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Add pts adjustment to the delayed flushed encoder packets

This commit is contained in:
Isaac Connor 2019-07-11 17:56:53 -04:00
parent 288f2f3e8f
commit 13c91bdf60
1 changed files with 50 additions and 24 deletions
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74
src/zm_videostore.cpp
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@ -476,7 +476,51 @@ VideoStore::~VideoStore() {
break;
}
#endif
// Need to adjust pts and dts and duration
pkt.stream_index = audio_out_stream->index;
pkt.duration = av_rescale_q(
pkt.duration,
audio_out_ctx->time_base,
audio_out_stream->time_base);
// Scale the PTS of the outgoing packet to be the correct time base
if ( pkt.pts != AV_NOPTS_VALUE ) {
pkt.pts = av_rescale_q(
pkt.pts,
audio_out_ctx->time_base,
audio_in_stream->time_base);
// audio_first_pts is in audio_in_stream time base
pkt.pts -= audio_first_pts;
pkt.pts = av_rescale_q(
pkt.pts,
audio_in_stream->time_base,
audio_out_stream->time_base);
Debug(2, "audio pkt.pts = %" PRId64 " from first_pts(%" PRId64 ")",
pkt.pts, audio_first_pts);
} else {
Debug(2, "pkt.pts = undef");
pkt.pts = AV_NOPTS_VALUE;
}
if ( pkt.dts != AV_NOPTS_VALUE ) {
pkt.dts = av_rescale_q(
pkt.dts,
audio_out_ctx->time_base,
audio_in_stream->time_base);
pkt.dts -= audio_first_dts;
pkt.dts = av_rescale_q(
pkt.dts,
audio_in_ctx->time_base,
audio_out_stream->time_base);
Debug(2, "pkt.dts = %" PRId64 " - first_dts(%" PRId64 ")",
pkt.dts, audio_first_dts);
} else {
pkt.dts = AV_NOPTS_VALUE;
}
dumpPacket(audio_out_stream, &pkt, "writing flushed packet");
av_interleaved_write_frame(oc, &pkt);
zm_av_packet_unref(&pkt);
@ -1196,28 +1240,9 @@ int VideoStore::resample_audio() {
av_make_error_string(ret).c_str());
return 0;
}
zm_dump_frame(out_frame, "Out frame after convert");
#if 0
// out_frame pts is in the input pkt pts... needs to be adjusted before sending to the encoder
if ( out_frame->pts != AV_NOPTS_VALUE ) {
if ( !video_first_pts ) {
video_first_pts = out_frame->pts;
Debug(1, "No video_first_pts setting to %" PRId64, video_first_pts);
out_frame->pts = 0;
} else {
out_frame->pts = out_frame->pts - video_first_pts;
}
//
} else {
// sending AV_NOPTS_VALUE doesn't really work but we seem to get it in ffmpeg 2.8
out_frame->pts = audio_next_pts;
}
audio_next_pts = out_frame->pts + out_frame->nb_samples;
#endif
if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + out_frame->nb_samples)) < 0) {
ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + out_frame->nb_samples);
if ( ret < 0 ) {
Error("Could not reallocate FIFO");
return 0;
}
@ -1232,7 +1257,7 @@ int VideoStore::resample_audio() {
// Reset frame_size to output_frame_size
int frame_size = audio_out_ctx->frame_size;
// AAC requires 1024 samples per encode. Our input tends to be 160, so need to buffer them.
// AAC requires 1024 samples per encode. Our input tends to be something else, so need to buffer them.
if ( frame_size > av_audio_fifo_size(fifo) ) {
return 0;
}
@ -1245,10 +1270,11 @@ int VideoStore::resample_audio() {
// resampling changes the duration because the timebase is 1/samples
// I think we should be dealing in codec timebases not stream
if ( in_frame->pts != AV_NOPTS_VALUE ) {
// pkt_duration is in avstream timebase units
out_frame->pkt_duration = av_rescale_q(
in_frame->pkt_duration,
audio_in_ctx->time_base,
audio_out_ctx->time_base);
audio_in_stream->time_base,
audio_out_stream->time_base);
out_frame->pts = av_rescale_q(
in_frame->pts,
audio_in_ctx->time_base,