This indicates that there was no error that needs to be reported to the
caller, so we can move on to parse the next packet immediately, if
available. The only error code that ff_mpegts_parse_packet can return
indicates that there was no packet to return from the provided data, for
which it returns -1.
Originally committed as revision 25496 to svn://svn.ffmpeg.org/ffmpeg/trunk
It may have returned a negative number for an error (e.g. AVERROR(EAGAIN),
if more data is required for it to be able to return a complete packet).
Originally committed as revision 25458 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes roundup issue 2270.
Patch by Robert Schlabbach, robert_s at gmx dot net
Originally committed as revision 25372 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Josh Allmann, joshua dot allmann at gmail, original code
by Ronald S Bultje.
Originally committed as revision 24236 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows very large value strings, needed for xiph extradata.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23859 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used for cleaning up code that is common among RTP depacketizers.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
If these aren't reset, the timestamps make a huge jump when the next RTCP
is received.
Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
In order to sync RTP streams that get their initial RTCP timestamp at
different times, propagate the NTP timestamp of the first RTCP packet
to all other streams.
This makes the timestamps of returned packets start at (near) zero instead
of at any random offset.
Originally committed as revision 22917 to svn://svn.ffmpeg.org/ffmpeg/trunk
the Vorbis / theora depacketizers.
Patch by Josh Allmann <joshua DOT allmann AT gmail DOT com>.
Originally committed as revision 22765 to svn://svn.ffmpeg.org/ffmpeg/trunk
but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.
Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
qualification task, see "RTP/Vorbis payload implementation (GSoC qual
task)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
SDP line handler that parses the streamID in the SDP so that ASF stream
data can be matched to their respective streams in the RTSP demuxer. See
"[PATCH] RTSP-MS 12/15: ASF payload support" thread on mailinglist.
Originally committed as revision 18061 to svn://svn.ffmpeg.org/ffmpeg/trunk
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
Martin Storsjö
Robert Schlabbach
Josh Allmann
Luca Abeni
Stefano Sabatini
Ronald S. Bultje
Alexis Ballier