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authorAnton Khirnov <anton@khirnov.net>2023-04-04 09:44:42 +0200
committerAnton Khirnov <anton@khirnov.net>2023-04-17 12:01:40 +0200
commit2178ff2162ba4898c5f1cc3f334b00392451efd6 (patch)
tree71db8ab65237ccd75dbbdd5ebbf522db4f752fc9 /fftools/ffmpeg_mux.c
parenta34f483291e5e635e55e0d326c23bb8c352852c3 (diff)
downloadffmpeg-2178ff2162ba4898c5f1cc3f334b00392451efd6.tar.gz
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.
Diffstat (limited to 'fftools/ffmpeg_mux.c')
-rw-r--r--fftools/ffmpeg_mux.c76
1 files changed, 76 insertions, 0 deletions
diff --git a/fftools/ffmpeg_mux.c b/fftools/ffmpeg_mux.c
index 40b439eea2..96ed50f73c 100644
--- a/fftools/ffmpeg_mux.c
+++ b/fftools/ffmpeg_mux.c
@@ -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;