Commit Graph

373 Commits

Author SHA1 Message Date
Martin Storsjö 3df54c6bf2 rtsp: Add a method parameter to ff_rtsp_read_reply
Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:06:21 +00:00
Martin Storsjö 3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
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
2011-01-01 22:27:16 +00:00
Martin Storsjö 9e99f84f7d rtsp: Check if the rtp stream actually has an RTPDemuxContext
For example MS-RTSP doesn't have RTPDemuxContexts for all streams.

This fixes issue 2448.

Originally committed as revision 26107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-27 09:56:19 +00:00
Martin Storsjö 8c579c1c60 rtsp: Require the transport reply from the server to match the request
This fixes a crash if we requested TCP interleaved transport, but the
server replies with transport data for UDP. According to the RFC, the
server isn't allowed to respond with another transport type than the
one requested.

Originally committed as revision 26077 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-23 15:05:24 +00:00
Martin Storsjö bbd8f5477d rtsp: Don't set the RTP time base from the sample rate if no sample rate is set
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.

The stream that triggered the fix in 26016 still works after this commit.

Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-15 21:06:25 +00:00
Martin Storsjö 86b6e387cc rtsp/rtpdec: Set the AVStream time_base directly in rtsp when it is known
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).

Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.

Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:29:44 +00:00
Martin Storsjö bb776f3b00 rtsp: Parse RealRTSP sample rate declarations from the SDP
The RTP time base can be different from the actual content sample rate.

Originally committed as revision 25907 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:28:45 +00:00
Martin Storsjö 6a7e31a901 rtsp: Look for RTP payload handlers for static payload types, too
Originally committed as revision 25893 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:44 +00:00
Martin Storsjö 003eb64217 rtsp: Factorize code for initializing the rtp payload handler
Originally committed as revision 25892 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:09 +00:00
Martin Storsjö 0b6a7ff4b4 rtsp: Do a forgotten reindenting
Originally committed as revision 25839 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-28 21:17:39 +00:00
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
2010-11-15 15:08:53 +00:00
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
2010-10-29 08:43:57 +00:00
Martin Storsjö c2688f3ac8 rtsp: Move rtsp_setup_output_streams into rtspenc.c
Originally committed as revision 25600 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-29 08:41:49 +00:00
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
2010-10-27 00:42:35 +00:00
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
2010-10-23 16:22:36 +00:00
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
2010-10-23 16:19:53 +00:00
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
2010-10-21 12:25:12 +00:00
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
2010-10-21 12:18:48 +00:00
Martin Storsjö 8bf0f96954 rtsp: Reorder functions
Originally committed as revision 25534 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-21 12:13:02 +00:00
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
2010-10-19 07:38:53 +00:00
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
2010-10-08 08:54:53 +00:00
Martin Storsjö 3d74223025 rtsp: Make rtsp_rtp_mux_open reusable
Originally committed as revision 25409 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-08 08:51:05 +00:00
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
2010-10-08 08:50:29 +00:00
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
2010-10-05 19:46:25 +00:00
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
2010-10-05 11:06:32 +00:00
Diego Biurrun 2e802e3855 Add some #endif comments to ease understanding.
Originally committed as revision 25342 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-05 11:03:48 +00:00
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
2010-10-03 11:56:38 +00:00
Martin Storsjö 6ecd741713 rtsp: Show the received SDP
Originally committed as revision 25322 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-03 11:55:16 +00:00
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
2010-10-01 17:52:26 +00:00
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
2010-10-01 17:50:24 +00:00
Martin Storsjö c690fa97e5 Reindent/rewrap
Originally committed as revision 25291 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:44:53 +00:00
Martin Storsjö 38f8c80b62 rtsp: Reorganize if statements in rtsp_read_play
Originally committed as revision 25290 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:44:18 +00:00
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
2010-10-01 17:43:27 +00:00
Martin Storsjö 96a7c9753e rtsp: Use a dynamically allocated receive buffer
Originally committed as revision 25288 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-10-01 17:41:31 +00:00
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
2010-09-15 17:39:25 +00:00
Ronald S. Bultje 7bac991fd9 Reindent after r25032.
Originally committed as revision 25033 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-09-03 19:26:27 +00:00
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
2010-09-03 19:25:59 +00:00
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
2010-09-03 07:10:21 +00:00
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
2010-08-29 10:25:16 +00:00
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
2010-08-29 10:16:54 +00:00
Martin Storsjö 7934b15d5a Handle IPv6 in the RTSP code
Originally committed as revision 24925 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:29 +00:00
Martin Storsjö 3fbd12d109 Handle IPv6 in the SDP demuxer
Originally committed as revision 24924 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:00 +00:00
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
2010-08-25 13:42:17 +00:00
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
2010-08-12 13:39:38 +00:00
Aurelien Jacobs be73ba2fa4 get rid of MAX_STREAMS limit in RTSP
Originally committed as revision 24752 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-09 23:00:13 +00:00
Reinhard Tartler 2901cc9a95 Fix spelling in comment(s)
Originally committed as revision 24737 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-07 14:11:43 +00:00
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
2010-08-07 11:16:07 +00:00
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
2010-08-06 10:26:30 +00:00
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
2010-07-30 12:04:27 +00:00
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
2010-07-28 09:26:15 +00:00
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
2010-07-21 17:27:28 +00:00
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
2010-07-12 10:17:20 +00:00
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
2010-06-27 14:16:46 +00:00
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
2010-06-25 08:02:50 +00:00
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
2010-06-25 08:00:05 +00:00
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
2010-06-25 07:58:38 +00:00
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
2010-06-25 07:56:45 +00:00
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
2010-06-22 14:15:00 +00:00
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
2010-06-21 19:41:02 +00:00
Martin Storsjö 10ed37b5d1 RTSP: Add the auth credentials to the HTTP tunnel URL, too
Originally committed as revision 23651 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-19 21:57:45 +00:00
Martin Storsjö 6217b6451a RTSP: Set the connection handles to null after closing them
This fixes a potential issue when doing redirects.

