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fftools/ffmpeg_mux: stop rescaling timestamps in of_streamcopy()

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This function converts packet timestamps from the input stream timebase
to OutputStream.mux_timebase, which may or may not be equal to the
actual output AVStream timebase (and even when it is, this may not
always be the optimal choice due to bitstream filtering).

Just keep the timestamps in input stream timebase, they will be rescaled
as needed before bitstream filtering and/or sending the packet to the
muxer.

Move the av_rescale_delta() call for audio (needed to preserve accuracy
with coarse demuxer timebases) to write_packet.

Drop now-unused OutputStream.mux_timebase.
This commit is contained in:
Anton Khirnov
2023-07-25 16:28:07 +02:00
parent ed5caaaf22
commit 8ecbb1f9af
5 changed files with 24 additions and 28 deletions
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44
fftools/ffmpeg_mux.c
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@@ -76,7 +76,23 @@ static int write_packet(Muxer *mux, OutputStream *ost, AVPacket *pkt)
if (ost->type == AVMEDIA_TYPE_VIDEO && ost->vsync_method == VSYNC_DROP)
pkt->pts = pkt->dts = AV_NOPTS_VALUE;
av_packet_rescale_ts(pkt, pkt->time_base, ost->st->time_base);
// rescale timestamps to the stream timebase
if (ost->type == AVMEDIA_TYPE_AUDIO && !ost->enc) {
// use av_rescale_delta() for streamcopying audio, to preserve
// accuracy with coarse input timebases
int duration = av_get_audio_frame_duration2(ost->st->codecpar, pkt->size);
if (!duration)
duration = ost->st->codecpar->frame_size;
pkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
(AVRational){1, ost->st->codecpar->sample_rate}, duration,
&ms->ts_rescale_delta_last, ost->st->time_base);
pkt->pts = pkt->dts;
pkt->duration = av_rescale_q(pkt->duration, pkt->time_base, ost->st->time_base);
} else
av_packet_rescale_ts(pkt, pkt->time_base, ost->st->time_base);
pkt->time_base = ost->st->time_base;
if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
@@ -392,7 +408,7 @@ int of_streamcopy(OutputStream *ost, const AVPacket *pkt, int64_t dts)
OutputFile *of = output_files[ost->file_index];
MuxStream *ms = ms_from_ost(ost);
int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->mux_timebase);
int64_t ts_offset;
AVPacket *opkt = ms->pkt;
int ret;
@@ -425,27 +441,17 @@ int of_streamcopy(OutputStream *ost, const AVPacket *pkt, int64_t dts)
if (ret < 0)
return ret;
opkt->time_base = ost->mux_timebase;
ts_offset = av_rescale_q(start_time, AV_TIME_BASE_Q, opkt->time_base);
if (pkt->pts != AV_NOPTS_VALUE)
opkt->pts = av_rescale_q(pkt->pts, pkt->time_base, opkt->time_base) - ost_tb_start_time;
opkt->pts -= ts_offset;
if (pkt->dts == AV_NOPTS_VALUE) {
opkt->dts = av_rescale_q(dts, AV_TIME_BASE_Q, opkt->time_base);
} else if (ost->st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
int duration = av_get_audio_frame_duration2(ost->par_in, pkt->size);
if(!duration)
duration = ost->par_in->frame_size;
opkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
(AVRational){1, ost->par_in->sample_rate}, duration,
&ms->ts_rescale_delta_last, opkt->time_base);
/* dts will be set immediately afterwards to what pts is now */
opkt->pts = opkt->dts - ost_tb_start_time;
} else
opkt->dts = av_rescale_q(pkt->dts, pkt->time_base, opkt->time_base);
opkt->dts -= ost_tb_start_time;
opkt->duration = av_rescale_q(pkt->duration, pkt->time_base, opkt->time_base);
opkt->pts = opkt->dts - ts_offset;
}
opkt->dts -= ts_offset;
{
int ret = trigger_fix_sub_duration_heartbeat(ost, pkt);
@@ -511,10 +517,6 @@ static int thread_start(Muxer *mux)
MuxStream *ms = ms_from_ost(ost);
AVPacket *pkt;
/* try to improve muxing time_base (only possible if nothing has been written yet) */
if (!av_fifo_can_read(ms->muxing_queue))
ost->mux_timebase = ost->st->time_base;
while (av_fifo_read(ms->muxing_queue, &pkt, 1) >= 0) {
ret = thread_submit_packet(mux, ost, pkt);
if (pkt) {