This allows libavcodec to use the input packet pts when determining output
frame pts.
---
avconv.c | 9 +++++----
1 files changed, 5 insertions(+), 4 deletions(-)
diff --git a/avconv.c b/avconv.c
index 2f30ebc..3767198 100644
--- a/avconv.c
+++ b/avconv.c
@@ -1085,6 +1085,11 @@ static int decode_audio(InputStream *ist, AVPacket *pkt,
int *got_output)
int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);
int i, ret, resample_changed;
+ /* rescale pts to decoder time base */
+ if (pkt->pts != AV_NOPTS_VALUE)
+ pkt->pts = av_rescale_q(pkt->pts, ist->st->time_base,
+ ist->st->codec->time_base);
+
if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))
return AVERROR(ENOMEM);
else
@@ -1210,10 +1215,6 @@ static int decode_audio(InputStream *ist, AVPacket *pkt,
int *got_output)
}
}
- if (decoded_frame->pts != AV_NOPTS_VALUE)
- decoded_frame->pts = av_rescale_q(decoded_frame->pts,
- ist->st->time_base,
- (AVRational){1,
ist->st->codec->sample_rate});
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_write_frame(ist->filters[i]->filter, decoded_frame);
--
1.7.1
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