Almost none of that function's complexity is useful for audio, it can
be replaced by a simple av_rescale_q().
---
fftools/ffmpeg.c | 11 +++++++++--
1 file changed, 9 insertions(+), 2 deletions(-)
diff --git a/fftools/ffmpeg.c b/fftools/ffmpeg.c
index 2ee40890a9..cb65f26100 100644
--- a/fftools/ffmpeg.c
+++ b/fftools/ffmpeg.c
@@ -933,15 +933,22 @@ static int submit_encode_frame(OutputFile *of,
OutputStream *ost,
static void do_audio_out(OutputFile *of, OutputStream *ost,
AVFrame *frame)
{
+ AVCodecContext *enc = ost->enc_ctx;
int ret;
- adjust_frame_pts_to_encoder_tb(of, ost, frame);
-
if (!check_recording_time(ost, ost->next_pts, ost->enc_ctx->time_base))
return;
if (frame->pts == AV_NOPTS_VALUE)
frame->pts = ost->next_pts;
+ else {
+ int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 :
of->start_time;
+ frame->pts =
+ av_rescale_q(frame->pts, frame->time_base, enc->time_base) -
+ av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base);
+ }
+ frame->time_base = enc->time_base;
+
ost->next_pts = frame->pts + frame->nb_samples;
ret = submit_encode_frame(of, ost, frame);
--
2.35.1
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