
by Xiubo Li 20 Jan '14

20 Jan '14

Makes the code slightly shorter Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5bace12..bfb012f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -388,7 +388,7 @@ int snd_dmaengine_pcm_register(struct device *dev, struct dmaengine_pcm *pcm; int ret; - pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + pcm = devm_kzalloc(dev, sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; @@ -408,7 +408,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); - kfree(pcm); return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -433,7 +432,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) snd_soc_remove_platform(platform); dmaengine_pcm_release_chan(pcm); - kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); -- 1.8.4

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[alsa-devel] [PATCH] ASoC: dapm: Fix double prefix addition
by Arun Shamanna Lakshmi 17 Jan '14

17 Jan '14

The prefix for the codec driver can be used during dual identical codec usecases. However, dapm adds prefix twice for codec DAI widget in snd_soc_dapm_add_route API. This change is to avoid double prefix addition for codec DAI widget and is needed while using identical dual codecs. Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> --- sound/soc/soc-dapm.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2a44fe9..dc8ff13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2476,7 +2476,8 @@ err: } static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_route *route) + const struct snd_soc_dapm_route *route, + unsigned int is_prefixed) { struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; struct snd_soc_dapm_widget *wtsource = NULL, *wtsink = NULL; @@ -2486,7 +2487,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_source[80]; int ret; - if (dapm->codec && dapm->codec->name_prefix) { + if (dapm->codec && dapm->codec->name_prefix && !is_prefixed) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", dapm->codec->name_prefix, route->sink); sink = prefixed_sink; @@ -2614,7 +2615,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - r = snd_soc_dapm_add_route(dapm, route); + r = snd_soc_dapm_add_route(dapm, route, false); if (r < 0) { dev_err(dapm->dev, "ASoC: Failed to add route %s -> %s -> %s\n", route->source, @@ -3670,7 +3671,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) cpu_dai->codec->name, r.source, codec_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } /* connect BE DAI capture if widgets are valid */ @@ -3681,7 +3682,7 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) codec_dai->codec->name, r.source, cpu_dai->platform->name, r.sink); - snd_soc_dapm_add_route(&card->dapm, &r); + snd_soc_dapm_add_route(&card->dapm, &r, true); } } -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Florian Meier 17 Jan '14

17 Jan '14

On 01/17/2014 07:33 PM, Mark Brown wrote: > On Fri, Jan 17, 2014 at 07:06:24PM +0100, Florian Meier wrote: >> > Intentionally off-list? Oh no - I am sorry! >> If I remember correctly the error was >> "codec can not start from non-off bias with idle_bias_off==true" > >> I think the solution is just to set idle_bias_off = false and everything >> seems to be working with that. I just don't know if there might be any >> side effects. > > Setting it to false increases power consumption since the device is > kept more powered on when idle but reduces startup time from idle. For > digital only devices like the wm8804 there shouldn't be any reason to > keep it powered up when not in use, the startup time is generally > negligable anyway. > So a better solution would be to set SND_SOC_BIAS_OFF instead of SND_SOC_BIAS_STANDBY at the end of probe?

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