FFmpeg

Commit Graph

Author	SHA1	Message	Date
Wei-Cheng Pan	f646cd4471	rtp: Make ff_rtp_codec_id() case insensitive Fixes handling of lower case pcmu Signed-off-by: Michael Niedermayer <michaelni@gmx.at>	11 years ago
Diego Biurrun	6c1a7d07eb	Use proper "" quotes for local header #includes	12 years ago
Martin Storsjö	932117171f	rtp: Make sure the output format pointer is set Not sure if this actually happens, but we do the same check when checking payload_type further above in the function, so it might be needed. Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Martin Storsjö	e90820d4f8	rtp: Make sure priv_data is set before reading it This fixes crashes with muxing H263 into RTSP. CC: libav-stable@libav.org Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Martin Storsjö	c44784c9bb	rtp: Rename a static variable to normal naming conventions Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Martin Storsjö	58b5971881	rtp: Cosmetic cleanup Remove leftover debug comments, fix brace placement and add whitespace, remove unnecessary and weirdly placed braces. Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Martin Storsjö	a925f723a9	rtp: Don't read priv_data unless it is allocated This makes all users of rtpenc_chain (rtsp muxer, sapenc, mov rtp hinting) work again, broken since `8034130e0`. Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Luca Barbato	8034130e06	rtp: set the payload type as stream id Support multiple video/audio streams with different format in the same session. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>	12 years ago
Anton Khirnov	36ef5369ee	Replace all CODEC_ID_* with AV_CODEC_ID_*	12 years ago
Adriano Pallavicino	999c63e4ca	rtp: Only choose static payload types if the sample rate and channels are right If using a different sample rate or number of channels, use a dynamic payload type instead, where the parameters are passed in the SDP. G722 is a special case where the normal rules don't apply. Signed-off-by: Martin Storsjö <martin@martin.st>	12 years ago
Mohamed Naufal Basheer	55c3a4f617	G.723.1 demuxer and decoder Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>	12 years ago
Martin Storsjö	c4584f3c1f	rtpenc: Allow packetizing H263 according to the old RFC 2190 According to newer RFCs, this packetization scheme should only be used for interfacing with legacy systems. Implementing this packetization mode properly requires parsing the full H263 bitstream to find macroblock boundaries (and knowing their macroblock and gob numbers and motion vector predictors). This implementation tries to look for GOB headers (which can be inserted by using -ps <small number>), but if the GOBs aren't small enough to fit into the MTU, the packetizer blindly splits packets at any offset and claims it to be a GOB boundary (by using Mode A from the RFC). While not correct, this seems to work with some receivers. Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Martin Storsjö	2e69dd66b6	rtp: Fix ff_rtp_get_payload_type It was broken in `3b3ea34655` "Remove all uses of deprecated AVOptions API", where any presence of a payload_type AVOption caused its value to be returned, even if it wasn't set (and thus had the default -1 value). This caused the RTP muxer to be broken. Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Anton Khirnov	3b3ea34655	Remove all uses of deprecated AVOptions API.	13 years ago
Mohamed Naufal Basheer	f990dc374e	Add the G723.1 demuxer and decoder	13 years ago
Rafaël Carré	1430ae44e8	rtp: Simplify ff_rtp_get_payload_type Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Rafaël Carré	9152880e95	rtpenc: Add a payload type private option Specifying the payload type is useful when the type number has already been negotiated before creating the stream, for example in SIP protocol. Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Rafaël Carré	0c378ea1f7	rtp: factorize dynamic payload type fallback Move the identical code in rtp_write_header() and ff_sdp_write_media() inside ff_rtp_get_payload_type() Signed-off-by: Luca Barbato <lu_zero@gentoo.org>	13 years ago
Mans Rullgard	2912e87a6c	Replace FFmpeg with Libav in licence headers Signed-off-by: Mans Rullgard <mans@mansr.com>	14 years ago
