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Code Issues Releases Wiki Activity

Improved audio drift correction support.

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This commit is contained in:
Johns
2012-02-29 16:35:49 +01:00
parent 144f22314f
commit 43b48224b5
3 changed files with 78 additions and 83 deletions
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112
codec.c
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@@ -609,13 +609,12 @@ struct _audio_decoder_
ReSampleContext *ReSample; ///< audio resampling context
int64_t StartPTS; ///< start PTS
struct timespec StartTime; ///< start time
int64_t LastDelay; ///< last delay
struct timespec LastTime; ///< last time
int64_t LastPTS; ///< last PTS
int Drift; ///< drift correction value
#define AVERAGE 10 ///< number of average values
int Average[AVERAGE]; ///< average for drift calculation
int Drift; ///< accumulated audio drift
int DriftCorr; ///< audio drift correction value
struct AVResampleContext *AvResample; ///< second audio resample context
#define MAX_CHANNELS 8 ///< max number of channels supported
@@ -728,7 +727,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
audio_decoder->Channels = 0;
audio_decoder->HwSampleRate = 0;
audio_decoder->HwChannels = 0;
audio_decoder->StartPTS = AV_NOPTS_VALUE;
audio_decoder->LastDelay = 0;
}
/**
@@ -851,56 +850,75 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
struct timespec nowtime;
int64_t delay;
int64_t tim_diff;
int64_t pts_diff;
int64_t drift;
int corr;
AudioSetClock(pts);
// start drift detection
if (audio_decoder->StartPTS == (int64_t) AV_NOPTS_VALUE && AudioGetDelay()) {
audio_decoder->StartPTS = AudioGetClock();
audio_decoder->LastPTS = audio_decoder->StartPTS;
clock_gettime(CLOCK_REALTIME, &audio_decoder->StartTime);
delay = AudioGetDelay();
if (!delay) {
return;
}
pts = AudioGetClock();
clock_gettime(CLOCK_REALTIME, &nowtime);
pts_diff = pts - audio_decoder->StartPTS;
tim_diff = (nowtime.tv_sec - audio_decoder->StartTime.tv_sec)
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
audio_decoder->StartTime.tv_nsec);
drift = pts_diff * 1000 * 1000 / 90 - tim_diff;
if (abs(drift) > 100 * 1000 * 1000) {
// drift too big, pts changed?
audio_decoder->StartPTS = pts;
audio_decoder->LastPTS = audio_decoder->StartPTS;
audio_decoder->StartTime = nowtime;
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
Debug(3, "codec/audio: inital delay %zd ms\n", delay / 90);
return;
}
// collect over some time
if (pts - audio_decoder->LastPTS < 10 * 1000 * 90) {
pts_diff = pts - audio_decoder->LastPTS;
if (pts_diff < 10 * 1000 * 90) {
return;
}
tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
* 1000 * 1000 * 1000 + (nowtime.tv_nsec -
audio_decoder->LastTime.tv_nsec);
drift =
(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
audio_decoder->LastDelay;
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift +=
(int)((10 * audio_decoder->SampleRate * drift) / tim_diff);
if (audio_decoder->AvResample) {
av_resample_compensate(audio_decoder->AvResample, audio_decoder->Drift,
10 * audio_decoder->SampleRate);
if (1) {
Debug(3, "codec/audio: interval P:%5zdms T:%5zdms D:%4zdms %f %d\n",
pts_diff / 90, tim_diff / (1000 * 1000), delay / 90, drift / 90.0,
audio_decoder->DriftCorr);
}
Info("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %g\n",
audio_decoder->Drift, drift / (1000 * 1000), drift,
(double)drift / tim_diff);
printf("codec/audio: drift(%3d) %3" PRId64 "ms %8" PRId64 " %d\n",
audio_decoder->Drift, drift / (1000 * 1000), drift,
(int)((10 * audio_decoder->SampleRate * drift) / tim_diff));
if (abs(drift) > 5 * 90) {
// drift too big, pts changed?
Debug(3, "codec/audio: drift(%5d) %3" PRId64 "ms reset\n",
audio_decoder->DriftCorr, drift);
audio_decoder->LastDelay = 0;
return;
}
drift += audio_decoder->Drift;
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
audio_decoder->DriftCorr -= corr;
if (audio_decoder->DriftCorr < -20000) { // limit correction
audio_decoder->DriftCorr = -20000;
} else if (audio_decoder->DriftCorr > 20000) {
audio_decoder->DriftCorr = 20000;
}
if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
av_resample_compensate(audio_decoder->AvResample,
audio_decoder->DriftCorr / 10, 10 * audio_decoder->HwSampleRate);
}
printf("codec/audio: drift(%5d) %8" PRId64 "us %4d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
/**
@@ -977,6 +995,12 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
if (!audio_decoder->AvResample) {
Error(_("codec/audio: AvResample setup error\n"));
} else {
// reset drift to some default value
audio_decoder->DriftCorr /= 2;
av_resample_compensate(audio_decoder->AvResample,
audio_decoder->DriftCorr / 10,
10 * audio_decoder->HwSampleRate);
}
}
}
@@ -1053,18 +1077,6 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
}
n *= 2;
#if 0
// FIXME: must split channels, filter, join channels
n = av_resample(audio_decoder->AvResample, buf, data, &consumed, count,
sizeof(buf), 1);
if (n < 0) {
Error(_("codec/audio: can't av_resample\n"));
return;
}
if (consumed != count) {
Error(_("codec/audio: av_resample didn't consume all samples\n"));
}
#endif
n *= audio_decoder->HwChannels;
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
AudioEnqueue(buf, n);