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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 | |
| parent | 0d8ee24c7b7a1ed0f28b0000bda77b07d4137135 (diff) | |
| download | ffmpeg-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')
| -rw-r--r-- | libavformat/rtp_h264.c | 2 | ||||
| -rw-r--r-- | libavformat/rtpdec.c | 3 |
2 files changed, 4 insertions, 1 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; } 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.... |