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vdr-plugin-softhddevice
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Code Issues Releases Wiki Activity

Improved audio drift correction support.

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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1
ChangeLog
View File

@ -1,6 +1,7 @@
User johns
Date:
Improved audio drift correction support.
Experimental audio drift correction support.
Add SVDRP HOTK command support.
Increased audio buffer time for PES packets.

48
audio.c
View File

@ -140,8 +140,6 @@ static volatile char AudioPaused; ///< audio paused
static unsigned AudioSampleRate; ///< audio sample rate in hz
static unsigned AudioChannels; ///< number of audio channels
static const int AudioBytesProSample = 2; ///< number of bytes per sample
static uint32_t AudioTicks; ///< audio ticks
static uint32_t AudioTickPTS; ///< audio pts of tick
static int64_t AudioPTS; ///< audio pts clock
static int AudioBufferTime = 336; ///< audio buffer time in ms
@ -294,7 +292,7 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
Debug(3, "audio/alsa: start %zd ms %d\n",
Debug(3, "audio/alsa: start %4zd ms %d v-buf\n",
(RingBufferUsedBytes(AlsaRingBuffer) * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample),
VideoGetBuffers());
@ -333,7 +331,8 @@ static int AlsaPlayRingbuffer(void)
if (n == -EAGAIN) {
continue;
}
Error(_("audio/alsa: underrun error?\n"));
Error(_("audio/alsa: avail underrun error? '%s'\n"),
snd_strerror(n));
err = snd_pcm_recover(AlsaPCMHandle, n, 0);
if (err >= 0) {
continue;
@ -401,7 +400,8 @@ static int AlsaPlayRingbuffer(void)
goto again;
}
*/
Error(_("audio/alsa: underrun error?\n"));
Error(_("audio/alsa: writei underrun error? '%s'\n"),
snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
goto again;
@ -433,7 +433,7 @@ static void AlsaFlushBuffers(void)
RingBufferReadAdvance(AlsaRingBuffer,
RingBufferUsedBytes(AlsaRingBuffer));
state = snd_pcm_state(AlsaPCMHandle);
Debug(3, "audio/alsa: state %d - %s\n", state,
Debug(3, "audio/alsa: flush state %d - %s\n", state,
snd_pcm_state_name(state));
if (state != SND_PCM_STATE_OPEN) {
if ((err = snd_pcm_drop(AlsaPCMHandle)) < 0) {
@ -446,7 +446,6 @@ static void AlsaFlushBuffers(void)
}
}
AudioRunning = 0;
AudioTicks = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@ -651,14 +650,15 @@ static void AlsaThread(void)
break;
}
// wait for space in kernel buffers
if ((err = snd_pcm_wait(AlsaPCMHandle, 100)) < 0) {
Error(_("audio/alsa: wait underrun error?\n"));
if ((err = snd_pcm_wait(AlsaPCMHandle, 24)) < 0) {
Error(_("audio/alsa: wait underrun error? '%s'\n"),
snd_strerror(err));
err = snd_pcm_recover(AlsaPCMHandle, err, 0);
if (err >= 0) {
continue;
}
Error(_("audio/alsa: snd_pcm_wait(): %s\n"), snd_strerror(err));
usleep(100 * 1000);
usleep(24 * 1000);
continue;
}
if (AlsaFlushBuffer || AudioPaused) {
@ -888,6 +888,9 @@ static uint64_t AlsaGetDelay(void)
if (!AlsaPCMHandle || !AudioSampleRate) {
return 0UL;
}
if (!AudioRunning) { // audio not running
return 0UL;
}
// FIXME: thread safe? __assert_fail_base in snd_pcm_delay
// delay in frames in alsa + kernel buffers
@ -1303,7 +1306,7 @@ static int OssAddToRingbuffer(const void *samples, int count)
}
if (!AudioRunning) {
Debug(3, "audio/oss: start %zd ms %d\n",
Debug(3, "audio/oss: start %4zd ms %d v-buf\n",
(RingBufferUsedBytes(OssRingBuffer) * 1000)
/ (AudioSampleRate * AudioChannels * AudioBytesProSample),
VideoGetBuffers());
@ -1387,7 +1390,6 @@ static void OssFlushBuffers(void)
}
}
AudioRunning = 0;
AudioTicks = 0;
AudioPTS = INT64_C(0x8000000000000000);
}
@ -1672,8 +1674,7 @@ static uint64_t OssGetDelay(void)
if (OssPcmFildes == -1) { // setup failure
return 0UL;
}
if (!AudioRunning) {
if (!AudioRunning) { // audio not running
return 0UL;
}
// delay in bytes in kernel buffers
@ -2185,25 +2186,6 @@ int64_t AudioGetClock(void)
int64_t delay;
if ((delay = AudioGetDelay())) {
#if 0
int64_t pts;
uint32_t ticks;
pts = AudioPTS - delay;
ticks = GetMsTicks();
if (AudioTicks) {
static int64_t drift;
drift += ((pts - AudioTickPTS) - (ticks - AudioTicks) * 90);
drift /= 2;
if (abs(drift) > 90) {
printf("audio-drift: %d\n", (int)drift);
}
}
AudioTicks = ticks;
AudioTickPTS = pts;
return pts;
#endif
return AudioPTS - delay;
}
}

112
codec.c
View File

@ -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);