From eafb17d140f6772c9aac8fbf31641f24a371b2c0 Mon Sep 17 00:00:00 2001
From: "Ronald S. Bultje" <rsbultje@gmail.com>
Date: Tue, 3 Mar 2009 13:51:34 +0000
Subject: [PATCH] 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
---
libavformat/rtp_h264.c | 2 ++
libavformat/rtpdec.c | 3 ++-
2 files changed, 4 insertions(+), 1 deletion(-)
diff --git a/libavformat/rtp_h264.c b/libavformat/rtp_h264.c
index f3ae6993f54..47cbf79db51 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 4a5257bc6d8..e1ba888cb1b 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....
--
GitLab