From bbd8f5477d29ae0d22ad0a8711d95299e009ec36 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Storsj=C3=B6?= Date: Wed, 15 Dec 2010 21:06:25 +0000 Subject: 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 --- libavformat/rtsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'libavformat/rtsp.c') diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 1dd166e..2703400 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( -- cgit v1.1