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authorRonald S. Bultje <rsbultje@gmail.com>2009-03-03 13:51:34 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2009-03-03 13:51:34 +0000
commiteafb17d140f6772c9aac8fbf31641f24a371b2c0 (patch)
tree20846edc04bef15e71e204bab43f99e2a6a3ac3a /libavformat/rtpdec.c
parent0d8ee24c7b7a1ed0f28b0000bda77b07d4137135 (diff)
downloadffmpeg-eafb17d140f6772c9aac8fbf31641f24a371b2c0.tar.gz
Don't let finalize_packet() touch pkt->stream_index. Instead, let individual
payload handlers take care of that themselves at their own option. What this patch really does is "fix" a bug in MS-RTSP protocol where incoming packets are always coming in over the connection (UDP) or interleave-id (TCP) of the stream-id of the first ASF packet in the RTP packet. However, RTP packets may contain multiple ASF packets (and usually do, from what I can see), and therefore this leads to playback bugs. The intended stream-id per ASF packet is given in the respective ASF packet header. The ASF demuxer will correctly read this and set pkt->stream_index, but since the "stream" parameter can not be known to rtpdec.c or any of the RTP/RTSP code, the "st" parameter in all these functions is basically invalid. Therefore, using st->id as pkt->stream_index leads to various playback bugs. The result of this patch is that pkt->stream_index is left untouched for RTP/ASF (and possibly for other payloads that have similar behaviour). The patch was discussed in the "[PATCH] rtpdec.c: don't overwrite pkt->stream_index in finalize_packet()" thread on the mailinglist. Originally committed as revision 17767 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtpdec.c')
-rw-r--r--libavformat/rtpdec.c3
1 files changed, 2 insertions, 1 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 4a5257bc6d..e1ba888cb1 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -382,7 +382,6 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
pkt->pts = addend + delta_timestamp;
}
- pkt->stream_index = s->st->index;
}
/**
@@ -536,6 +535,8 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
memcpy(pkt->data, buf, len);
break;
}
+
+ pkt->stream_index = st->index;
}
// now perform timestamp things....