FFmpeg

Commit Graph

Author	SHA1	Message	Date
Martin Storsjö	0b6a7ff4b4	rtsp: Do a forgotten reindenting Originally committed as revision 25839 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	dd22cfb101	rtsp: Parse and use the Content-Base reply header, if present This fixes playing RTSP urls with query parameters. Originally committed as revision 25755 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	0526c6f7c7	rtsp: Split out the RTSP demuxer functions to a separate, new file Originally committed as revision 25601 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	c2688f3ac8	rtsp: Move rtsp_setup_output_streams into rtspenc.c Originally committed as revision 25600 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	47bfe49c64	rtsp: Add stub declarations of the setup_in/output_streams functions This may be needed to avoid calls to implicitly defined functions (that will be removed by dead code elimination later anyway). Originally committed as revision 25585 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Aurelien Jacobs	a5cea13202	drop rtsp_default_protocols which is not part of public API and not used anymore Originally committed as revision 25557 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Aurelien Jacobs	67f34aaa97	use rtp_get_local_rtp_port() instead of the deprecated rtp_get_local_port() Originally committed as revision 25554 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	eced8fa02e	rtsp: Move the rtsp_probe function to the demuxer code block This function is only used by the RTSP demuxer. Originally committed as revision 25537 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	44b70ce563	rtsp: Untangle the dependencies between the RTSP/SDP demuxers and RTSP muxer This allows compilation of one of them without requiring the others' dependencies to be present. Originally committed as revision 25535 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	8bf0f96954	rtsp: Reorder functions Originally committed as revision 25534 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	44594cc798	Add a demuxer for receiving raw rtp:// URLs without an SDP description The demuxer inspects the payload type of a received RTP packet and handles the cases where the content is fully described by the payload type. Originally committed as revision 25527 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	a493f80a2c	rtsp: Factorize out code for opening a chained RTP muxer The new object file is added to the SDP demuxer in the makefile, since it is needed in both the RTSP muxer and demuxer and in the SDP demuxer, due to the current code coupling. Originally committed as revision 25410 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	3d74223025	rtsp: Make rtsp_rtp_mux_open reusable Originally committed as revision 25409 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	9e6acc7884	rtsp: Remove the start_time field from RTSPState, use AVFormatContext->start_time_realtime instead Originally committed as revision 25408 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	5fe8021a6a	rtsp/sdp: Move code into correct ifdefs This makes the code dependencies correct. Previously, the SDP demuxer wasn't buildable on its own. This also reverts rev 25343. Originally committed as revision 25354 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Diego Biurrun	a44da176ac	Remove some pointless CONFIG_RTSP_DEMUXER #ifdefs. They reside within a large CONFIG_RTSP_DEMUXER block and are thus pointless. Originally committed as revision 25343 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Diego Biurrun	2e802e3855	Add some #endif comments to ease understanding. Originally committed as revision 25342 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	d7810f4541	rtsp: In the muxer, show the generated with verbose log level It is only useful for debugging, so it doesn't have to be shown every time. Originally committed as revision 25323 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	6ecd741713	rtsp: Show the received SDP Originally committed as revision 25322 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	321259c1ab	rtsp: Return a queued packet if it has been in the queue for longer than max_delay Originally committed as revision 25295 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	58ee09911e	rtpdec: Reorder received RTP packets according to the seq number Reordering is enabled only when receiving over UDP. Originally committed as revision 25294 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	c690fa97e5	Reindent/rewrap Originally committed as revision 25291 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	38f8c80b62	rtsp: Reorganize if statements in rtsp_read_play Originally committed as revision 25290 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	ad4ad27fb6	rtsp/rtpdec: Allow rtp_parse_packet to take ownership of the packet buffer 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	14 years ago
