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Fix issue #42 and changes for faster a/v sync

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jojo61
2020-03-31 13:57:43 +02:00
parent fecb81486d
commit a6e65d953e
4 changed files with 83 additions and 526 deletions
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421
codec.c
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@@ -96,7 +96,6 @@ static pthread_mutex_t CodecLockMutex;
/// Flag prefer fast channel switch
char CodecUsePossibleDefectFrames;
AVBufferRef *hw_device_ctx;
//----------------------------------------------------------------------------
// Video
//----------------------------------------------------------------------------
@@ -495,7 +494,7 @@ void DisplayPts(AVCodecContext * video_ctx, AVFrame * frame)
*/
extern int CuvidTestSurfaces();
#ifdef YADIF
#if defined YADIF || defined (VAAPI)
extern int init_filters(AVCodecContext * dec_ctx, void *decoder, AVFrame * frame);
extern int push_filters(AVCodecContext * dec_ctx, void *decoder, AVFrame * frame);
#endif
@@ -599,6 +598,7 @@ void CodecVideoDecode(VideoDecoder * decoder, const AVPacket * avpkt)
}
// printf("got %s packet from decoder\n",got_frame?"1":"no");
if (got_frame) { // frame completed
// printf("video frame pts %#012" PRIx64 " %dms\n",frame->pts,(int)(apts - frame->pts) / 90);
#ifdef YADIF
if (decoder->filter) {
if (decoder->filter == 1) {
@@ -647,7 +647,7 @@ void CodecVideoDecode(VideoDecoder * decoder, const AVPacket * avpkt)
void CodecVideoFlushBuffers(VideoDecoder * decoder)
{
if (decoder->VideoCtx) {
avcodec_flush_buffers(decoder->VideoCtx);
avcodec_flush_buffers(decoder->VideoCtx);
}
}
@@ -679,9 +679,6 @@ struct _audio_decoder_
AVFrame *Frame; ///< decoded audio frame buffer
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
ReSampleContext *ReSample; ///< old resampling context
#endif
#ifdef USE_SWRESAMPLE
#if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100)
struct SwrContext *Resample; ///< ffmpeg software resample context
@@ -704,16 +701,6 @@ struct _audio_decoder_
int Drift; ///< accumulated audio drift
int DriftCorr; ///< audio drift correction value
int DriftFrac; ///< audio drift fraction for ac3
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
struct AVResampleContext *AvResample; ///< second audio resample context
#define MAX_CHANNELS 8 ///< max number of channels supported
int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers
int BufferSize; ///< size of sample buffer
int16_t *Remain[MAX_CHANNELS]; ///< filter remaining samples
int RemainSize; ///< size of remain buffer
int RemainCount; ///< number of remaining samples
#endif
};
///
@@ -832,27 +819,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, int codec_id)
void CodecAudioClose(AudioDecoder * audio_decoder)
{
// FIXME: output any buffered data
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
if (audio_decoder->AvResample) {
int ch;
av_resample_close(audio_decoder->AvResample);
audio_decoder->AvResample = NULL;
audio_decoder->RemainCount = 0;
audio_decoder->BufferSize = 0;
audio_decoder->RemainSize = 0;
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
free(audio_decoder->Buffer[ch]);
audio_decoder->Buffer[ch] = NULL;
free(audio_decoder->Remain[ch]);
audio_decoder->Remain[ch] = NULL;
}
}
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
}
#endif
#ifdef USE_SWRESAMPLE
if (audio_decoder->Resample) {
swr_free(&audio_decoder->Resample);
@@ -1144,385 +1111,7 @@ static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder, const AVPac
return 0;
}
#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE)
/**
** Set/update audio pts clock.
**
** @param audio_decoder audio decoder data
** @param pts presentation timestamp
*/
static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
{
struct timespec nowtime;
int64_t delay;
int64_t tim_diff;
int64_t pts_diff;
int drift;
int corr;
AudioSetClock(pts);
delay = AudioGetDelay();
if (!delay) {
return;
}
clock_gettime(CLOCK_MONOTONIC, &nowtime);
if (!audio_decoder->LastDelay) {
audio_decoder->LastTime = nowtime;
audio_decoder->LastPTS = pts;
audio_decoder->LastDelay = delay;
audio_decoder->Drift = 0;
audio_decoder->DriftFrac = 0;
Debug(3, "codec/audio: inital drift delay %" PRId64 "ms\n", delay / 90);
return;
}
// collect over some time
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;
// 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 (0) {
Debug(3, "codec/audio: interval P:%5" PRId64 "ms T:%5" PRId64 "ms D:%4" PRId64 "ms %f %d\n", pts_diff / 90,
tim_diff / (1000 * 1000), delay / 90, drift / 90.0, audio_decoder->DriftCorr);
}
// underruns and av_resample have the same time :(((
if (abs(drift) > 10 * 90) {
// drift too big, pts changed?
