August 2011 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Daniel Mack 26 Aug '11

26 Aug '11

On Aug 26, 2011 3:35 PM, "Grant Diffey" <gdiffey(a)gmail.com> wrote: > > So on my freshly built 3.1-rc1 this applied cleanly but I'm getting errors every time I try to open the device. on the FTU. > > attempting to start jack gives this in dmesg: > > [ 2091.951806] stop_endpoints() > [ 2197.403224] Re-using EP 81 @ffff880003560000 > [ 2197.403243] attr 4 > [ 2197.403929] stop_endpoints() > [ 2197.403931] -- data ffff880003560000 > [ 2197.403933] -- sync (null) > [ 2197.403935] Unable to change format on ep #81: already in use > > and hitting play on audacity results in the attached dmesg errors. > > is this expected currently? Can you try to just open either the playback or the capture stream seperately, and not both at the same time? Daniel

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[alsa-devel] [RFC PATCH 1/1] ASoC: soc-core: symmetry checking for each DAIs separately
by Dong Aisheng 26 Aug '11

26 Aug '11

The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dai.h | 3 ++ include/sound/soc.h | 2 - sound/soc/soc-pcm.c | 64 +++++++++++++++++++++++++++++++++------------- 3 files changed, 49 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..844d4ed 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,6 +242,9 @@ struct snd_soc_dai { void *playback_dma_data; void *capture_dma_data; + /* Symmetry data */ + unsigned int rate; + /* parent platform/codec */ union { struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..5449139 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -849,8 +849,6 @@ struct snd_soc_pcm_runtime { unsigned int complete:1; unsigned int dev_registered:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..3f7ded7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -32,33 +32,54 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int race; + unsigned int force_rate; int ret; + race = 0; + force_rate = 0; + if (!codec_dai->driver->symmetric_rates && !cpu_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; + if (codec_dai->active && codec_dai->driver->symmetric_rates || + codec_dai->active && rtd->dai_link->symmetric_rates) { + if (codec_dai->rate != 0) + force_rate = codec_dai->rate; + else + race = 1; + } + + if (cpu_dai->active && cpu_dai->driver->symmetric_rates || + codec_dai->active && rtd->dai_link->symmetric_rates) { + if (cpu_dai->rate != 0) + force_rate = cpu_dai->rate; + else + race = 1; + } + + if (force_rate) { + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + force_rate, force_rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + } + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { + if (race) dev_warn(&rtd->dev, - "Not enforcing symmetric_rates due to race\n"); - return 0; - } - - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } + "Not enforcing symmetric_rates due to race\n"); return 0; } @@ -287,9 +308,14 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->active--; codec_dai->active--; codec->active--; + rtd->active--; + + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +473,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.0.4

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[alsa-devel] Will be on vacation
by Takashi Iwai 26 Aug '11

26 Aug '11

Hi, I'll be on vacation from tomorrow for two weeks. As I'll be in a round trip in Sri Lanka, likely unreachable in most time, and I'm going to catch up things after back from vacation. Or, if anyone wants to take a maintenance of 3.1/3.2 sound git trees during my absence, feel free to do that. thanks, Takashi

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[alsa-devel] [PATCH v2 1/1] ASoC: soc-core: check rate for symmetry only when pcm is active
by Dong Aisheng 26 Aug '11

26 Aug '11

For the playback and record using different dai links, checking !rtd->rate for symmetry may not be accurate because that pcm may be acutually not running and the default new open rate is 0, then the warning message "Not enforcing symmetric_rates" will happen each time with running arecord | aplay. Now we only check rate for symmetry when the pcm is really active which seems more sensible. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 5 +++++ 2 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..17970fd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -851,6 +851,7 @@ struct snd_soc_pcm_runtime { /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; + unsigned int active; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..72a7b34 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -39,6 +39,9 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) !rtd->dai_link->symmetric_rates) return 0; + if (!rtd->active) + return 0; + /* This can happen if multiple streams are starting simultaneously - * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. @@ -209,6 +212,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; + rtd->active++; mutex_unlock(&rtd->pcm_mutex); return 0; @@ -287,6 +291,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->active--; codec_dai->active--; codec->active--; + rtd->active--; if (!cpu_dai->active && !codec_dai->active) rtd->rate = 0; -- 1.7.0.4

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[alsa-devel] [PATCH v2 0/2] ASoC: omap-mcpdm: New McPDM driver
by Peter Ujfalusi 26 Aug '11

