frodovdr/vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Video enhancements.

Browse Source 
Pass PTS to video codec.
Flush audio buffer on channel change.
Sync audio + video stream.
Add workaround for Intel VA-API driver that forgets OSD position/size.
This commit is contained in:
Johns
2011-12-10 00:15:38 +01:00
parent e1d8609143
commit 749b0d6721
8 changed files with 271 additions and 221 deletions
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73
audio.c
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@@ -152,7 +152,9 @@ static int AlsaPlayRingbuffer(void)
// happens with broken alsa drivers
Error(_("audio/alsa: broken driver %d\n"), avail);
}
//break;
Debug(4, "audio/alsa: break state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
break;
}
n = RingBufferGetReadPointer(AlsaRingBuffer, &p);
@@ -320,8 +322,7 @@ void AudioEnqueue(const void *samples, int count)
}
#endif
state = snd_pcm_state(AlsaPCMHandle);
Debug(3, "audio/alsa: state %d - %s\n", state,
snd_pcm_state_name(state));
Debug(3, "audio/alsa: state %s\n", snd_pcm_state_name(state));
Debug(3, "audio/alsa: unpaused\n");
AudioPaused = 0;
}
@@ -381,10 +382,15 @@ static void *AudioPlayHandlerThread(void *dummy)
Debug(3, "audio/alsa: flushing buffers\n");
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
if ((err = snd_pcm_drain(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_drain(): %s\n"),
#if 1
if ((err = snd_pcm_drop(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_drop(): %s\n"), snd_strerror(err));
}
if ((err = snd_pcm_prepare(AlsaPCMHandle))) {
Error(_("audio: snd_pcm_prepare(): %s\n"),
snd_strerror(err));
}
#endif
AlsaFlushBuffer = 0;
break;
}
@@ -504,10 +510,11 @@ static void AlsaInitPCM(void)
SND_PCM_NONBLOCK)) < 0) {
Fatal(_("audio/alsa: playback open '%s' error: %s\n"), device,
snd_strerror(err));
// FIXME: no fatal error for plugins!
}
AlsaPCMHandle = handle;
if ((err = snd_pcm_nonblock(handle, SND_PCM_NONBLOCK)) < 0) {
if ((err = snd_pcm_nonblock(handle, 0)) < 0) {
Error(_("audio/alsa: can't set block mode: %s\n"), snd_strerror(err));
}
@@ -610,13 +617,27 @@ static void AlsaInitMixer(void)
void AudioSetClock(int64_t pts)
{
if (AudioPTS != pts) {
Debug(3, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n",
Debug(4, "audio: set clock to %#012" PRIx64 " %#012" PRIx64 " pts\n",
AudioPTS, pts);
AudioPTS = pts;
}
}
/**
** Get current audio clock.
*/
int64_t AudioGetClock(void)
{
int64_t delay;
delay = AudioGetDelay();
if (delay) {
return AudioPTS - delay;
}
return INT64_C(0x8000000000000000);
}
/**
** Get audio delay in time stamps.
*/
@@ -626,17 +647,19 @@ uint64_t AudioGetDelay(void)
snd_pcm_sframes_t delay;
uint64_t pts;
// delay in frames in alsa + kernel buffers
if ((err = snd_pcm_delay(AlsaPCMHandle, &delay)) < 0) {
//Debug(3, "audio/alsa: no hw delay\n");
delay = 0UL;
} else if (snd_pcm_state(AlsaPCMHandle) != SND_PCM_STATE_RUNNING) {
//Debug(3, "audio/alsa: %ld delay ok, but not running\n", delay);
//Debug(3, "audio/alsa: %ld frames delay ok, but not running\n", delay);
}
pts = ((uint64_t) delay * 90000) / AudioSampleRate;
pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90000)
/ (AudioSampleRate * AudioChannels);
//Debug(3, "audio/alsa: hw+sw delay %"PRId64" ms\n", pts / 90);
//Debug(3, "audio/alsa: %ld frames hw delay\n", delay);
pts = ((uint64_t) delay * 90 * 1000) / AudioSampleRate;
pts += ((uint64_t) RingBufferUsedBytes(AlsaRingBuffer) * 90 * 1000)
/ (AudioSampleRate * AudioChannels * 2);
Debug(4, "audio/alsa: hw+sw delay %zd %" PRId64 " ms\n",
RingBufferUsedBytes(AlsaRingBuffer), pts / 90);
return pts;
}
@@ -676,13 +699,18 @@ int AudioSetup(int *freq, int *channels)
// flush any buffered data
#ifdef USE_AUDIO_THREAD
if (AudioRunning) {
AlsaFlushBuffer = 1;
while (AudioRunning) {
AlsaFlushBuffer = 1;
usleep(1 * 1000);
}
AlsaFlushBuffer = 0;
} else
#endif
{
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
}
AudioPTS = INT64_C(0x8000000000000000);
ret = 0;
try_again:
@@ -692,6 +720,15 @@ int AudioSetup(int *freq, int *channels)
SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
125 * 1000))) {
Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));
/*
if ( err == -EBADFD ) {
snd_pcm_close(AlsaPCMHandle);
AlsaPCMHandle = NULL;
goto try_again;
}
*/
switch (*channels) {
case 1:
// FIXME: enable channel upmix
@@ -795,10 +832,10 @@ int AudioSetup(int *freq, int *channels)
Debug(3, "audio/alsa: state %s\n",
snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, buffer_size);
// min 500ms
if (AlsaStartThreshold < (*freq * *channels * 2U) / 2) {
AlsaStartThreshold = (*freq * *channels * 2U) / 2;
AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
// min 333ms
if (AlsaStartThreshold < (*freq * *channels * 2U) / 3) {
AlsaStartThreshold = (*freq * *channels * 2U) / 3;
}
Debug(3, "audio/alsa: delay %u ms\n", (AlsaStartThreshold * 1000)
/ (AudioSampleRate * AudioChannels * 2));