* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Handle unicode file names on windows
rtp: Rename the open/close functions to alloc/free
Lowercase all ff* program names.
Refer to ff* tools by their lowercase names.
NOT Pulled Replace more FFmpeg instances by Libav or ffmpeg.
Replace `` by $() syntax in shell scripts.
patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled Remove stray libavcore and _g binary references.
vorbis: Rename decoder/encoder files to follow general file naming scheme.
aacenc: Fix whitespace after last commit.
cook: Fix small typo in av_log_ask_for_sample message.
aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
Remove RDFT dependency from AAC decoder.
Add some debug log messages to AAC extradata
Fix mov debug (u)int64_t format strings.
bswap: use native types for av_bwap16().
doc: FLV muxing is supported.
applehttp: Handle AES-128 encrypted streams
Add a protocol handler for AES CBC decryption with PKCS7 padding
doc: Mention that DragonFly BSD requires __BSD_VISIBLE set
Conflicts:
ffplay.c
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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
