April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 20 Apr '15

20 Apr '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection" mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 4 +++ drivers/extcon/extcon-arizona.c | 34 ++++++++++++++++++++--- include/linux/mfd/arizona/pdata.h | 3 ++ 3 files changed, 37 insertions(+), 4 deletions(-) -- 2.0.0.390.gcb682f8

2 7

[alsa-devel] How to correct params_format on DPCM ?
by Kuninori Morimoto 20 Apr '15

20 Apr '15

Hi Mark, Lars, all Now, I'm using DPCM based sound, but FE/BE gets different params format. How to correct it ? This sound can be used both non-DPCM card / DPCM card -- non-DPCM card -- cpu: rsnd codec: ak4642 -- DPCM card -- FE : cpu: rsnd codec: snd-soc-dummy BE : cpu: snd-soc-dummy codec: ak4642 Here, SNDRV_PCM_FMTBIT_S16_LE is always selected on non-DPCM card. This is OK. rsnd: SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE ak4642: SNDRV_PCM_FMTBIT_S16_LE On DPCM card, FE gets S24, BE gets S16 if aplay selects S24 wav. But, FE needs S16, same as non-DPCM card. I know we can fixup BE side via dai_link.be_hw_params_fixup, but this is FE side. It can't use correct sound playback in corrent DPCM card. But, how to solve this issue ?? Best regards --- Kuninori Morimoto

2 7

[alsa-devel] [PATCH] ASoC: add static inline funcs to fix a compiling issue
by Jie Yang 19 Apr '15

19 Apr '15

When CONFIG_PM_SLEEP is not selected, calling funcs snd_soc_suspend and _resume will generate a compiling issue. Here add static inline stub functions to fix it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2afdd2e..64041f5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -387,8 +387,20 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); +#ifdef CONFIG_PM_SLEEP int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); +#else +static inline int snd_soc_suspend(struct device *dev) +{ + return 0; +} + +static inline int snd_soc_resume(struct device *dev) +{ + return 0; +} +#endif int snd_soc_poweroff(struct device *dev); int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix a compiling issue
by Jie Yang 19 Apr '15

19 Apr '15

Fix the compiling issue that we don't have snd_soc_suspend/resume() when CONFIG_PM_SLEEP is not defined. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae040..f84eb14 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1313,8 +1313,9 @@ static void hsw_pcm_complete(struct device *dev) dev_err(dev, "failed to restore context for PCM %d\n", i); } +#ifdef CONFIG_PM_SLEEP snd_soc_resume(pdata->soc_card->dev); - +#endif err = sst_hsw_dsp_runtime_resume(hsw); if (err < 0) return; @@ -1367,7 +1368,9 @@ static int hsw_pcm_prepare(struct device *dev) sst_hsw_dsp_runtime_sleep(hsw); } +#ifdef CONFIG_PM_SLEEP snd_soc_suspend(pdata->soc_card->dev); +#endif snd_soc_poweroff(pdata->soc_card->dev); pdata->pm_state = HSW_PM_STATE_D3; -- 1.9.1

5 6

[alsa-devel] [PATCH] ASoC: at91sam9g20ek: Automatically disconnect non-connected pins
by Lars-Peter Clausen 18 Apr '15

18 Apr '15

According to the schematics the both LHPOUT and RHPOUT are connected to the external connector. RHPOUT is missing from the DAPM routes, but otherwise they seem to be complete. This patch adds the missing route and then sets the fully_routed flag for the card. This allows to remove all the manual calls to snd_soc_dapm_nc_pin(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/sam9g20_wm8731.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 8de8361..d7469cd 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -95,8 +95,9 @@ static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { static const struct snd_soc_dapm_route intercon[] = { - /* speaker connected to LHPOUT */ + /* speaker connected to LHPOUT/RHPOUT */ {"Ext Spk", NULL, "LHPOUT"}, + {"Ext Spk", NULL, "RHPOUT"}, /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, @@ -108,9 +109,7 @@ static const struct snd_soc_dapm_route intercon[] = { */ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; printk(KERN_DEBUG @@ -124,10 +123,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* not connected */ - snd_soc_dapm_nc_pin(dapm, "RLINEIN"); - snd_soc_dapm_nc_pin(dapm, "LLINEIN"); - #ifndef ENABLE_MIC_INPUT snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic"); #endif @@ -158,6 +153,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), .dapm_routes = intercon, .num_dapm_routes = ARRAY_SIZE(intercon), + .fully_routed = true, }; static int at91sam9g20ek_audio_probe(struct platform_device *pdev) -- 1.8.0

3 5

Re: [alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Use specific variable for lo_vag
by Mark Brown 18 Apr '15

18 Apr '15

On Thu, Apr 16, 2015 at 02:51:56PM +0200, Alexander Stein wrote: > This is a preparation for calculating lo_vol which needs both vag > and lo_vag. Applied both, thanks.

