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
This commit is contained in:
Ronald S. Bultje 2009-03-03 13:51:34 +00:00
parent 0d8ee24c7b
commit eafb17d140
2 changed files with 4 additions and 1 deletions

View File

@ -310,6 +310,8 @@ static int h264_handle_packet(AVFormatContext *ctx,
break;
}
pkt->stream_index = st->index;
return result;
}

View File

@ -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....