Patchwork [RESEND,1/3] pulseaudio: process 1/4 buffer max at once

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Submitter Gerd Hoffmann
Date Jan. 24, 2011, 9:07 p.m.
Message ID <1295903266-6458-1-git-send-email-kraxel@redhat.com>
Download mbox | patch
Permalink /patch/80252/
State New
Headers show

Comments

Gerd Hoffmann - Jan. 24, 2011, 9:07 p.m.
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>
---
 audio/paaudio.c |   22 +++++++++-------------
 1 files changed, 9 insertions(+), 13 deletions(-)
malc - Jan. 25, 2011, 4:57 p.m.
On Mon, 24 Jan 2011, Gerd Hoffmann wrote:

> 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.
> 
[..snip...]

Thanks, applied.

Patch

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 9cf685d..858ca81 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -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);