Debug(3, "codec/audio: drift(%6d) %3dms reset\n", audio_decoder->DriftCorr, drift / 90);
audio_decoder->LastDelay = 0;
#ifdef DEBUG
corr = 0; // keep gcc happy
#endif
} else {
drift += audio_decoder->Drift;
audio_decoder->Drift = drift;
corr = (10 * audio_decoder->HwSampleRate * drift) / (90 * 1000);
// SPDIF/HDMI passthrough
if ((CodecAudioDrift & CORRECT_AC3) && (!(CodecPassthrough & CodecAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_AC3)
&& (!(CodecPassthrough & CodecEAC3)
|| audio_decoder->AudioCtx->codec_id != AV_CODEC_ID_EAC3)) {
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;
}
}
// FIXME: this works with libav 0.8, and only with >10ms with ffmpeg 0.10
if (audio_decoder->AvResample && audio_decoder->DriftCorr) {
int distance;
// try workaround for buggy ffmpeg 0.10
if (abs(audio_decoder->DriftCorr) < 2000) {
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
} else {
distance = (pts_diff * audio_decoder->HwSampleRate) / (90 * 1000);
}
av_resample_compensate(audio_decoder->AvResample, audio_decoder->DriftCorr / 10, distance);
}
if (1) {
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n", audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
}
/**
** Handle audio format changes.
**
** @param audio_decoder audio decoder data
**
** @note this is the old not good supported version
*/
static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
{
int passthrough;
const AVCodecContext *audio_ctx;
int err;
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
}
if (audio_decoder->AvResample) {
av_resample_close(audio_decoder->AvResample);
audio_decoder->AvResample = NULL;
audio_decoder->RemainCount = 0;
}
audio_ctx = audio_decoder->AudioCtx;
if ((err = CodecAudioUpdateHelper(audio_decoder, &passthrough))) {
Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d err %d\n", audio_ctx->sample_rate, audio_ctx->channels,
audio_decoder->HwSampleRate, audio_decoder->HwChannels, err);
if (err == 1) {
audio_decoder->ReSample =
av_audio_resample_init(audio_decoder->HwChannels, audio_ctx->channels, audio_decoder->HwSampleRate,
audio_ctx->sample_rate, audio_ctx->sample_fmt, audio_ctx->sample_fmt, 16, 10, 0, 0.8);
// libav-0.8_pre didn't support 6 -> 2 channels
if (!audio_decoder->ReSample) {
Error(_("codec/audio: resample setup error\n"));
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
}
return;
}
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
}
// prepare audio drift resample
#ifdef USE_AUDIO_DRIFT_CORRECTION
if (CodecAudioDrift & CORRECT_PCM) {
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);
if (!audio_decoder->AvResample) {
Error(_("codec/audio: AvResample setup error\n"));
} 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
}
/**
** Codec enqueue audio samples.
**
** @param audio_decoder audio decoder data
** @param data samples data
** @param count number of bytes in sample data
*/
void CodecAudioEnqueue(AudioDecoder * audio_decoder, int16_t * data, int count)
{
#ifdef USE_AUDIO_DRIFT_CORRECTION
if ((CodecAudioDrift & CORRECT_PCM) && audio_decoder->AvResample) {
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 + AV_INPUT_BUFFER_PADDING_SIZE]
__attribute__((aligned(16)));
int16_t buftmp[MAX_CHANNELS][(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4];
int consumed;
int i;
int n;
int ch;
int bytes_n;
bytes_n = count / audio_decoder->HwChannels;
// resize sample buffer, if needed
if (audio_decoder->RemainCount + bytes_n > audio_decoder->BufferSize) {
audio_decoder->BufferSize = audio_decoder->RemainCount + bytes_n;
for (ch = 0; ch < MAX_CHANNELS; ++ch) {
audio_decoder->Buffer[ch] = realloc(audio_decoder->Buffer[ch], audio_decoder->BufferSize);
}
}
// copy remaining bytes into sample buffer
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
memcpy(audio_decoder->Buffer[ch], audio_decoder->Remain[ch], audio_decoder->RemainCount);
}
// deinterleave samples into sample buffer
for (i = 0; i < bytes_n / 2; i++) {
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
audio_decoder->Buffer[ch][audio_decoder->RemainCount / 2 + i]
= data[i * audio_decoder->HwChannels + ch];
}
}
bytes_n += audio_decoder->RemainSize;
n = 0; // keep gcc lucky
// resample the sample buffer into tmp buffer
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
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);
// fixme remaining channels
if (bytes_n - consumed * 2 > audio_decoder->RemainSize) {
audio_decoder->RemainSize = bytes_n - consumed * 2;
}
audio_decoder->Remain[ch] = realloc(audio_decoder->Remain[ch], audio_decoder->RemainSize);
memcpy(audio_decoder->Remain[ch], audio_decoder->Buffer[ch] + consumed, audio_decoder->RemainSize);
audio_decoder->RemainCount = audio_decoder->RemainSize;
}
// interleave samples from sample buffer
for (i = 0; i < n; i++) {
for (ch = 0; ch < audio_decoder->HwChannels; ++ch) {
buf[i * audio_decoder->HwChannels + ch] = buftmp[ch][i];
}
}
n *= 2;
n *= audio_decoder->HwChannels;
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(buf, n, audio_decoder->HwChannels);
}
AudioEnqueue(buf, n);
return;
}
#endif
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame(data, count, audio_decoder->HwChannels);
}
AudioEnqueue(data, count);
}
int myavcodec_decode_audio3(AVCodecContext * avctx, int16_t * samples, int *frame_size_ptr, AVPacket * avpkt)
{
AVFrame *frame = av_frame_alloc();
int ret, got_frame = 0;
if (!frame)
return AVERROR(ENOMEM);
#if 0
ret = avcodec_decode_audio4(avctx, frame, &got_frame, avpkt);
#else
// SUGGESTION
// Now that avcodec_decode_audio4 is deprecated and replaced
// by 2 calls (receive frame and send packet), this could be optimized
// into separate routines or separate threads.
