Vaapi changed to egl

This commit is contained in:
jojo61
2019-10-04 10:37:57 +02:00
parent cdac2bcc3d
commit 4d2735a971
7 changed files with 1895 additions and 1504 deletions

290
codec.c
View File

@@ -315,7 +315,7 @@ void CodecVideoOpen(VideoDecoder * decoder, int codec_id)
decoder->VideoCtx->pkt_timebase.den = 90000;
decoder->VideoCtx->framerate.num = 50;
decoder->VideoCtx->framerate.den = 1;
decoder->VideoCtx->extra_hw_frames = 8; // VIDEO_SURFACES_MAX +1
pthread_mutex_lock(&CodecLockMutex);
// open codec
@@ -323,6 +323,7 @@ void CodecVideoOpen(VideoDecoder * decoder, int codec_id)
deint = 2;
#endif
#ifdef VAAPI
decoder->VideoCtx->extra_hw_frames = 8; // VIDEO_SURFACES_MAX +1
if (video_codec->capabilities & (AV_CODEC_CAP_AUTO_THREADS)) {
Debug(3,"codec: auto threads enabled");
decoder->VideoCtx->thread_count = 0;
@@ -1220,23 +1221,23 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
delay = AudioGetDelay();
if (!delay) {
return;
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;
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;
return;
}
tim_diff = (nowtime.tv_sec - audio_decoder->LastTime.tv_sec)
@@ -1262,51 +1263,51 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
// 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;
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
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;
}
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;
}
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;
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);
// 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;
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
}
@@ -1324,64 +1325,64 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder)
int err;
if (audio_decoder->ReSample) {
audio_resample_close(audio_decoder->ReSample);
audio_decoder->ReSample = NULL;
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;
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);
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"));
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;
}
Debug(3, "codec/audio: audio setup error\n");
// FIXME: handle errors
audio_decoder->HwChannels = 0;
audio_decoder->HwSampleRate = 0;
return;
return;
}
if (passthrough) { // pass-through no conversion allowed
return;
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);
}
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
}
@@ -1718,22 +1719,22 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts)
#endif
#ifdef USE_AVRESAMPLE
if (audio_decoder->Resample && audio_decoder->DriftCorr) {
int distance;
int distance;
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
if (avresample_set_compensation(audio_decoder->Resample,
audio_decoder->DriftCorr / 10, distance)) {
Debug(3, "codec/audio: swr_set_compensation failed\n");
}
distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000);
if (avresample_set_compensation(audio_decoder->Resample,
audio_decoder->DriftCorr / 10, distance)) {
Debug(3, "codec/audio: swr_set_compensation failed\n");
}
}
#endif
if (1) {
static int c;
static int c;
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
if (!(c++ % 10)) {
Debug(3, "codec/audio: drift(%6d) %8dus %5d\n",
audio_decoder->DriftCorr, drift * 1000 / 90, corr);
}
}
#else
AudioSetClock(pts);
@@ -1825,55 +1826,56 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
AVCodecContext *audio_ctx = audio_decoder->AudioCtx;
if (audio_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
int ret;
AVPacket pkt[1];
AVFrame *frame = audio_decoder->Frame;
int ret;
AVPacket pkt[1];
AVFrame *frame = audio_decoder->Frame;
av_frame_unref(frame);
*pkt = *avpkt; // use copy
ret = avcodec_send_packet(audio_ctx, pkt);
if (ret < 0) {
Debug(3, "codec: sending audio packet failed");
return;
}
ret = avcodec_receive_frame(audio_ctx, frame);
if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
Debug(3, "codec: receiving audio frame failed");
return;
}
if (ret >= 0) {
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
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) {
return; // unsupported sample format
}
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
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),
(const uint8_t **)frame->extended_data, frame->nb_samples);
if (ret > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf, ret * 2 * audio_decoder->HwChannels,
audio_decoder->HwChannels);
}
AudioEnqueue(outbuf, ret * 2 * audio_decoder->HwChannels);
av_frame_unref(frame);
*pkt = *avpkt; // use copy
ret = avcodec_send_packet(audio_ctx, pkt);
if (ret < 0) {
Debug(3, "codec: sending audio packet failed");
return;
}
ret = avcodec_receive_frame(audio_ctx, frame);
if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
Debug(3, "codec: receiving audio frame failed");
return;
}
if (ret >= 0) {
// update audio clock
if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
CodecAudioSetClock(audio_decoder, avpkt->pts);
}
// format change
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) {
return; // unsupported sample format
}
if (CodecAudioPassthroughHelper(audio_decoder, avpkt)) {
return;
}
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),
(const uint8_t **)frame->extended_data, frame->nb_samples);
if (ret > 0) {
if (!(audio_decoder->Passthrough & CodecPCM)) {
CodecReorderAudioFrame((int16_t *) outbuf, ret * 2 * audio_decoder->HwChannels,
audio_decoder->HwChannels);
}
AudioEnqueue(outbuf, ret * 2 * audio_decoder->HwChannels);
}
return;
}
}
return;
}
}
}
}