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| author | Martin Storsjö <martin@martin.st> | 2010-12-15 21:06:25 +0000 |
|---|---|---|
| committer | Martin Storsjö <martin@martin.st> | 2010-12-15 21:06:25 +0000 |
| commit | bbd8f5477d29ae0d22ad0a8711d95299e009ec36 (patch) | |
| tree | 8f4e41523d9d8b1f80f113e8f2943539c8e85e4d /libavformat/rtsp.c | |
| parent | 1c3e117e0bd73ffc5a3abeb35b521fd048988f06 (diff) | |
| download | ffmpeg-bbd8f5477d29ae0d22ad0a8711d95299e009ec36.tar.gz | |
rtsp: Don't set the RTP time base from the sample rate if no sample rate is set
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.
The stream that triggered the fix in 26016 still works after this commit.
Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtsp.c')
| -rw-r--r-- | libavformat/rtsp.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 1dd166e3f1..2703400720 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -333,7 +333,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, RTPDynamicProtocolHandler *handler; /* if standard payload type, we can find the codec right now */ ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && + st->codec->sample_rate > 0) av_set_pts_info(st, 32, 1, st->codec->sample_rate); /* Even static payload types may need a custom depacketizer */ handler = ff_rtp_handler_find_by_id( |