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| author | Ronald S. Bultje <rsbultje@gmail.com> | 2009-03-03 13:51:34 +0000 |
|---|---|---|
| committer | Ronald S. Bultje <rsbultje@gmail.com> | 2009-03-03 13:51:34 +0000 |
| commit | eafb17d140f6772c9aac8fbf31641f24a371b2c0 (patch) | |
| tree | 20846edc04bef15e71e204bab43f99e2a6a3ac3a /libavformat/file.c | |
| parent | 0d8ee24c7b7a1ed0f28b0000bda77b07d4137135 (diff) | |
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/file.c')
0 files changed, 0 insertions, 0 deletions