does it mean that avail_min cannot be larger than buffer size ?
Is this a bug of snd_pcm_sw_params_set_avail_min() ?
PA server set avail_min to 4661 which is even larger than buffer size 2048 when use 2 periods of 4K bytes with au8830
avail will never greater than runtime->control->avail_min (4661)
However au8830 work quite well on Fedora 10 pulseaudio-0.9.14
D: alsa-util.c: buffer_size : 2048 D: alsa-util.c: period_size : 1024 D: alsa-util.c: period_time : 23219 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 4661
2009/12/10 pl bossart bossart.nospam@gmail.com
- why PA use snd_pcm_hw_params_get_buffer_size_max() instead of
snd_pcm_hw_params_get_buffer_size() after snd_pcm_hw_params() ?
Precisely to use the maximum preallocated buffer size.
D: alsa-util.c: Maximum hw buffer size is 371 ms I: module-alsa-sink.c: Successfully opened device front:0. I: module-alsa-sink.c: Successfully enabled mmap() mode. I: module-alsa-sink.c: Successfully enabled timer-based scheduling mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Using mixer control "Master". I: sink.c: Created sink 0 "alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: Created source 0 "alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: module-alsa-sink.c: Using 2 fragments of size 4096 bytes, buffer time is 46.44ms I: module-alsa-sink.c: Time scheduling watermark is 20.00ms D: module-alsa-sink.c: hwbuf_unused_frames=0 D: module-alsa-sink.c: setting avail_min=4661 I: module-alsa-sink.c: Volume ranges from 0 to 31. I: module-alsa-sink.c: Volume ranges from -46.50 dB to 0.00 dB. I: alsa-util.c: All 2 channels can be mapped to mixer channels. I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 0 'Aureal Vortex au8830' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 2048 D: alsa-util.c: period_size : 1024 D: alsa-util.c: period_time : 23219 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 4661 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : -1 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 1073741824 D: alsa-util.c: appl_ptr : 0 D: alsa-util.c: hw_ptr : 0
2009/12/24 Jaroslav Kysela perex@perex.cz
On Wed, 23 Dec 2009, pl bossart wrote:
Thanks to Takashi's advice, I managed to find out the reason why I was seeing null events returned by poll(). This could explain why PulseAudio doesn't seem to sleep much. It turns out that we have two calls to wakeup() in pcm_lib.c, and a nice race condition it seems. See the log below.
A wake-up is generated during the period interrupt, and a second wake-up is generated during the write loop, after the application was awaken but just before the pointers are updated. This second wake-up shouldn't exist, since the write loop actually fills the ring buffer. By the time the second wake-up is actually handled, there's really no space left in the buffer and a null event is generated; it'll wake-up the application a second time for nothing. Maybe we should move the call to snd_pcm_update_hw_ptr() after the transfer took place?
The right fix should be to preserve wakeups when write operation is in progress (also for interrupts). Something like this (untested):
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a7..8112834 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -272,6 +272,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */
unsigned int nowake: 1; /* do not wakeup */ /* -- HW params -- */ snd_pcm_access_t access; /* access mode */
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f4108..26cf3ff 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -208,7 +208,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } }
if (avail >= runtime->control->avail_min)
if (!runtime->nowake && avail >= runtime->control->avail_min) wake_up(&runtime->sleep); return 0;
} @@ -1776,6 +1776,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; }
runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail;
@@ -1786,17 +1787,18 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (!avail) { if (nonblock) { err = -EAGAIN;
goto _end_unlock;
goto _end_wake; } err = wait_for_avail_min(substream, &avail); if (err < 0)
goto _end_unlock;
goto _end_wake; } frames = size > avail ? avail : size; cont = runtime->buffer_size - runtime->control->appl_ptr %
runtime->buffer_size; if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) {
runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; }
@@ -1809,10 +1811,10 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE;
goto _end_unlock;
goto _end; case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE;
goto _end_unlock;
goto _end; default: break; }
@@ -1830,12 +1832,18 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) { err = snd_pcm_start(substream); if (err < 0)
goto _end_unlock;
goto _end_wake; } }
- _end_wake:
runtime->nowake = 0;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
_end_unlock: snd_pcm_stream_unlock_irq(substream);snd_pcm_update_hw_ptr_post(substream, runtime);
_end:return xfer > 0 ? (snd_pcm_sframes_t)xfer : err;
runtime->nowake = 0; return xfer > 0 ? (snd_pcm_sframes_t)xfer : err;
}
Jaroslav Kysela perex@perex.cz Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.