26 Aug '11

Hello Mark, Liam, I tried to address the comments regarding to version 1 of the new McPDM driver: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/041690.html The omap hwmod conversion part of the v1 series has been sent as part of the fixup series for 3.1 to make basic audio work out of box. This series is aimed for 3.2 due to the scale of the change. I tried to split patch 2, but I did not managed to find an incremental path from the old driver to the new one. I have updated the commit message for patch 2 to explain the reasoning behind the rewritten driver, so it can be easily looked up. Notable changes since v1: - delayed_work from the new driver has been removed, and replaced with a DAPM_SUPPLY widget. This widget need to be connected to DAC/ADC of the codec by the machine driver in order to have the correct sequence for power off. The first patch adds private_data pointer for dapm widgets. This private_data can be used to add widget specific data to any widget. I think this can be handy for others as well in the future. - Full duplex audio in legacy mode has been fixed, so we can run arecord | aplay This series depends on the McPDM fixup patches: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042401.html Peter PS: to make the review easier, I have asked git format-patch to treat the change as rewrite. --- Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (1): ASoC: DAPM: Add private data pointer for DAPM widget include/sound/soc-dapm.h | 1 + sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 472 ------------------------- sound/soc/omap/mcpdm.h | 152 -------- sound/soc/omap/omap-mcpdm.c | 823 ++++++++++++++++++++++++++++--------------- sound/soc/omap/omap-mcpdm.h | 97 +++++ sound/soc/omap/sdp4430.c | 14 +- 7 files changed, 656 insertions(+), 905 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h rewrite sound/soc/omap/omap-mcpdm.c (46%) create mode 100644 sound/soc/omap/omap-mcpdm.h -- 1.7.6

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[alsa-devel] ALSA calls don't work from a shared library
by Olivier Guillion - Myriad 25 Aug '11

25 Aug '11

Hi, I opened a thread yesterday called "ALSA calls don't work within Mozilla plug- ins", but after more testing, I realize that it is a more global problem, so I submit another request for help with a proper title. It seems that several ALSA functions either crash or return unpredictable values when called from a dynamic library (.so). I created a very short program that opens a pcm device, sets the sample rate to 44100 and reads the sample rate back to ensure it's correct : --------------- #include "alsa/asoundlib.h" static char displayString[1024]; char * ALSATest(void) { int err; snd_pcm_t * handle; snd_pcm_hw_params_t *params; unsigned int val,val2; char status[1024]; err = snd_pcm_open(&handle, "plughw:0,0",SND_PCM_STREAM_PLAYBACK, 0); if(err>=0) { snd_pcm_hw_params_alloca(&params); snd_pcm_hw_params_any(handle, params); val=44100; err=snd_pcm_hw_params_set_rate_near(handle,params, &val, NULL); val2=0; err|=snd_pcm_hw_params_get_rate(params, &val2, NULL); if(val!=44100 || val2!=44100 || err!=0) sprintf(status,"ALSA is not working (val2=%d err=%d)",val2,err); else strcpy(status,"ALSA is OK"); snd_pcm_close(handle); } else strcpy(status,"device cannot be opened"); sprintf(displayString,"ALSA %s\nStatus:%s",snd_asoundlib_version(),status); return(displayString); } ------------- If I call this function directly from the main program, it works well (I get "ALSA is OK"). Then if I compile it as a shared library (mysharedlib.so), and in another project I write: --------------- #include <dlfcn.h> #include "alsa/asoundlib.h" typedef char * (*ALSAProc)(void); int main() { void *lib_handle; void *fn; char * str; lib_handle=dlopen("mysharedlib.so",RTLD_NOW); if (lib_handle) { fn=dlsym(lib_handle,"ALSATest"); if(fn) str=((ALSAProc)fn)(); else str="function not found"; } else str="lib not found"; fprintf(stderr,"%s\n",str); if(lib_handle) dlclose(lib_handle); return 0; } ------------- I get unpredictable results. Either; - it crashes in snd_pcm_hw_params_any (in the subroutine snd_pcm_hw_params_refine) or - it outputs "ALSA is not working (val2=0 err=2147483647)" or - it outputs ALSA is not working (val2=2147483647 err=2147483647)" Does anyone have an idea of what could cause this behaviour? Olivier

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Re: [alsa-devel] [PATCH] ALSA: usb: refine delay information with USB frame counter
by Clemens Ladisch 25 Aug '11

