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https://projects.vdr-developer.org/git/vdr-plugin-softhddevice.git
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Experimental ac3 audio drift correction support.
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Johns
parent
dda9011abc
commit
7e1a42f7ed
@ -1,6 +1,7 @@
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User johns
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Date:
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Experimental ac3 audio drift correction support.
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Removes LPCM detection from TS parser.
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Rewrote video/audio start code.
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Add support for attach/detach plugin.
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92
codec.c
92
codec.c
@ -352,7 +352,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
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{
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AVCodec *video_codec;
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Debug(3, "codec: using codec %s or ID %#04x\n", name, codec_id);
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Debug(3, "codec: using video codec %s or ID %#06x\n", name, codec_id);
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if (decoder->VideoCtx) {
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Error(_("codec: missing close\n"));
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@ -377,7 +377,7 @@ void CodecVideoOpen(VideoDecoder * decoder, const char *name, int codec_id)
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if (name && (video_codec = avcodec_find_decoder_by_name(name))) {
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Debug(3, "codec: vdpau decoder found\n");
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} else if (!(video_codec = avcodec_find_decoder(codec_id))) {
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Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
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Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
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// FIXME: none fatal
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}
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decoder->VideoCodec = video_codec;
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@ -615,6 +615,7 @@ struct _audio_decoder_
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int Drift; ///< accumulated audio drift
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int DriftCorr; ///< audio drift correction value
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int DriftFrac; ///< audio drift fraction for ac3
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struct AVResampleContext *AvResample; ///< second audio resample context
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#define MAX_CHANNELS 8 ///< max number of channels supported
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@ -675,10 +676,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
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{
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AVCodec *audio_codec;
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Debug(3, "codec: using audio codec %s or ID %#06x\n", name, codec_id);
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if (name && (audio_codec = avcodec_find_decoder_by_name(name))) {
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Debug(3, "codec: audio decoder '%s' found\n", name);
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} else if (!(audio_codec = avcodec_find_decoder(codec_id))) {
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Fatal(_("codec: codec ID %#04x not found\n"), codec_id);
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Fatal(_("codec: codec ID %#06x not found\n"), codec_id);
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// FIXME: errors aren't fatal
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}
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audio_decoder->AudioCodec = audio_codec;
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@ -846,6 +849,7 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
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** Set/update audio pts clock.
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**
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** @param audio_decoder audio decoder data
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** @param pts presentation timestamp
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*/
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static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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{
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@ -868,6 +872,7 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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audio_decoder->LastPTS = pts;
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audio_decoder->LastDelay = delay;
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audio_decoder->Drift = 0;
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audio_decoder->DriftFrac = 0;
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Debug(3, "codec/audio: inital delay %zd ms\n", delay / 90);
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return;
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}
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@ -885,46 +890,52 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
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(tim_diff * 90) / (1000 * 1000) - pts_diff + delay -
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audio_decoder->LastDelay;
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// adjust rounding error
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nowtime.tv_nsec -= nowtime.tv_nsec % (1000 * 1000 / 90);
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audio_decoder->LastTime = nowtime;
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audio_decoder->LastPTS = pts;
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audio_decoder->LastDelay = delay;
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if (1) {
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if (0) {
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Debug(3, "codec/audio: interval P:%5zdms T:%5zdms D:%4zdms %f %d\n",
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pts_diff / 90, tim_diff / (1000 * 1000), delay / 90, drift / 90.0,
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audio_decoder->DriftCorr);
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}
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if (abs(drift) > 5 * 90) {
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// underruns and av_resample have the same time :(((
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if (abs(drift) > 10 * 90) {
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// drift too big, pts changed?
