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vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Support upsampling 3,5 to 8 channels. 

Improve next ring buffer handling.
Improve audio/video sync debug.
This commit is contained in:
Johns 2013-01-25 17:05:53 +01:00
parent d5e111238d
commit 7ed6975330
1 changed files with 30 additions and 9 deletions
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39
audio.c
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@ -546,6 +546,13 @@ static void AudioUpmix(const int16_t * in, int in_chan, int frames,
/**
** Resample ffmpeg sample format to hardware format.
**
** FIXME: use libswresample for this and move it to codec.
** FIXME: ffmpeg to alsa conversion is already done in codec.c.
**
** ffmpeg L R C Ls Rs -> alsa L R Ls Rs C
** ffmpeg L R C LFE Ls Rs -> alsa L R Ls Rs C LFE
** ffmpeg L R C LFE Ls Rs Rl Rr -> alsa L R Ls Rs C LFE Rl Rr
**
** @param in input sample buffer
** @param in_chan nr. of input channels
** @param frames number of frames in sample buffer
@ -581,6 +588,9 @@ static void AudioResample(const int16_t * in, int in_chan, int frames,
AudioSurround2Stereo(in, in_chan, frames, out);
break;
case 5 * 8 + 6:
case 3 * 8 + 8:
case 5 * 8 + 8:
case 6 * 8 + 8:
AudioUpmix(in, in_chan, frames, out, out_chan);
break;
@ -1956,6 +1966,7 @@ static int AudioNextRing(void)
int use_ac3;
int sample_rate;
int channels;
size_t used;
// update audio format
// not always needed, but check if needed is too complex
@ -1975,12 +1986,18 @@ static int AudioNextRing(void)
AudioResetCompressor();
AudioResetNormalizer();
Debug(3, "audio: a/v next buf(%d,%4zdms)\n", atomic_read(&AudioRingFilled),
(RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer) * 1000)
/ (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample));
// stop, if not enough in next buffer
if (AudioStartThreshold >=
RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer)) {
return 1;
used = RingBufferUsedBytes(AudioRing[AudioRingRead].RingBuffer);
if (AudioStartThreshold * 4 < used || (AudioVideoIsReady
&& AudioStartThreshold < used)) {
return 0;
}
return 0;
return 1;
}
/**
@ -2029,10 +2046,13 @@ static void *AudioPlayHandlerThread(void *dummy)
}
if (flush) {
Debug(3, "audio: flush\n");
AudioUsedModule->FlushBuffers();
if (AudioNextRing()) {
Debug(3, "audio: break after flush\n");
break;
}
Debug(3, "audio: continue after flush\n");
}
// try to play some samples
err = AudioUsedModule->Thread();
@ -2275,11 +2295,12 @@ void AudioVideoReady(int64_t pts)
(used * 90 * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample);
Debug(3, "audio: a/v buf:%4zdms %s|%s = %dms video ready\n",
Debug(3, "audio: a/v sync buf(%d,%4zdms) %s|%s = %dms %s\n",
atomic_read(&AudioRingFilled),
(used * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels * AudioBytesProSample),
Timestamp2String(audio_pts), Timestamp2String(pts),
(int)(pts - audio_pts) / 90);
Timestamp2String(pts), Timestamp2String(audio_pts),
(int)(pts - audio_pts) / 90, AudioRunning ? "running" : "ready");
if (!AudioRunning) {
int skip;
@ -2290,7 +2311,7 @@ void AudioVideoReady(int64_t pts)
pts - 15 * 20 * 90 - AudioBufferTime * 90 - audio_pts +
VideoAudioDelay;
#ifdef DEBUG
printf("%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
fprintf(stderr, "%dms %dms %dms\n", (int)(pts - audio_pts) / 90,
VideoAudioDelay / 90, skip / 90);
#endif
// guard against old PTS
@ -2303,7 +2324,7 @@ void AudioVideoReady(int64_t pts)
AudioSkip = skip - used;
skip = used;
}
Debug(3, "audio: advance %dms %d/%zd\n",
Debug(3, "audio: sync advance %dms %d/%zd\n",
(skip * 1000) / (AudioRing[AudioRingWrite].HwSampleRate *
AudioRing[AudioRingWrite].HwChannels *
AudioBytesProSample), skip, used);