Improve timestamp calculation leading to better A/V Sync (ALSA and Pu…

…lse only so far) GetBufferedOnSoundCard should return the length of audio in the buffer including latency ; it was only returning the latency. Also in pulse, is some cases the end calculation could overflow leading to incorrect value. Changed it to 64 bits int
MythTV · Dec 16, 2010 · 5eac843 · 5eac843
1 parent 3fbe948
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Showing 3 changed files with 55 additions and 14 deletions.
diff --git a/mythtv/libs/libmyth/audio/audiooutputalsa.cpp b/mythtv/libs/libmyth/audio/audiooutputalsa.cpp
@@ -562,22 +562,43 @@ void AudioOutputALSA::WriteAudio(uchar *aubuf, int size)
 
 int AudioOutputALSA::GetBufferedOnSoundcard(void) const
 {
+
     if (pcm_handle == NULL)
     {
         VBERROR("getBufferedOnSoundcard() called with pcm_handle == NULL!");
         return 0;
     }
 
-    snd_pcm_sframes_t delay = 0;
+    snd_pcm_sframes_t delay = 0, buffered = 0;
+
+    if (snd_pcm_avail_delay(pcm_handle, &buffered, &delay) < 0)
+    {
+        return 0;
+    }
 
     snd_pcm_state_t state = snd_pcm_state(pcm_handle);
+
     if (state == SND_PCM_STATE_RUNNING || state == SND_PCM_STATE_DRAINING)
-        snd_pcm_delay(pcm_handle, &delay);
+    {
+        delay *= output_bytes_per_frame;
+    }
+    else
+    {
+        delay = 00;
+    }
 
-    if (delay <= 0)
-        return 0;
+    if (buffered < 0)
+    {
+        buffered = 0;
+    }
+
+    buffered *= output_bytes_per_frame;
+    if (buffered > soundcard_buffer_size)
+    {
+        buffered = soundcard_buffer_size;
+    }
 
-    return delay * output_bytes_per_frame;
+    return delay + buffered;
 }
 
 /*

diff --git a/mythtv/libs/libmyth/audio/audiooutputbase.h b/mythtv/libs/libmyth/audio/audiooutputbase.h
@@ -111,6 +111,10 @@ class AudioOutputBase : public AudioOutput, public QThread
     virtual bool OpenDevice(void) = 0;
     virtual void CloseDevice(void) = 0;
     virtual void WriteAudio(uchar *aubuf, int size) = 0;
+    /*
+     * Return the size in bytes of frames currently in the audio buffer adjusted
+     * with the audio playback latency
+     */
     virtual int  GetBufferedOnSoundcard(void) const = 0;
     // Default implementation only supports 2ch s16le at 48kHz
     virtual AudioOutputSettings* GetOutputSettings(void)
@@ -122,7 +126,8 @@ class AudioOutputBase : public AudioOutput, public QThread
     virtual bool StartOutputThread(void);
     virtual void StopOutputThread(void);
 
-    int GetAudioData(uchar *buffer, int buf_size, bool fill_buffer, int *local_raud = NULL);
+    int GetAudioData(uchar *buffer, int buf_size, bool fill_buffer,
+                     int *local_raud = NULL);
 
     void OutputAudioLoop(void);
 

diff --git a/mythtv/libs/libmyth/audio/audiooutputpulse.cpp b/mythtv/libs/libmyth/audio/audiooutputpulse.cpp
@@ -226,10 +226,6 @@ void AudioOutputPulseAudio::WriteAudio(uchar *aubuf, int size)
     QString fn_log_tag = "WriteAudio, ";
     pa_stream_state_t sstate = pa_stream_get_state(pstream);
 
-    // Do not write anything to pulse server if we are in pause mode
-    if (IsPaused())
-        return;
-
     VBAUDIOTS(fn_log_tag + QString("writing %1 bytes").arg(size));
 
     /* NB This "if" check can be replaced with PA_STREAM_IS_GOOD() in
@@ -299,21 +295,40 @@ void AudioOutputPulseAudio::WriteAudio(uchar *aubuf, int size)
 
 int AudioOutputPulseAudio::GetBufferedOnSoundcard(void) const
 {
-    pa_usec_t latency;
+    pa_usec_t latency = (pa_usec_t) -1;
+    size_t buffered = 0;
 
     if (!pcontext || pa_context_get_state(pcontext) != PA_CONTEXT_READY)
         return 0;
 
     if (!pstream || pa_stream_get_state(pstream) != PA_STREAM_READY)
         return 0;
 
+    const pa_buffer_attr *buf_attr =  pa_stream_get_buffer_attr(pstream);
+    size_t bfree = pa_stream_writable_size(pstream);
+    buffered = buf_attr->tlength - bfree;
+
     pa_threaded_mainloop_lock(mainloop);
 
-    if(pa_stream_get_latency(pstream, &latency, NULL) < 0)
-        latency = 0;
+    while (pa_stream_get_latency(pstream, &latency, NULL) < 0)
+    {
+        if (pa_context_errno(pcontext) != PA_ERR_NODATA)
+        {
+            latency = 0;
+            break;
+        }
+        pa_threaded_mainloop_wait(mainloop);
+    }
 
     pa_threaded_mainloop_unlock(mainloop);
-    return (int)latency * samplerate * output_bytes_per_frame / 1000000;
+
+    if (latency < 0)
+    {
+        latency = 0;
+    }
+
+    return ((uint64_t)latency * samplerate *
+            output_bytes_per_frame / 1000000) + buffered;
 }
 
 int AudioOutputPulseAudio::GetVolumeChannel(int channel) const