* qatar/master:
lavf: bump minor and add an APIChanges entry for avformat cleanup
lavf: get rid of ffm-specific stuff in avformat.h
Not pulled: avio: deprecate av_protocol_next().
avio: add a function for iterating though protocol names.
lavf: rename a parameter of av_sdp_create from buff->buf
lavf: rename avf_sdp_create to av_sdp_create.
lavf: make av_guess_image2_codec internal
avio: make URLProtocol internal.
avio: make URLContext internal.
lavf: mark av_pkt_dump(_log) for remove on $next+1 bump.
lavf: use designated initializers for all protocols
applehttp: don't use deprecated url_ functions.
avio: move two ff_udp_* functions from avio_internal to url.h
asfdec: remove a forgotten declaration of nonexistent function
avio: deprecate the typedef for URLInterruptCB
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.
Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
Do the same change for ff_rdt_parse_packet, too, to keep the interfaces
similar.
Originally committed as revision 25289 to svn://svn.ffmpeg.org/ffmpeg/trunk
This will be used for cleaning up code that is common among RTP depacketizers.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
If these aren't reset, the timestamps make a huge jump when the next RTCP
is received.
Originally committed as revision 22918 to svn://svn.ffmpeg.org/ffmpeg/trunk
In order to sync RTP streams that get their initial RTCP timestamp at
different times, propagate the NTP timestamp of the first RTCP packet
to all other streams.
This makes the timestamps of returned packets start at (near) zero instead
of at any random offset.
Originally committed as revision 22917 to svn://svn.ffmpeg.org/ffmpeg/trunk
but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.
Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
what e.g. RealPlayer does. This allows proper port forwarding setup in NAT-
based environments.
Patch by Martin Storsjö <$firstname at $firstname dot st>.
Originally committed as revision 21856 to svn://svn.ffmpeg.org/ffmpeg/trunk
associated with the I/O handle (e.g. the fd returned by open()). See
"[RFC] rtsp.c EOF support" thread.
There were previously some URI-specific implementations of the same idea,
e.g. rtp_get_file_handles() and udp_get_file_handle(). All of these are
deprecated by this patch and will be removed at the next major API bump.
Originally committed as revision 17779 to svn://svn.ffmpeg.org/ffmpeg/trunk
under review. See "[FFmpeg-devel] RTP mark bit not passed to parse_packet"
thread on mailinglist.
Originally committed as revision 17616 to svn://svn.ffmpeg.org/ffmpeg/trunk
in common except for this one value. Change was requested by Luca in the
"[FFmpeg-devel] RTP mark bit not passed to parse_packet" thread.
Originally committed as revision 17615 to svn://svn.ffmpeg.org/ffmpeg/trunk
