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We now allow for the possibility of the RTCP audio/video synchronization being

incorrect.  (I encounted a stream for which this was the case.)  Now, if
audio and video are out-of-sync by >60 seconds, we assume that the RTCP
sync is incorrect, and we don't discard any packets.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12008 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
rsf 2004-03-02 08:52:59 +00:00
parent 376e3abf5c
commit 162310efe9

View File

@ -305,8 +305,14 @@ extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
// audio and video streams get this far apart.)
// (We don't do this when streaming over TCP, because then the audio and
// video streams are interleaved.)
// (Also, if the stream is *excessively* far behind, then we allow
// the packet, because in this case it probably means that there was
// an error in the source's timestamp synchronization.)
const float ptsBehindThreshold = 1.0; // seconds
if (ptsBehind < ptsBehindThreshold || rtspStreamOverTCP) { // packet's OK
const float ptsBehindLimit = 60.0; // seconds
if (ptsBehind < ptsBehindThreshold ||
ptsBehind > ptsBehindLimit ||
rtspStreamOverTCP) { // packet's OK
ds_add_packet(ds, dp);
break;
}