1 0

[alsa-devel] Understanding Atmel SSC SND_SOC_DAIFMT_CBS_CFS mode
by Angelo Compagnucci 18 Apr '15

18 Apr '15

Hi List, I'm looking for a way to understand how to correctly configure the Atmel SSC dai on a SAMA5D3 to work in master mode with an I2S codec. >From this comment: /* * I2S format, SSC provides BCLK and LRC clocks. * * The SSC transmit and receive clocks are generated * from the MCK divider, and the BCLK signal * is output on the SSC TK line. */ My question is: should be MCK divider configured manually? If yes how? Actually I'm trying to poke around in atmel_ssc_dai and found a way to generate a stable bit clock, but frame clock is not working as expected. Have you any advice on how to set correctly SSC parameters to generate a stable bit clock and frame clock? Any advice? Thank you! -- Profile: http://it.linkedin.com/in/compagnucciangelo

1 0

[alsa-devel] Support for Synchronous Audio
by Anoop Rajan 18 Apr '15

18 Apr '15

Hi All, My self Anoop, an audio frame work developer, but a newbie to linux ALSA. We have requirements to play synchronous multi-room audio with very minimal node to node latency(<100 us).( Node to node latency: How much one speaker is out of sync with other). Right now we have logic in user space to synchronize the audio using time stamps derived from a common clock, but the node to node latency is over 1ms. Though I make sure that I start the PCM from user layer synchronously, the actual I2S out is out of sync(in consistent values ranging from 100 to 1ms). I beleive the un-deterministic path delay from user to ALSA buffer, big ALSA buffer(8k Samples), scheduling variances(high priority/interrupt preemption) all might have added to this increased latency. Rather than fine tuning the above discrepancies , I was thinking about some designs to pass the audio frames with time stamps to kernel layer and to implement some logic in kernel (before pushing to DMA) to present the audio at exact stamped time to have a better node to node latency. Before doing that I was trying to find out what ALSA supports for this requirement and from web could see various comments regarding this (proposal to snd_pcm_start_at()<http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/082091.ht…>) and further proposed discussions on Audio mini summit, but I couldn't find further information's on it. Would you please guide me. Does ALSA provides some means to achieve minimal Node to Node latency? would you please provide some pointers/documents to me to understand better. Thanks, Anoop

2 3

[alsa-devel] [PATCH] sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND)
by Alexey Khoroshilov 18 Apr '15

18 Apr '15

A deadlock can be initiated by userspace via ioctl(SNDCTL_SEQ_OUTOFBAND) on /dev/sequencer with TMR_ECHO midi event. In this case the control flow is: sound_ioctl() -> case SND_DEV_SEQ: case SND_DEV_SEQ2: sequencer_ioctl() -> case SNDCTL_SEQ_OUTOFBAND: spin_lock_irqsave(&lock,flags); play_event(); -> case EV_TIMING: seq_timing_event() -> case TMR_ECHO: seq_copy_to_input() -> spin_lock_irqsave(&lock,flags); It seems that spin_lock_irqsave() around play_event() is not necessary, because the only other call location in seq_startplay() makes the call without acquiring spinlock. So, the patch just removes spinlocks around play_event(). By the way, it removes unreachable code in seq_timing_event(), since (seq_mode == SEQ_2) case is handled in the beginning. Compile tested only. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/oss/sequencer.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c0eea1dfe90f..f19da4b47c1d 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec) break; case TMR_ECHO: - if (seq_mode == SEQ_2) - seq_copy_to_input(event_rec, 8); - else - { - parm = (parm << 8 | SEQ_ECHO); - seq_copy_to_input((unsigned char *) &parm, 4); - } + parm = (parm << 8 | SEQ_ECHO); + seq_copy_to_input((unsigned char *) &parm, 4); break; default:; @@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a int mode = translate_mode(file); struct synth_info inf; struct seq_event_rec event_rec; - unsigned long flags; int __user *p = arg; orig_dev = dev = dev >> 4; @@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a case SNDCTL_SEQ_OUTOFBAND: if (copy_from_user(&event_rec, arg, sizeof(event_rec))) return -EFAULT; - spin_lock_irqsave(&lock,flags); play_event(event_rec.arr); - spin_unlock_irqrestore(&lock,flags); return 0; case SNDCTL_MIDI_INFO: -- 1.9.1

2 1

Re: [alsa-devel] Lost Interrput on VIA VT1708S
by Raymond Yau 17 Apr '15

17 Apr '15

> > The second opint is what you have an issue with - see "Lost interrupts" in comment #28. > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/c… hwptr tracking was replaced by tracepoint This seem to be the reason that you only find those Lost Interrput in the system log https://bugs.freedesktop.org/show_bug.cgi?id=49608#c28 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/cor…

1 0