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Debug(3, "codec/audio: drift(%5d) %3dms reset\n",
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audio_decoder->DriftCorr, drift);
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Debug(3, "codec/audio: drift(%6d) %3dms reset\n",
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audio_decoder->DriftCorr, drift / 90);
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audio_decoder->LastDelay = 0;
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return;
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}
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} else {
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drift += audio_decoder->Drift;
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audio_decoder->Drift = drift;
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corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
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#ifdef USE_PASSTHROUGH
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// SPDIF/HDMI passthrough
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if (!CodecPassthroughAC3
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|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
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drift += audio_decoder->Drift;
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audio_decoder->Drift = drift;
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corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
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#if defined(USE_PASSTHROUGH) && !defined(USE_AC3_DRIFT_CORRECTION)
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// SPDIF/HDMI passthrough
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if (!CodecPassthroughAC3
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|| audio_decoder->AudioCtx->codec_id != CODEC_ID_AC3)
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#endif
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{
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audio_decoder->DriftCorr -= corr;
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}
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{
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audio_decoder->DriftCorr = -corr;
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}
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if (audio_decoder->DriftCorr < -20000) { // limit correction
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audio_decoder->DriftCorr = -20000;
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} else if (audio_decoder->DriftCorr > 20000) {
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audio_decoder->DriftCorr = 20000;
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if (audio_decoder->DriftCorr < -20000) { // limit correction
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audio_decoder->DriftCorr = -20000;
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} else if (audio_decoder->DriftCorr > 20000) {
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audio_decoder->DriftCorr = 20000;
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}
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}
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// FIXME: this works with libav 0.8, and only with >10ms with ffmpeg 0.10
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if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
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int distance;
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distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
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av_resample_compensate(audio_decoder->AvResample,
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audio_decoder->DriftCorr / 10, 10 * audio_decoder->HwSampleRate);
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audio_decoder->DriftCorr / 10, distance);
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}
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printf("codec/audio: drift(%5d) %8dus %4d\n", audio_decoder->DriftCorr,
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Debug(3, "codec/audio: drift(%6d) %8dus %5d\n", audio_decoder->DriftCorr,
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drift * 1000 / 90, corr);
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}
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@ -996,7 +1007,11 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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}
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}
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// prepare audio drift resample
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#ifdef USE_AUDIO_DRIFT_CORRECTION
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if (!isAC3) {
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if (audio_decoder->AvResample) {
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Error(_("codec/audio: overwrite resample\n"));
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}
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audio_decoder->AvResample =
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av_resample_init(audio_decoder->HwSampleRate,
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audio_decoder->HwSampleRate, 16, 10, 0, 0.8);
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@ -1005,11 +1020,13 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
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} else {
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// reset drift to some default value
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audio_decoder->DriftCorr /= 2;
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audio_decoder->DriftFrac = 0;
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av_resample_compensate(audio_decoder->AvResample,
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audio_decoder->DriftCorr / 10,
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10 * audio_decoder->HwSampleRate);
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}
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}
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#endif
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}
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/**
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@ -1063,7 +1080,6 @@ void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
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n = av_resample(audio_decoder->AvResample, buftmp[ch],
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audio_decoder->Buffer[ch], &consumed, bytes_n / 2,
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sizeof(buftmp[ch]) / 2, ch == audio_decoder->HwChannels - 1);
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//printf("remain%d: %d = %d/%d\n", ch, n, consumed, bytes_n /2);
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// fixme remaining channels
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if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
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audio_decoder->RemainSize = bytes_n - consumed * 2;
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@ -1178,6 +1194,27 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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// build SPDIF header and append A52 audio to it
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// avpkt is the original data
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buf_sz = 6144;
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#ifdef USE_AC3_DRIFT_CORRECTION
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if (1) {
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int x;
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x = (audio_decoder->DriftFrac +
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(audio_decoder->DriftCorr * buf_sz)) / (10 *
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audio_decoder->HwSampleRate * 100);
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audio_decoder->DriftFrac =
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(audio_decoder->DriftFrac +
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(audio_decoder->DriftCorr * buf_sz)) % (10 *
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audio_decoder->HwSampleRate * 100);
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x *= audio_decoder->HwChannels * 4;
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if (x < -64) { // limit correction
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x = -64;
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} else if (x > 64) {
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x = 64;
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}
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buf_sz += x;
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}
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#endif
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if (buf_sz < avpkt->size + 8) {
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Error(_
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("codec/audio: decoded data smaller than encoded\n"));
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@ -1193,7 +1230,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
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memset(buf + 4 + avpkt->size / 2, 0, buf_sz - 8 - avpkt->size);
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// don't play with the ac-3 samples
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AudioEnqueue(buf, buf_sz);
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// FIXME: handle drift.
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return;
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}
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#if 0
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