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

1 files changed, 2 insertions, 0 deletions

diff --git a/libavformat/rtp_h264.c b/libavformat/rtp_h264.c
index f3ae6993f5..47cbf79db5 100644
--- a/libavformat/rtp_h264.c
+++ b/libavformat/rtp_h264.c
@@ -310,6 +310,8 @@ static int h264_handle_packet(AVFormatContext *ctx,
         break;
     }
 
+    pkt->stream_index = st->index;
+
     return result;
 }