Originally committed as revision 23649 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-19 21:46:39 +00:00
Josh Allmann 00e4a1f4e2 RTSP: Don't store the connection handles in local variables
This removes some useless copying of handles, and simplifies error handling.

Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23648 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-19 21:36:13 +00:00
Martin Storsjö d3f84dfc0e RTSP: Clean up rtsp_hd on failure
Since rtsp_hd isn't assigned to rt->rtsp_hd until after the setup phase,
the initialized URLContext could be leaked on failures.

Originally committed as revision 23643 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-18 17:54:56 +00:00
Martin Storsjö 48e77473e9 Cosmetics: Change connexion to connection in code comments
Originally committed as revision 23601 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-14 09:09:59 +00:00
Josh Allmann afcea58c53 RTSP: Shrink SDP fmtp parsing buffer size
Since the parsing of Vorbis/Theora fmtp headers is handled by the
parse_sdp_a_line function pointer now, the buffer in sdp_parse_fmtp
doesn't need to be this large any longer.

Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23599 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-14 08:23:59 +00:00
Josh Allmann 41874d0a5d Reindent
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23598 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-14 08:12:40 +00:00
Josh Allmann f5d33f5241 Add RTSP tunneling over HTTP
Patch by Josh Allmann, joshua dot allmann at gmail dot com

Originally committed as revision 23536 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-08 12:40:34 +00:00
Martin Storsjö fc490fcf71 Cosmetics: Reindent/align/wrap
Originally committed as revision 23498 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-05 19:49:55 +00:00
Josh Allmann d0382374b7 RTSP: Propagate errors up from ff_rtsp_send_cmd*
Patch by Josh Allmann, joshua dot allmann at gmail

Originally committed as revision 23497 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-05 19:45:46 +00:00
Martin Storsjö c453d1bb8c Remove unused local variables
Originally committed as revision 23496 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-05 19:43:57 +00:00
Josh Allmann b8c2c41d66 RTSP: Add a second URLContext for outgoing messages
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
2010-06-05 19:41:43 +00:00
Martin Storsjö 8d168a9207 Fix a crash when opening WMS RTSP streams
Fixes issue 1948

Originally committed as revision 23181 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-05-19 09:46:29 +00:00
Stefano Sabatini 2ef6c1242a Mark av_metadata_set() as deprecated, and use av_metadata_set2()
in its place.

av_metadata_set() is going to be dropped at the next major bump.

Originally committed as revision 22961 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-25 14:27:42 +00:00
Martin Storsjö 5948f82227 Reset RTCP timestamps after seeking, add range start offset to the packets timestamps
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
2010-04-20 07:38:52 +00:00
Martin Storsjö 2cab6b48ad Revert svn rev 21857, readd first_rtcp_ntp_time in RTPDemuxContext
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
2010-04-20 07:34:28 +00:00
Ramiro Polla adef229efb AVERROR(FF_NETERROR(x)) -> FF_NETERROR(x)
FF_NETERROR is implicitly an AVERROR.

