Commit Graph

425 Commits

Author SHA1 Message Date
Michael Niedermayer
612122b187 Merge remote branch 'qatar/master'
* qatar/master: (32 commits)
  10-bit H.264 x86 chroma v loopfilter asm
  Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
  Fix crash of interlaced MPEG2 decoding
  h264pred: fix one more aliasing violation.
  doc/APIchanges: fill in missing hashes and dates.
  flacenc: use proper initializers for AVOption default values.
  lavc: deprecate named constants for deprecated antialias_algo.
  aac: workaround for compilation on cygwin
  swscale: extend YUV422p support to 10bits depth
  tiff: add support for inverted FillOrder for uncompressed data
  Remove unused softfloat implementation.
  h264pred: fix aliasing violations.
  rotozoom: Eliminate French variable name.
  rotozoom: Check return value of fread().
  rotozoom: Return an error value instead of calling exit().
  rotozoom: Make init_demo() return int and check for errors on invocation.
  rotozoom: Drop silly UINT8 typedef.
  rotozoom: Drop some unnecessary parentheses.
  rotozoom: K&R coding style cosmetics
  rtsp: Only do keepalive using GET_PARAMETER if the server supports it
  ...

Conflicts:
	Changelog
	cmdutils.c
	doc/APIchanges
	doc/general.texi
	ffmpeg.c
	ffplay.c
	libavcodec/h264pred_template.c
	libavcodec/resample.c
	libavutil/pixfmt.h
	libavutil/softfloat.c
	libavutil/softfloat.h
	tests/rotozoom.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-12 04:51:24 +02:00
Martin Storsjö
0b4949b518 rtsp: Only do keepalive using GET_PARAMETER if the server supports it
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>
2011-05-11 10:42:34 +03:00
Michael Niedermayer
7b376b398a Merge remote branch 'qatar/master'
* qatar/master:
  Handle unicode file names on windows
  rtp: Rename the open/close functions to alloc/free
  Lowercase all ff* program names.
  Refer to ff* tools by their lowercase names.
NOT Pulled  Replace more FFmpeg instances by Libav or ffmpeg.
  Replace `` by $() syntax in shell scripts.
  patcheck: Allow overiding grep program(s) through environment variables.
NOT Pulled  Remove stray libavcore and _g binary references.
  vorbis: Rename decoder/encoder files to follow general file naming scheme.
  aacenc: Fix whitespace after last commit.
  cook: Fix small typo in av_log_ask_for_sample message.
  aacenc: Finish 3GPP psymodel analysis for non mid/side cases.
  Remove RDFT dependency from AAC decoder.
  Add some debug log messages to AAC extradata
  Fix mov debug (u)int64_t format strings.
  bswap: use native types for av_bwap16().
  doc: FLV muxing is supported.
  applehttp: Handle AES-128 encrypted streams
  Add a protocol handler for AES CBC decryption with PKCS7 padding
  doc: Mention that DragonFly BSD requires __BSD_VISIBLE set

Conflicts:
	ffplay.c
	ffprobe.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-24 03:41:22 +02:00
Martin Storsjö
9261e6cf3f rtp: Rename the open/close functions to alloc/free
This avoids clashes if we internally want to override the global
open function.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-04-24 00:05:37 +03:00
Michael Niedermayer
efb5fa79f5 Merge remote branch 'qatar/master'
* qatar/master: (37 commits)
  In avcodec_open(), set return code to an error value only when an error occurs instead of unconditionally at the start of the function.
  lavc: remove reference to opt.h from Makefile.
  prefer avio_check() over url_exist()
  avio: remove AVIO_* access symbols in favor of new AVIO_FLAG_* symbols
  lavu: remove misc disabled cruft
  lavu: remove FF_API_OLD_IMAGE_NAMES cruft
NOT PULLED  lavu: remove FF_API_OLD_EVAL_NAMES cruft
  lavc: remove misc disabled cruft.
  lavc: remove the FF_API_INOFFICIAL cruft.
  lavc: remove the FF_API_SET_STRING_OLD cruft.
  lavc: remove the FF_API_USE_LPC cruft.
  lavc: remove the FF_API_SUBTITLE_OLD cruft.
  lavc: remove the FF_API_VIDEO_OLD cruft.
  lavc: remove the FF_API_AUDIO_OLD cruft.
  lavc: remove the FF_API_OPT_SHOW cruft.
  lavc: remove the FF_API_MM_FLAGS cruft.
  lavf: remove misc disabled cruft.
  lavf: remove FF_API_INDEX_BUILT cruft
  lavf: remove FF_API_URL_CLASS cruft.
  lavf: remove FF_API_SYMVER cruft
  ...