Martin Storsjö	96a7c9753e	rtsp: Use a dynamically allocated receive buffer Originally committed as revision 25288 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	160918d588	rtsp: Handle standard assigned codec names for private payload types, too Originally committed as revision 25126 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Ronald S. Bultje	7bac991fd9	Reindent after r25032. Originally committed as revision 25033 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
John Wimer	619298a84d	Send NAT punching messages to the address specified in the Transport: 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	14 years ago
Martin Storsjö	744a882f6c	rtsp: 10l, try to update the correct rtp stream This fixes a bug from rev 22917. Now RTSP streams where the individual RTCP sender reports aren't sent at the same time actually are synced properly. Originally committed as revision 25029 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Josh Allmann	b20359f51a	rtsp: Return AVERROR_EOF when all streams have received an RTCP BYE packet Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 24965 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Josh Allmann	a1ba71aace	rtsp: Check the RTCP file handle for new packets, too Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 24962 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	7934b15d5a	Handle IPv6 in the RTSP code Originally committed as revision 24925 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	3fbd12d109	Handle IPv6 in the SDP demuxer Originally committed as revision 24924 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	2401660d2f	rtsp: Return EOF if the TCP control channel is closed Originally committed as revision 24920 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Ronald S. Bultje	27014bf5a3	Send OPTIONS request at a regular basis to standard RTSP servers as well, this prevents a time-out which closes the TCP connection and kills our session. see "Re: [FFmpeg-devel] [PATCH] rtsp.c: keep-alive" thread on mailinglist. Originally committed as revision 24785 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Aurelien Jacobs	be73ba2fa4	get rid of MAX_STREAMS limit in RTSP Originally committed as revision 24752 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Reinhard Tartler	2901cc9a95	Fix spelling in comment(s) Originally committed as revision 24737 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Josh Allmann	91af5601c1	Add RTP packetization of Theora and Vorbis Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 24735 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Luca Barbato	d93fdcbf5c	Preserve status reason It is used to provide meaningful error messages. Originally committed as revision 24714 to svn://svn.ffmpeg.org/ffmpeg/trunk	14 years ago
Martin Storsjö	965a3ddb1f	Remove mostly unnecessary rtpdec_*.h files, store the declarations in one file Originally committed as revision 24596 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Martin Storsjö	2845006608	rtsp: Move the definition of SDP_MAX_SIZE up, use it in the RTSP muxer, too Originally committed as revision 24571 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Axel Holzinger	354b757300	Zero-initialize structs/arrays with {0} instead of {}, which isn't proper C99 Patch by Axel Holzinger, aholzinger at gmx dot de Originally committed as revision 24391 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Luca Barbato	bf55cf19ca	Report when a method gets an error status code That makes easier understand what went wrong. In debug mode the whole reply gets printed. Originally committed as revision 24212 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Måns Rullgård	f3bfe388b5	Make ff_url_split() public ff_url_split() is retained as an alias, as it was used by ffserver, to avoid breaking ABI compatibility with it. Originally committed as revision 23822 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Josh Allmann	ca937a5508	RTSP, rtpdec: Move RTPPayloadData into rtpdec_mpeg4 and remove all 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	15 years ago
Josh Allmann	7fc8ac7fd8	RTSP: Move more SDP/FMTP stuff from rtsp.c to rtpdec_mpeg4.c Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23770 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Josh Allmann	9b3788efc3	RTSP: Decouple MPEG-4 and AAC specific parts from rtsp.c Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23769 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Josh Allmann	30619e6e59	RTSP: Remove skip_spaces in favor of strspn Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23768 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Martin Storsjö	9290f15d00	Make the http protocol open the connection immediately in http_open again Also make the RTSP protocol use url_alloc and url_connect instead of relying on the delay open behaviour. Originally committed as revision 23710 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago
Martin Storsjö	a8ead3322f	RTSP: Use the same authentication for the HTTP POST session as for the GET Originally committed as revision 23686 to svn://svn.ffmpeg.org/ffmpeg/trunk	15 years ago

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514 Commits (2124a97a4998413c7e81539b16b01ef6ac570ea9)