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-rw-r--r--codec.c92
1 files changed, 64 insertions, 28 deletions
diff --git a/codec.c b/codec.c
index 9d497c6..400dd64 100644
--- a/codec.c
+++ b/codec.c
@@ -352,7 +352,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
{
AVCodec *video_codec;
- Debug(3, "codec: using codec %s or ID %#04x\n", name, codec_id);
+ Debug(3, "codec: using video codec %s or ID %#06x\n", name, codec_id);
if (decoder->VideoCtx) {
Error(_("codec: missing close\n"));
@@ -377,7 +377,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
if (name && (video_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: vdpau decoder found\n");
} else if (!(video_codec = avcodec_find_decoder(codec_id))) {
- Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
+ Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
// FIXME: none fatal
}
decoder->VideoCodec = video_codec;
@@ -615,6 +615,7 @@ struct _audio_decoder_
int Drift; ///< accumulated audio drift
int DriftCorr; ///< audio drift correction value
+ int DriftFrac; ///< audio drift fraction for ac3
struct AVResampleContext *AvResample; ///< second audio resample context
#define MAX_CHANNELS 8 ///< max number of channels supported
@@ -675,10 +676,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
{
AVCodec *audio_codec;
+ Debug(3, "codec: using audio codec %s or ID %#06x\n", name, codec_id);
+
if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
Debug(3, "codec: audio decoder '%s' found\n", name);
} else if (!(audio_codec = avcodec_find_decoder(codec_id))) {
- Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
+ Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
// FIXME: errors aren't fatal
}
audio_decoder->AudioCodec = audio_codec;
@@ -846,6 +849,7 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
** Set/update audio pts clock.
**
** @param audio_decoder audio decoder data
+** @param pts presentation timestamp
*/
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
@@ -868,6 +872,7 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
+ audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital delay %zd ms\n", delay / 90);
return;
}
@@ -885,46 +890,52 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
audio_decoder->LastDelay;
+ // adjust rounding error
+ nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
- if (1) {
+ if (0) {
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);
}
-
- if (abs(drift) > 5 * 90) {
+ // underruns and av_resample have the same time :(((
+ if (abs(drift) > 10 * 90) {
// drift too big, pts changed?
- Debug(3, "codec/audio: drift(%5d) %3dms reset\n",
- audio_decoder->DriftCorr, drift);
+ Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
+ audio_decoder->DriftCorr, drift / 90);
audio_decoder->LastDelay = 0;
- return;
- }
+ } else {
- drift += audio_decoder->Drift;
- audio_decoder->Drift = drift;
- corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
-#ifdef USE_PASSTHROUGH
- // SPDIF/HDMI passthrough
- if (!CodecPassthroughAC3
- || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
+ drift += audio_decoder->Drift;
+ audio_decoder->Drift = drift;
+ corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
+#if defined(USE_PASSTHROUGH) && !defined(USE_AC3_DRIFT_CORRECTION)
+ // SPDIF/HDMI passthrough
+ if (!CodecPassthroughAC3
+ || audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
#endif
- {
- audio_decoder->DriftCorr -= corr;
- }
+ {
+ 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->DriftCorr < -20000) { // limit correction
+ audio_decoder->DriftCorr = -20000;
+ } else if (audio_decoder->DriftCorr > 20000) {
+ audio_decoder->DriftCorr = 20000;
+ }
}
+ // FIXME: this works with libav 0.8, and only with >10ms with ffmpeg 0.10
if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
+ int distance;
+
+ distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
av_resample_compensate(audio_decoder->AvResample,
- audio_decoder->DriftCorr / 10, 10 * audio_decoder->HwSampleRate);
+ audio_decoder->DriftCorr / 10, distance);
}
- printf("codec/audio: drift(%5d) %8dus %4d\n", audio_decoder->DriftCorr,
+ Debug(3, "codec/audio: drift(%6d) %8dus %5d\n", audio_decoder->DriftCorr,
drift * 1000 / 90, corr);
}
@@ -996,7 +1007,11 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
}
}
// prepare audio drift resample
+#ifdef USE_AUDIO_DRIFT_CORRECTION
if (!isAC3) {
+ if (audio_decoder->AvResample) {
+ Error(_("codec/audio: overwrite resample\n"));
+ }
audio_decoder->AvResample =
av_resample_init(audio_decoder->HwSampleRate,
audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
@@ -1005,11 +1020,13 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
} else {
// reset drift to some default value
audio_decoder->DriftCorr /= 2;
+ audio_decoder->DriftFrac = 0;
av_resample_compensate(audio_decoder->AvResample,
audio_decoder->DriftCorr / 10,
10 * audio_decoder->HwSampleRate);
}
}
+#endif
}
/**
@@ -1063,7 +1080,6 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
n = av_resample(audio_decoder->AvResample, buftmp[ch],
audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
- //printf("remain%d: %d = %d/%d\n", ch, n, consumed, bytes_n /2);
// fixme remaining channels
if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
audio_decoder->RemainSize = bytes_n - consumed * 2;
@@ -1178,6 +1194,27 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
// build SPDIF header and append A52 audio to it
// avpkt is the original data
buf_sz = 6144;
+
+#ifdef USE_AC3_DRIFT_CORRECTION
+ if (1) {
+ int x;
+
+ x = (audio_decoder->DriftFrac +
+ (audio_decoder->DriftCorr * buf_sz)) / (10 *
+ audio_decoder->HwSampleRate * 100);
+ audio_decoder->DriftFrac =
+ (audio_decoder->DriftFrac +
+ (audio_decoder->DriftCorr * buf_sz)) % (10 *
+ audio_decoder->HwSampleRate * 100);
+ x *= audio_decoder->HwChannels * 4;
+ if (x < -64) { // limit correction
+ x = -64;
+ } else if (x > 64) {
+ x = 64;
+ }
+ buf_sz += x;
+ }
+#endif
if (buf_sz < avpkt->size + 8) {
Error(_
("codec/audio: decoded data smaller than encoded\n"));
@@ -1193,7 +1230,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
// don't play with the ac-3 samples
AudioEnqueue(buf, buf_sz);
- // FIXME: handle drift.
return;
}
#if 0
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-14 16:46:56 +0000