b20359f51a
rtsp: Return AVERROR_EOF when all streams have received an RTCP BYE packet
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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
a1ba71aace
rtsp: Check the RTCP file handle for new packets, too
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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
7934b15d5a
Handle IPv6 in the RTSP code
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Originally committed as revision 24925 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:29 +00:00
3fbd12d109
Handle IPv6 in the SDP demuxer
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Originally committed as revision 24924 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 15:32:00 +00:00
2401660d2f
rtsp: Return EOF if the TCP control channel is closed
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Originally committed as revision 24920 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-25 13:42:17 +00:00
27014bf5a3
Send OPTIONS request at a regular basis to standard RTSP servers as well,
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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
be73ba2fa4
get rid of MAX_STREAMS limit in RTSP
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Originally committed as revision 24752 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-09 23:00:13 +00:00
2901cc9a95
Fix spelling in comment(s)
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Originally committed as revision 24737 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-08-07 14:11:43 +00:00
91af5601c1
Add RTP packetization of Theora and Vorbis
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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
d93fdcbf5c
Preserve status reason
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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
965a3ddb1f
Remove mostly unnecessary rtpdec_*.h files, store the declarations in one file
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Originally committed as revision 24596 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-30 12:04:27 +00:00
2845006608
rtsp: Move the definition of SDP_MAX_SIZE up, use it in the RTSP muxer, too
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Originally committed as revision 24571 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-07-28 09:26:15 +00:00
354b757300
Zero-initialize structs/arrays with {0} instead of {}, which isn't proper C99
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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
bf55cf19ca
Report when a method gets an error status code
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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
f3bfe388b5
Make ff_url_split() public
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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
ca937a5508
RTSP, rtpdec: Move RTPPayloadData into rtpdec_mpeg4 and remove all references to rtp_payload_data in rtpdec and rtsp
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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
7fc8ac7fd8
RTSP: Move more SDP/FMTP stuff from rtsp.c to rtpdec_mpeg4.c
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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
9b3788efc3
RTSP: Decouple MPEG-4 and AAC specific parts from rtsp.c
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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
30619e6e59
RTSP: Remove skip_spaces in favor of strspn
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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
9290f15d00
Make the http protocol open the connection immediately in http_open again
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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
a8ead3322f
RTSP: Use the same authentication for the HTTP POST session as for the GET
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Originally committed as revision 23686 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-21 19:41:02 +00:00
10ed37b5d1
RTSP: Add the auth credentials to the HTTP tunnel URL, too
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Originally committed as revision 23651 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-19 21:57:45 +00:00
6217b6451a
RTSP: Set the connection handles to null after closing them
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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
00e4a1f4e2
RTSP: Don't store the connection handles in local variables
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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
d3f84dfc0e
RTSP: Clean up rtsp_hd on failure
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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
48e77473e9
Cosmetics: Change connexion to connection in code comments
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Originally committed as revision 23601 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-14 09:09:59 +00:00
afcea58c53
RTSP: Shrink SDP fmtp parsing buffer size
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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
41874d0a5d
Reindent
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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
f5d33f5241
Add RTSP tunneling over HTTP
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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
fc490fcf71
Cosmetics: Reindent/align/wrap
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Originally committed as revision 23498 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-05 19:49:55 +00:00
d0382374b7
RTSP: Propagate errors up from ff_rtsp_send_cmd*
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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
c453d1bb8c
Remove unused local variables
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Originally committed as revision 23496 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-06-05 19:43:57 +00:00
b8c2c41d66
RTSP: Add a second URLContext for outgoing messages
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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
8d168a9207
Fix a crash when opening WMS RTSP streams
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Fixes issue 1948
Originally committed as revision 23181 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-05-19 09:46:29 +00:00
2ef6c1242a
Mark av_metadata_set() as deprecated, and use av_metadata_set2()
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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
5948f82227
Reset RTCP timestamps after seeking, add range start offset to the packets timestamps
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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
2cab6b48ad
Revert svn rev 21857, readd first_rtcp_ntp_time in RTPDemuxContext
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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
adef229efb
AVERROR(FF_NETERROR(x)) -> FF_NETERROR(x)
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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
4aecee7fc3
Fix compile error on mingw where ETIMEDOUT is missing (because it's a WSA error).
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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
3370289a4c
Zero-initialize the reply struct
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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
0e64218889
Remove a redundant assignment, found by clang
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Originally committed as revision 22790 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-03 12:16:33 +00:00
f3c68c5b45
ETIME -> ETIMEDOUT. Patch by Sam Gerstein <sgerstein bluefinlab com>.
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Originally committed as revision 22785 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-04-02 20:14:55 +00:00
339f5f3957
Merge Vorbis / Theora depayloaders.
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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
72415b2adb
Define AVMediaType enum, and use it instead of enum CodecType, which
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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
32e543f866
Replace @returns by @return.
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Originally committed as revision 22729 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 15:50:57 +00:00
c2bfd81605
Some spelling fixes.
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Originally committed as revision 22720 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-29 19:17:49 +00:00
9cba6f5f40
Add a timeout to the select() call. Patch by Sam Gerstein <sgerstein bluefinlab
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com>.
Originally committed as revision 22718 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-29 17:36:08 +00:00
4bc5cc2313
Reassemble the RTSP URL before replacing hostname with the numerical IP
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Originally committed as revision 22681 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 22:21:09 +00:00
7b4a36450b
Simplify ff_rtsp_send_cmd_with_content_async, remove an unnecessary buffer
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Originally committed as revision 22680 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 22:04:41 +00:00
30af077942
Don't force basic auth in RTSP, but retry with the server-specified method on failure
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Originally committed as revision 22678 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-25 21:49:43 +00:00
Josh Allmann
Martin Storsjö
Ronald S. Bultje
Aurelien Jacobs
Reinhard Tartler
Luca Barbato
Axel Holzinger
Måns Rullgård
Stefano Sabatini
Ramiro Polla
Sam Gerstein
Benoit Fouet
Reimar Döffinger