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* rtsp: Support mpegts in raw udp packetsMartin Storsjö2012-08-09
| | | | | | | This is basically the same way as mpegts packets are parsed in rtpdec.c. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Support receiving plain data over UDP without any RTP encapsulationMartin Storsjö2012-08-09
| | | | | | | EvoStream Media Server can serve data in this format, and VLC/live555 already supports it. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Add listen modeJordi Ortiz2012-07-10
| | | | | | | This makes the RTSP demuxer act as a server, listening for an incoming connection. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Make rtsp_open_transport_ctx() non-staticJordi Ortiz2012-07-10
| | | | | | This is required for the upcoming RTSP listen mode. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Parse the mode=receive/record parameter in transport linesJordi Ortiz2012-07-10
| | | | | | | We need to support the nonstandard mode=receive, for compatibility with older libavformat clients. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Add content-type message header parsingJordi Ortiz2012-05-08
| | | | Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* rtsp: Allow specifying the UDP port range via AVOptionsMartin Storsjö2012-01-22
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Remove extern declarations for variables that don't existMartin Storsjö2012-01-21
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* doxygen: misc consistency, spelling and wording fixesDiego Biurrun2011-12-12
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* rtsp: add allowed_media_types optionJohn Brooks2011-11-02
| | | | | | | | Streams from RTSP or SDP that do not match an allowed type will be skipped entirely, which allows video-only or audio-only streaming from servers that provide both. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Remove the separate filter_source variableMartin Storsjö2011-10-17
| | | | | | Read it as a flag from the flags field instead. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Accept options via private avoptions instead of URL optionsMartin Storsjö2011-10-17
| | | | | | | | | | | | Eventually, the old way of passing options by adding stuff to the URL can be dropped. This avoids having to tamper with the user-specified URL to pass options on the transport mode. This also works better with redirects, since the options don't need to be parsed out from the URL. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Merge the AVOption listsMartin Storsjö2011-10-17
| | | | | | | | This eases adding options that are common for both. The AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong to the muxer or demuxer. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Parse the x-Accept-Dynamic-Rate headerMartin Storsjö2011-10-12
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: remove disabled codeDiego Biurrun2011-07-18
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* doxygen: Make sure parameter names match between .c and .h files.Diego Biurrun2011-07-14
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* RTSP: Doxygen comment cleanupDiego Biurrun2011-07-03
| | | | | Do not use Doxygen for comments that apply to specific implementation details; merge some duplicated Doxygen comment blocks.
* rtspenc: Add RTP muxer optionsMartin Storsjö2011-06-10
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtspdec: add initial_pause private option.Anton Khirnov2011-05-27
| | | | Deprecate corresponding AVFormatParameters field.
* rtsp: Only do keepalive using GET_PARAMETER if the server supports itMartin Storsjö2011-05-11
| | | | | | | | | | | | | | | | This is more like what VLC does. If the server doesn't mention supporting GET_PARAMETER in response to an OPTIONS request, VLC doesn't send any keepalive requests at all. After this patch, libavformat will still send OPTIONS keepalives if GET_PARAMETER isn't explicitly said to be supported. Some RTSP cameras don't support GET_PARAMETER, and will close the connection if this is sent as keepalive request (but support OPTIONS just fine, but probably don't need any keepalive at all). Some other cameras don't support using OPTIONS as keepalive, but require GET_PARAMETER instead. Signed-off-by: Martin Storsjö <martin@martin.st>
* Replace FFmpeg with Libav in licence headersMans Rullgard2011-03-19
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* avio: rename ByteIOContext to AVIOContext.Anton Khirnov2011-02-20
| | | | Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* os: replace select with pollLuca Barbato2011-01-28
| | | | | Select has limitations on the fd values it could accept and silently breaks when it is reached.