Martin Storsjö	0048a2a8d3	Handle G.722 in RTP, and all the exceptions mandated in RFC 3551 Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Stefano Sabatini	72415b2adb	Define AVMediaType enum, and use it instead of enum CodecType, which is deprecated and will be dropped at the next major bump. Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Luca Abeni	4bf0faaafe	Remove the inclusion of unneeded headers Originally committed as revision 21152 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Stefano Sabatini	9106a698e7	Rename bitstream.h to get_bits.h. Originally committed as revision 18494 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Luca Abeni	215037887d	Do not return payload type 34 for H.263 (it is deprecated) Originally committed as revision 18346 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Luca Abeni	bf6d981806	Remame rtp_get_codec_info() to ff_rtp_get_codec_info(), as it is not a static function Originally committed as revision 17390 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Luca Abeni	0550b58f4e	Rename rtp_get_payload_type() to ff_rtp_get_payload_type(), as it is not a static function Originally committed as revision 17364 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Luca Abeni	20631a9c15	Merge rtp_internal.h in rtp.h, and remove rtp_internal.h Originally committed as revision 16817 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Diego Biurrun	406792e7b0	cosmetics: Remove pointless period after copyright statement non-sentences. Originally committed as revision 16684 to svn://svn.ffmpeg.org/ffmpeg/trunk	16 years ago
Luca Abeni	309d32b0db	Do not set sample_rate = 90000 for mp2 and mp3 audio over RTP Originally committed as revision 13943 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Diego Biurrun	245976da2a	Use full path for #includes from another directory. Originally committed as revision 13098 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	2ccf0a4290	Add a comment about missing entries Originally committed as revision 12646 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	87cb064359	Use the correct size for the enc_name field (removing the arbitrary "50" size) Originally committed as revision 12645 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	19faa9f469	Remove useless entries from AVRtpPayloadTypes Originally committed as revision 12644 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	e4ed1fbf91	Support mp3 audio in the RTP muxer Originally committed as revision 12643 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Diego Pettenò	7d51edddd4	Make AVRtpPayloadTypes static and constant Patch by Diego 'Flameeyes' Pettenò (flameeyes AT gmail DOT com) Originally committed as revision 11432 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	83a0d3878c	Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	8eb793c459	Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies Originally committed as revision 11406 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Diego Biurrun	d0b53d05c8	Fix some spelling mistakes. Originally committed as revision 11125 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Björn Axelsson	899681cd1d	Use dynamically allocated ByteIOContext in AVFormatContext patch by: Björn Axelsson, bjorn d axelsson a intinor d se thread: [PATCH] Remove static ByteIOContexts, 06 nov 2007 Originally committed as revision 11071 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	db628029c4	Add MPEG2 support to the RTP muxer Originally committed as revision 11047 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	7ed19d7fbf	Remove the "AVRtpPayloadTypes[i].pt == i" assumption from RTP and RTSP code (this is needed for supporting MPEG2 video in the RTP muxer) Originally committed as revision 11046 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	18c05a375b	Do not send too many RTCP packets (according to RFC 3550, the minimum RTCP interval should be 5 seconds) Originally committed as revision 10930 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	0aa7a2e690	Use a symbolic name for the payload size of an RTCP Sender Report packet Originally committed as revision 10929 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	e0d21bfe83	Allow to set the maximum number of frames per RTP packet (and add support for this in the AAC packetizer) Originally committed as revision 10647 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	d0c3be9568	Fix a warning by removing an useless assignment (buf_ptr should be only used in the RTP muxer, and not in the demuxer) Originally committed as revision 10561 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	171dce486c	Support for AAC streaming over RTP. Fragmentation is not implemented yet Originally committed as revision 10491 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	af74c95a08	Fix timestamps in RTP packets (now, MPEG1 video with B frames works correctly) Originally committed as revision 10469 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	1b31b02ed1	Properly set RTP and NTP timestamps in RTCP SR packets Originally committed as revision 10468 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Luca Abeni	98561024ac	Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c) Originally committed as revision 10201 to svn://svn.ffmpeg.org/ffmpeg/trunk	17 years ago
Panagiotis Issaris	6f3e0b2174	Replace all occurrences of AVERROR_IO with AVERROR(EIO). Originally committed as revision 9760 to svn://svn.ffmpeg.org/ffmpeg/trunk	18 years ago

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110 Commits (591c0c26b5986cacfc74ab58347f7ac4a8d841ed)