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-rw-r--r--libavformat/ogg.c4
-rw-r--r--libavformat/utils.c13
2 files changed, 9 insertions, 8 deletions
diff --git a/libavformat/ogg.c b/libavformat/ogg.c
index 8277d91821..65c507c240 100644
--- a/libavformat/ogg.c
+++ b/libavformat/ogg.c
@@ -73,10 +73,6 @@ static int ogg_write_packet(AVFormatContext *avfcontext,
pts= av_rescale(pts, avctx->sample_rate, AV_TIME_BASE);
- if(!size){
-// av_log(avfcontext, AV_LOG_DEBUG, "zero packet\n");
- return 0;
- }
// av_log(avfcontext, AV_LOG_DEBUG, "M%d\n", size);
/* flush header packets so audio starts on a new page */
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 14f965d697..6a445e2589 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -1698,8 +1698,14 @@ int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
st = s->streams[stream_index];
pts_mask = (1LL << s->pts_wrap_bits) - 1;
- ret = s->oformat->write_packet(s, stream_index, buf, size,
- st->pts.val & pts_mask);
+
+ /* HACK/FIXME we skip all zero size audio packets so a encoder can pass pts by outputing zero size packets */
+ if(st->codec.codec_type==CODEC_TYPE_AUDIO && size==0)
+ ret = 0;
+ else
+ ret = s->oformat->write_packet(s, stream_index, buf, size,
+ st->pts.val & pts_mask);
+
if (ret < 0)
return ret;
@@ -1708,9 +1714,8 @@ int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
case CODEC_TYPE_AUDIO:
frame_size = get_audio_frame_size(&st->codec, size);
- /* note, we skip the initial 0-size packets as they are most likely equal to the encoder delay,
+ /* HACK/FIXME, we skip the initial 0-size packets as they are most likely equal to the encoder delay,
but it would be better if we had the real timestamps from the encoder */
-// av_log(s, AV_LOG_DEBUG, "%d %lld %lld\n", size, st->pts.num, st->pts.val);
if (frame_size >= 0 && (size || st->pts.num!=st->pts.den>>1 || st->pts.val)) {
av_frac_add(&st->pts,
(int64_t)s->pts_den * frame_size);