25 Aug '11

Pierre-Louis Bossart wrote: > > > + if (frame_diff < 0) > > > + frame_diff += 1024; /* handle 10-bit wrap-around */ > > > > After reading through some HCD source files, I deduced the following > > wrap-arounds: > > > > EHCI: 8-10 bits (default 9) > > FHCI: 11 bits > > IMX21: 16 bits > > ISP1760: 10 bits > > OHCI: 16 bits > > OXU210HP: 10 bits > > R8A66597: 10 bits > > SL811: 5 bits(!?) > > UHCI: 32 bits > > xHCI: ? > > > > As long as there isn't an API that tells us about the valid range of the > > frame counter, we should probably mask off all except the lowest few bits. > > If we only used the 8 least-significant bits, it'd still leave us with up to > 256 ms. That should work I guess. But the SL811 with 5 bits is a problem, > that makes 32ms, this is probably too low. Sorry, I read the code wrong; that driver returns a 16-bit frame counter, it just doesn't allow to schedule more than 32 frame AFAICT. And that driver doesn't support USB audio anyway, so 8 bits should be fine. Regards, Clemens

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[alsa-devel] [PATCH] ALSA: usb: refine delay information with USB frame counter
by Pierre-Louis Bossart 25 Aug '11

25 Aug '11

Existing code only updates the audio delay when URBs were submitted/retired. This can introduce an uncertainty of 8ms on the number of samples played out with the default settings, and a lot more when URBs convey more packets to reduce the interrupt rate and power consumption. This patch relies on the USB frame counter to reduce the uncertainty to ~1ms. The delay information essentially becomes independent of the URB size and number of packets. This should improve audio/video sync and help applications like PulseAudio which require accurate audio timing. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/card.h | 2 ++ sound/usb/pcm.c | 23 +++++++++++++++++++++++ sound/usb/pcm.h | 3 +++ sound/usb/urb.c | 20 +++++++++++++++++--- 4 files changed, 45 insertions(+), 3 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ae4251d..a39edcc 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -94,6 +94,8 @@ struct snd_usb_substream { spinlock_t lock; struct snd_urb_ops ops; /* callbacks (must be filled at init) */ + int last_frame_number; /* stored frame number */ + int last_delay; /* stored delay */ }; struct snd_usb_stream { diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b8dcbf4..1696e2b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -34,6 +34,27 @@ #include "clock.h" #include "power.h" +/* return the estimated delay based on USB frame counters */ +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate) +{ + int current_frame_number; + int frame_diff; + int est_delay; + + current_frame_number = usb_get_current_frame_number(subs->dev); + frame_diff = current_frame_number-subs->last_frame_number; + if (frame_diff < 0) + frame_diff += 1024; /* handle 10-bit wrap-around */ + + /* Approximation based on number of samples per USB frame (ms), + some truncation for 44.1 but the estimate is good enough */ + est_delay = subs->last_delay-(frame_diff*rate/1000L); + if (est_delay < 0) + est_delay = 0; + return est_delay; +} + /* * return the current pcm pointer. just based on the hwptr_done value. */ @@ -45,6 +66,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; + substream->runtime->delay = snd_usb_pcm_delay(subs, + substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index ed3e283..df7a003 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -1,6 +1,9 @@ #ifndef __USBAUDIO_PCM_H #define __USBAUDIO_PCM_H +snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, + unsigned int rate); + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, diff --git a/sound/usb/urb.c b/sound/usb/urb.c index e184349..ccf77d1 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -718,7 +718,15 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -737,12 +745,18 @@ static int retire_playback_urb(struct snd_usb_substream *subs, unsigned long flags; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + spin_unlock_irqrestore(&subs->lock, flags); return 0; } -- 1.7.6

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[alsa-devel] M-Audio FTU remaining mixer controls and external clocking support.
by Aurélien Leblond 25 Aug '11

25 Aug '11

>Yes, sorry for this taking longer than expected. I've been working on >this for a while and will continue doing so next week. I just got too >much things to do in parallel these days. I'll get back to you once I >can show patches for testing. That's a great news, I'm ready any time for testing! Aurélien

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[alsa-devel] [PATCH] ALSA: hda: Conexant: Allow different output types to share DAC
by David Henningsson 25 Aug '11

25 Aug '11

[PATCH] ALSA: hda: Conexant: Allow different output types to share DAC Headphones has stopped working for the original reported (a regression compared to 2.6.38). This is because Speaker and Headphones share the same DAC, in which case no Headphones volume control was created. This patch fixes so that both Speaker and Headphones volume controls are created in such scenario. Buglink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Link to codec-proc: https://launchpadlibrarian.net/76215662/Card0.Codecs.codec.0.txt -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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