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authorMichael Niedermayer <michaelni@gmx.at>2012-02-17 00:35:06 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-02-17 00:35:06 +0100
commit8c1ebdcea2d3f84bd5c01ce3dce5e619c845bc0f (patch)
tree977c2d42de80b6c292ff0f9ad79f14762ba792f6 /libavformat/rtspenc.c
parent7bdefc0f1281c2c18375197985fcd5109e711829 (diff)
parent204cb29b3c84a74cbcd059d353c70c8bdc567d98 (diff)
Merge remote-tracking branch 'qatar/master'
* qatar/master: shorten: Use separate pointers for the allocated memory for decoded samples. atrac3: Fix crash in tonal component decoding. ws_snd1: Fix wrong samples counts. movenc: Don't set a default sample duration when creating ismv rtp: Factorize the check for distinguishing RTCP packets from RTP golomb: avoid infinite loop on all-zero input (or end of buffer). bethsoftvid: synchronize video timestamps with audio sample rate bethsoftvid: add audio stream only after getting the first audio packet bethsoftvid: Set video packet duration instead of accumulating pts. bethsoftvid: set packet key frame flag for audio and I-frame video packets. bethsoftvid: fix read_packet() return codes. bethsoftvid: pass palette in side data instead of in a separate packet. sdp: Ignore RTCP packets when autodetecting RTP streams proresenc: initialise 'sign' variable mpegaudio: replace memcpy by SIMD code vc1: prevent using last_frame as a reference for I/P first frame. Conflicts: libavcodec/atrac3.c libavcodec/golomb.h libavcodec/shorten.c libavcodec/ws-snd1.c tests/ref/fate/bethsoft-vid Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtspenc.c')
-rw-r--r--libavformat/rtspenc.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/libavformat/rtspenc.c b/libavformat/rtspenc.c
index 7bdacc29f7..f3c846a5fb 100644
--- a/libavformat/rtspenc.c
+++ b/libavformat/rtspenc.c
@@ -159,7 +159,7 @@ static int tcp_write_packet(AVFormatContext *s, RTSPStream *rtsp_st)
size -= 4;
if (packet_len > size || packet_len < 2)
break;
- if (ptr[1] >= RTCP_SR && ptr[1] <= RTCP_APP)
+ if (RTP_PT_IS_RTCP(ptr[1]))
id = rtsp_st->interleaved_max; /* RTCP */
else
id = rtsp_st->interleaved_min; /* RTP */