// Also now that it always consumes a whole buffer some code
// in the caller may be able to be optimized.
ret = avcodec_receive_frame(avctx, frame);
if (ret == 0)
got_frame = 1;
if (ret == AVERROR(EAGAIN))
ret = 0;
if (ret == 0)
ret = avcodec_send_packet(avctx, avpkt);
if (ret == AVERROR(EAGAIN))
ret = 0;
else if (ret < 0) {
// Debug(3, "codec/audio: audio decode error: %1 (%2)\n",av_make_error_string(error, sizeof(error), ret),got_frame);
return ret;
} else
ret = avpkt->size;
#endif
if (ret >= 0 && got_frame) {
int i, ch;
int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
int data_size = av_get_bytes_per_sample(avctx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i = 0; i < frame->nb_samples; i++) {
for (ch = 0; ch < avctx->channels; ch++) {
memcpy(samples, frame->extended_data[ch] + data_size * i, data_size);
samples = (char *)samples + data_size;
}
}
// Debug(3,"data_size %d nb_samples %d sample_fmt %d channels %d planar %d\n",data_size,frame->nb_samples,avctx->sample_fmt,avctx->channels,planar);
*frame_size_ptr = data_size * avctx->channels * frame->nb_samples;
} else {
*frame_size_ptr = 0;
}
av_frame_free(&frame);
return ret;
}
/**
** Decode an audio packet.
**
** PTS must be handled self.
**
** @param audio_decoder audio decoder data
** @param avpkt audio packet
*/
void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
{
int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 + AV_INPUT_BUFFER_PADDING_SIZE] __attribute__((aligned(16)));
int buf_sz;
int l;
AVCodecContext *audio_ctx;
audio_ctx = audio_decoder->AudioCtx;
// FIXME: don't need to decode pass-through codecs
buf_sz = sizeof(buf);
l = myavcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *) avpkt);
if (avpkt->size != l) {
if (l == AVERROR(EAGAIN)) {
Error(_("codec: latm\n"));
return;
}
if (l < 0) { // no audio frame could be decompressed
Error(_("codec: error audio data\n"));
return;
}
Error(_("codec: error more than one frame data\n"));
}
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// FIXME: must first play remainings bytes, than change and play new.
if (audio_decoder->Passthrough != CodecPassthrough || audio_decoder->SampleRate != audio_ctx->sample_rate
|| audio_decoder->Channels != audio_ctx->channels) {
CodecAudioUpdateFormat(audio_decoder);
}
if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
// need to resample audio
if (audio_decoder->ReSample) {
int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 + AV_INPUT_BUFFER_PADDING_SIZE]
__attribute__((aligned(16)));
int outlen;
// FIXME: libav-0.7.2 crash here
outlen = audio_resample(audio_decoder->ReSample, outbuf, buf, buf_sz);
#ifdef DEBUG
if (outlen != buf_sz) {
Debug(3, "codec/audio: possible fixed ffmpeg\n");
}
#endif
if (outlen) {
// outlen seems to be wrong in ffmpeg-0.9
outlen /= audio_decoder->Channels * av_get_bytes_per_sample(audio_ctx->sample_fmt);
outlen *= audio_decoder->HwChannels * av_get_bytes_per_sample(audio_ctx->sample_fmt);
Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
CodecAudioEnqueue(audio_decoder, outbuf, outlen);
}
} else {
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
CodecAudioEnqueue(audio_decoder, buf, buf_sz);
}
}
}
#endif
#if defined(USE_SWRESAMPLE) || defined(USE_AVRESAMPLE)
@@ -1757,7 +1346,6 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
if (audio_decoder->Resample) {
uint8_t outbuf[8192 * 2 * 8];
uint8_t *out[1];
out[0] = outbuf;
ret =
swr_convert(audio_decoder->Resample, out, sizeof(outbuf) / (2 * audio_decoder->HwChannels),
@@ -1784,6 +1372,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
*/
void CodecAudioFlushBuffers(AudioDecoder * decoder)
{
avcodec_flush_buffers(decoder->AudioCtx);
}