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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-28 20:53:54 +02:00

fftools/ffmpeg: move do_streamcopy() to ffmpeg_mux

do_streamcopy() is muxing code, so this is a more appropriate place for
this. The last uses of InputStream in it will be removed in following
commits.
This commit is contained in:
Anton Khirnov 2023-04-04 09:44:42 +02:00
parent a34f483291
commit 2178ff2162
3 changed files with 84 additions and 80 deletions

View File

@ -914,85 +914,6 @@ int ifilter_parameters_from_codecpar(InputFilter *ifilter, AVCodecParameters *pa
return 0;
}
/**
* @param dts predicted packet dts in AV_TIME_BASE_Q
*/
static void do_streamcopy(InputStream *ist, OutputStream *ost,
const AVPacket *pkt, int64_t dts)
{
OutputFile *of = output_files[ost->file_index];
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);
AVPacket *opkt = ost->pkt;
av_packet_unref(opkt);
// EOF: flush output bitstream filters.
if (!pkt) {
of_output_packet(of, opkt, ost, 1);
return;
}
if (!ost->streamcopy_started && !(pkt->flags & AV_PKT_FLAG_KEY) &&
!ost->copy_initial_nonkeyframes)
return;
if (!ost->streamcopy_started) {
if (!ost->copy_prior_start &&
(pkt->pts == AV_NOPTS_VALUE ?
dts < ost->ts_copy_start :
pkt->pts < av_rescale_q(ost->ts_copy_start, AV_TIME_BASE_Q, pkt->time_base)))
return;
if (of->start_time != AV_NOPTS_VALUE && dts < of->start_time)
return;
}
if (of->recording_time != INT64_MAX &&
dts >= of->recording_time + start_time) {
close_output_stream(ost);
return;
}
if (av_packet_ref(opkt, pkt) < 0)
exit_program(1);
opkt->time_base = ost->mux_timebase;
if (pkt->pts != AV_NOPTS_VALUE)
opkt->pts = av_rescale_q(pkt->pts, pkt->time_base, opkt->time_base) - ost_tb_start_time;
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(ist->par, pkt->size);
if(!duration)
duration = ist->par->frame_size;
opkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
(AVRational){1, ist->par->sample_rate}, duration,
&ist->filter_in_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);
{
int ret = trigger_fix_sub_duration_heartbeat(ost, pkt);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR,
"Subtitle heartbeat logic failed in %s! (%s)\n",
__func__, av_err2str(ret));
exit_program(1);
}
}
of_output_packet(of, opkt, ost, 0);
ost->streamcopy_started = 1;
}
static void check_decode_result(InputStream *ist, int *got_output, int ret)
{
if (*got_output || ret<0)
@ -1847,7 +1768,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt, int no_eo
continue;
}
do_streamcopy(ist, ost, pkt, ist->dts);
of_streamcopy(ist, ost, pkt, ist->dts);
}
return !eof_reached;

View File

@ -872,6 +872,13 @@ void of_enc_stats_close(void);
* must be supplied in this case.
*/
void of_output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof);
/**
* @param dts predicted packet dts in AV_TIME_BASE_Q
*/
void of_streamcopy(InputStream *ist, OutputStream *ost,
const AVPacket *pkt, int64_t dts);
int64_t of_filesize(OutputFile *of);
int ifile_open(const OptionsContext *o, const char *filename);

View File

@ -378,6 +378,82 @@ fail:
}
void of_streamcopy(InputStream *ist, OutputStream *ost,
const AVPacket *pkt, int64_t dts)
{
OutputFile *of = output_files[ost->file_index];
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);
AVPacket *opkt = ost->pkt;
av_packet_unref(opkt);
// EOF: flush output bitstream filters.
if (!pkt) {
of_output_packet(of, opkt, ost, 1);
return;
}
if (!ost->streamcopy_started && !(pkt->flags & AV_PKT_FLAG_KEY) &&
!ost->copy_initial_nonkeyframes)
return;
if (!ost->streamcopy_started) {
if (!ost->copy_prior_start &&
(pkt->pts == AV_NOPTS_VALUE ?
dts < ost->ts_copy_start :
pkt->pts < av_rescale_q(ost->ts_copy_start, AV_TIME_BASE_Q, pkt->time_base)))
return;
if (of->start_time != AV_NOPTS_VALUE && dts < of->start_time)
return;
}
if (of->recording_time != INT64_MAX &&
dts >= of->recording_time + start_time) {
close_output_stream(ost);
return;
}
if (av_packet_ref(opkt, pkt) < 0)
exit_program(1);
opkt->time_base = ost->mux_timebase;
if (pkt->pts != AV_NOPTS_VALUE)
opkt->pts = av_rescale_q(pkt->pts, pkt->time_base, opkt->time_base) - ost_tb_start_time;
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(ist->par, pkt->size);
if(!duration)
duration = ist->par->frame_size;
opkt->dts = av_rescale_delta(pkt->time_base, pkt->dts,
(AVRational){1, ist->par->sample_rate}, duration,
&ist->filter_in_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);
{
int ret = trigger_fix_sub_duration_heartbeat(ost, pkt);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR,
"Subtitle heartbeat logic failed in %s! (%s)\n",
__func__, av_err2str(ret));
exit_program(1);
}
}
of_output_packet(of, opkt, ost, 0);
ost->streamcopy_started = 1;
}
static int thread_stop(Muxer *mux)
{
void *ret;