* Make ff_rtsp_send_cmd_with_content_async static to rtsp.c.Diego Elio Pettenò2011-01-25
| | | | Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
* rtspdec: Retry with TCP if UDP failedMartin Storsjö2011-01-24
| | | | Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
* rtsp: Split out a function undoing the setup made by ff_rtsp_make_setup_requestMartin Storsjo2011-01-24
| | | | Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
* rtsp: Make make_setup_request a nonstatic functionMartin Storsjo2011-01-24
| | | | Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
* rtsp: Allow requesting of filtering of source packetsMartin Storsjö2011-01-06
| | | | | | | | | | | | | | | | | If filtered, only packets from the right source address and port are received. To test, play back e.g. some mpeg4 video RTSP stream (where the video stream is the first stream in the presentation) over UDP. While receiving this stream, send another stream to the same port: ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp rtp://127.0.0.1:5000?localport=1234 Normally, the RTSP playback reports lots of errors at this point. If the RTSP stream has the ?filter_src option enabled, these interferring packets are ignored. Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Store the Content-Base header value straight to the targetMartin Storsjö2011-01-02
| | | | | | | This avoids having a large temporary buffer in the struct used for storing the rtsp reply headers. Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Pass the method name to ff_rtsp_parse_lineMartin Storsjö2011-01-02
| | | | Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthStateMartin Storsjö2011-01-02
| | | | | | | | This allows ff_rtsp_parse_line to do more changes directly in RTSPState when parsing the reply, instead of having to store large amounts of temporary data in RTSPMessageHeader. Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Add a method parameter to ff_rtsp_read_replyMartin Storsjö2011-01-02
| | | | Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Parse and use the Content-Base reply header, if presentMartin Storsjö2010-11-15
| | | | | | This fixes playing RTSP urls with query parameters. Originally committed as revision 25755 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Split out the RTSP demuxer functions to a separate, new fileMartin Storsjö2010-10-29
| | | | Originally committed as revision 25601 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Move rtsp_setup_output_streams into rtspenc.cMartin Storsjö2010-10-29
| | | | Originally committed as revision 25600 to svn://svn.ffmpeg.org/ffmpeg/trunk
* drop rtsp_default_protocols which is not part of public API and not used anymoreAurelien Jacobs2010-10-23
| | | | Originally committed as revision 25557 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Remove the start_time field from RTSPState, use ↵Martin Storsjö2010-10-08
| | | | | | AVFormatContext->start_time_realtime instead Originally committed as revision 25408 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Use a dynamically allocated receive bufferMartin Storsjö2010-10-01
| | | | Originally committed as revision 25288 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Send NAT punching messages to the address specified in the Transport:John Wimer2010-09-03
| | | | | | | | message, if available (RFC 2326, section 12.39), fixes issue 2212. Patch by John Wimer <john at god vtic net>. Originally committed as revision 25032 to svn://svn.ffmpeg.org/ffmpeg/trunk
* rtsp: Return AVERROR_EOF when all streams have received an RTCP BYE packetJosh Allmann2010-08-29
| | | | | | Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 24965 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Handle IPv6 in the RTSP codeMartin Storsjö2010-08-25
| | | | Originally committed as revision 24925 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Handle IPv6 in the SDP demuxerMartin Storsjö2010-08-25
| | | | Originally committed as revision 24924 to svn://svn.ffmpeg.org/ffmpeg/trunk
* get rid of MAX_STREAMS limit in RTSPAurelien Jacobs2010-08-09
| | | | Originally committed as revision 24752 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Preserve status reasonLuca Barbato2010-08-06
| | | | | | It is used to provide meaningful error messages. Originally committed as revision 24714 to svn://svn.ffmpeg.org/ffmpeg/trunk
* RTSP, rtpdec: Move RTPPayloadData into rtpdec_mpeg4 and remove all ↵Josh Allmann2010-06-25
| | | | | | | | references to rtp_payload_data in rtpdec and rtsp Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23772 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Cosmetics: Change connexion to connection in code commentsMartin Storsjö2010-06-14
| | | | Originally committed as revision 23601 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Add RTSP tunneling over HTTPJosh Allmann2010-06-08
| | | | | | Patch by Josh Allmann, joshua dot allmann at gmail dot com Originally committed as revision 23536 to svn://svn.ffmpeg.org/ffmpeg/trunk
* Cosmetics: Reindent/align/wrapMartin Storsjö2010-06-05
| | | | Originally committed as revision 23498 to svn://svn.ffmpeg.org/ffmpeg/trunk
* RTSP: Propagate errors up from ff_rtsp_send_cmd*Josh Allmann2010-06-05
| | | | | | Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23497 to svn://svn.ffmpeg.org/ffmpeg/trunk
* RTSP: Add a second URLContext for outgoing messagesJosh Allmann2010-06-05
| | | | | | | Done in preparation for RTSP over HTTP. Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23494 to svn://svn.ffmpeg.org/ffmpeg/trunk