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pulseaudio: process 1/4 buffer max at once 

Limit the size of data pieces processed by the pulseaudio worker
threads.  Never ever process more than 1/4 of the buffer at once.

Background: The buffer area currently processed by the pulseaudio thread
is blocked, i.e. the main thread (or iothread) can't fill in more data
there.  The buffer processing time is roughly real-time due to the
pa_simple_write() call blocking when the output queue to the pulse
server is full.  Thus processing big chunks at once means blocking
a large part of the buffer for a long time.  This brings high latency
and can lead to dropouts.

When processing the buffer in smaller chunks the rpos handling becomes a
problem though.  The thread reads hw->rpos without knowing whenever
qpa_run_out has already seen the last (small) chunk processed and
updated rpos accordingly.  There is no point in reading hw->rpos though,
pa->rpos can be used instead.  We just need to take care to initialize
pa->rpos before kicking the thread.

Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
Signed-off-by: malc <av1474@comtv.ru>
This commit is contained in:
Gerd Hoffmann 2011-01-24 22:07:44 +01:00 committed by malc
parent d00b261816
commit 6315633b25
1 changed files with 9 additions and 13 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

22
audio/paaudio.c
View File

@ -57,9 +57,6 @@ static void *qpa_thread_out (void *arg)
{
PAVoiceOut *pa = arg;
HWVoiceOut *hw = &pa->hw;
int threshold;
threshold = conf.divisor ? hw->samples / conf.divisor : 0;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
@ -73,7 +70,7 @@ static void *qpa_thread_out (void *arg)
goto exit;
}
if (pa->live > threshold) {
if (pa->live > 0) {
break;
}
@ -82,8 +79,8 @@ static void *qpa_thread_out (void *arg)
}
}
decr = to_mix = pa->live;
rpos = hw->rpos;
decr = to_mix = audio_MIN (pa->live, conf.samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
return NULL;
@ -110,8 +107,8 @@ static void *qpa_thread_out (void *arg)
return NULL;
}
pa->live = 0;
pa->rpos = rpos;
pa->live -= decr;
pa->decr += decr;
}
@ -152,9 +149,6 @@ static void *qpa_thread_in (void *arg)
{
PAVoiceIn *pa = arg;
HWVoiceIn *hw = &pa->hw;
int threshold;
threshold = conf.divisor ? hw->samples / conf.divisor : 0;
if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
return NULL;
@ -168,7 +162,7 @@ static void *qpa_thread_in (void *arg)
goto exit;
}
if (pa->dead > threshold) {
if (pa->dead > 0) {
break;
}
@ -177,8 +171,8 @@ static void *qpa_thread_in (void *arg)
}
}
incr = to_grab = pa->dead;
wpos = hw->wpos;
incr = to_grab = audio_MIN (pa->dead, conf.samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
return NULL;
@ -323,6 +317,7 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);
@ -377,6 +372,7 @@ static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = conf.samples;
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
dolog ("Could not allocate buffer (%d bytes)\n",
hw->samples << hw->info.shift);