Originally committed as revision 22888 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-16 00:20:11 +00:00
Ronald S. Bultje 4aecee7fc3 Fix compile error on mingw where ETIMEDOUT is missing (because it's a WSA error).
This patch also changes FF_NETERROR() to be an AVERROR(), i.e. it is always
negative, whereas it was previously positive.

Originally committed as revision 22887 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-15 18:27:27 +00:00
Martin Storsjö 3370289a4c Zero-initialize the reply struct
The status_code field is read in the fail codepath, where it could be
read uninitialized earlier. Found by clang.

Originally committed as revision 22801 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-04 21:59:06 +00:00
Martin Storsjö 0e64218889 Remove a redundant assignment, found by clang
Originally committed as revision 22790 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-03 12:16:33 +00:00
Sam Gerstein f3c68c5b45 ETIME -> ETIMEDOUT. Patch by Sam Gerstein <sgerstein bluefinlab com>.
Originally committed as revision 22785 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-02 20:14:55 +00:00
Josh Allmann 339f5f3957 Merge Vorbis / Theora depayloaders.
Patch by Josh Allmann <joshua DOT allmann AT gmail DOT com>.

Originally committed as revision 22768 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-01 21:43:22 +00:00
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
2010-03-30 23:30:55 +00:00
Benoit Fouet 32e543f866 Replace @returns by @return.
Originally committed as revision 22729 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 15:50:57 +00:00
Reimar Döffinger c2bfd81605 Some spelling fixes.
Originally committed as revision 22720 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-29 19:17:49 +00:00
Sam Gerstein 9cba6f5f40 Add a timeout to the select() call. Patch by Sam Gerstein <sgerstein bluefinlab
com>.

Originally committed as revision 22718 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-29 17:36:08 +00:00
Martin Storsjö 4bc5cc2313 Reassemble the RTSP URL before replacing hostname with the numerical IP
Originally committed as revision 22681 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 22:21:09 +00:00
Martin Storsjö 7b4a36450b Simplify ff_rtsp_send_cmd_with_content_async, remove an unnecessary buffer
Originally committed as revision 22680 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 22:04:41 +00:00
Martin Storsjö 30af077942 Don't force basic auth in RTSP, but retry with the server-specified method on failure
Originally committed as revision 22678 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 21:49:43 +00:00
Martin Storsjö 2626308abb Actually parse the auth headers in RTSP
Originally committed as revision 22677 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 21:48:58 +00:00
Martin Storsjö aa8bf2fb80 Make RTSP use the generic http authentication code
Still hardcoded to use Basic auth, without parsing the reply headers

Originally committed as revision 22676 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 21:47:33 +00:00
Martin Storsjö b17d11c632 Add separate method/url parameters to the rtsp_send_cmd functions
Originally committed as revision 22675 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 21:46:14 +00:00
Martin Storsjö e9fea405a7 Reindent
Originally committed as revision 22672 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 19:47:26 +00:00
Martin Storsjö b1cc5540e7 Make ff_rtsp_send_cmd simply call ff_rtsp_send_cmd_with_content
Originally committed as revision 22663 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-24 23:06:58 +00:00
Luca Barbato 7ed8211b3e Issue a warning if the received CSeq isn't the expected one
Originally committed as revision 22661 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-24 22:38:48 +00:00
Martin Storsjö 3032276b18 Handle errors returned from ff_rtsp_read_reply in udp_read_packet properly
Originally committed as revision 22657 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-24 18:52:27 +00:00
Martin Storsjö 7a033e08ea Handle multiple RTSP transport options properly by adding all of them into the mask
Originally committed as revision 22644 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-23 08:00:37 +00:00
Martin Storsjö 602eb77975 Parse options in the RTSP URL only from the last question mark onwards
This helps if the URL (erroneously?) contains question marks within the path.

Originally committed as revision 22643 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-23 07:59:23 +00:00
Martin Storsjö 2a21adf924 Reconstruct the RTSP URL, in order to remove the auth part from the URL sent to the server
Don't modify the user-specified s->filename at all, keep all modifications
locally and in rt->control_uri.

Originally committed as revision 22642 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-23 07:55:15 +00:00
Martin Storsjö 685e76b554 Reindent
Originally committed as revision 22635 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-22 15:07:36 +00:00
Martin Storsjö b7dc88fc68 Add support for TCP as lower transport in the RTSP muxer
Originally committed as revision 22634 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-22 15:07:05 +00:00