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-20 04:48:23 +02:00
Stefano Sabatini
59d96941f0 avio: remove AVIO_* access symbols in favor of new AVIO_FLAG_* symbols
Make AVIO_FLAG_ access constants work as flags, and in particular fix
the behavior of functions (such as avio_check()) which expect them to
be flags rather than modes.

This breaks API.
2011-04-19 19:47:58 +02:00
Michael Niedermayer
c88caa522c Merge remote branch 'qatar/master'
* qatar/master:
  proto: include os_support.h in network.h
  matroskaenc: don't write an empty Cues element.
  lavc: add a FF_API_REQUEST_CHANNELS deprecation macro
  avio: move extern url_interrupt_cb declaration from avio.h to url.h
  avio: make av_register_protocol2 internal.
  avio: avio_ prefix for url_set_interrupt_cb.
  avio: AVIO_ prefixes for URL_ open flags.
  proto: introduce listen option in tcp
  doc: clarify configure features
  proto: factor ff_network_wait_fd and use it on udp

Conflicts:
	ffmpeg.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-08 02:50:13 +02:00
Anton Khirnov
f87b1b373a avio: AVIO_ prefixes for URL_ open flags. 2011-04-07 18:07:16 +02:00
Michael Niedermayer
434f248723 Merge remote branch 'qatar/master'
* qatar/master: (22 commits)
  ac3enc: move extract_exponents inner loop to ac3dsp
  avio: deprecate url_get_filename().
  avio: deprecate url_max_packet_size().
  avio: make url_get_file_handle() internal.
  avio: make url_filesize() internal.
  avio: make url_close() internal.
  avio: make url_seek() internal.
  avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
  avio: make url_write() internal.
  avio: make url_read_complete() internal.
  avio: make url_read() internal.
  avio: make url_open() internal.
  avio: make url_connect internal.
  avio: make url_alloc internal.
  applehttp: Merge two for loops
  applehttp: Restructure the demuxer to use a custom AVIOContext
  applehttp: Move finished and target_duration to the variant struct
  aacenc: reduce the number of loop index variables
  avio: deprecate url_open_protocol
  avio: deprecate url_poll and URLPollEntry
  ...

Conflicts:
	libavformat/applehttp.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-05 02:31:56 +02:00
Anton Khirnov
1869ea03b7 avio: make url_get_file_handle() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
e52a9145c8 avio: make url_close() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
925e908bc7 avio: make url_write() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
dce3756459 avio: make url_read_complete() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
bc371aca46 avio: make url_read() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
0589da0aa5 avio: make url_open() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
62eaaeacb5 avio: make url_connect internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov
5652bb9471 avio: make url_alloc internal. 2011-04-04 17:45:19 +02:00
Michael Niedermayer
2cae9809e2 Merge remote branch 'qatar/master'
* qatar/master:
  fate: fix partial run when no samples path is specified
  ARM: NEON fixed-point forward MDCT
  ARM: NEON fixed-point FFT
  lavf: bump minor version and add an APIChanges entry for avio changes
  avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
  avio: make url_fdopen internal.
  avio: make url_open_dyn_packet_buf internal.
  avio: avio_ prefix for url_close_dyn_buf
  avio: avio_ prefix for url_open_dyn_buf
  avio: introduce an AVIOContext.seekable field
  ac3enc: use generic fixed-point mdct
  lavfi: add fade filter
  Change yadif to not use out of picture lines.
  lavc: deprecate AVCodecContext.antialias_algo
  lavc: mark mb_qmin/mb_qmax for removal on next major bump.

Conflicts:
	doc/filters.texi
	libavcodec/ac3enc_fixed.h
	libavcodec/ac3enc_float.h
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/vf_fade.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-04 02:15:12 +02:00
Anton Khirnov
6dc7d80de7 avio: avio_ prefix for url_close_dyn_buf 2011-04-03 22:47:05 +02:00
Ilya
907783f221 Use strtoul to parse rtptime and seq values.
strtol could return negative values, leading to various error messages,
mainly "non-monotonically increasing dts".
2011-03-26 17:13:36 +01:00
Michael Niedermayer
4fa0e24736 Merge remote-tracking branch 'newdev/master'
* newdev/master: (33 commits)
  Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
  Add kbdwin.o to AC3 decoder
  Detect byte-swapped AC-3 and support decoding it directly.
  cosmetics: indentation
  Always copy input data for AC3 decoder.
  ac3enc: make sym_quant() branch-free
  cosmetics: indentation
  Add a CPU flag for the Atom processor.
  id3v2: skip broken tags with invalid size
  id3v2: don't explicitly skip padding
  Make sure kbhit() is in conio.h
  fate: update wmv8-drm reference
  vc1: make P-frame deblock filter bit-exact.
  configure: Add the -D parameter to the dlltool command
  amr: Set the AVFMT_GENERIC_INDEX flag
  amr: Set the pkt->pos field properly to the start of the packet
  amr: Set the codec->bit_rate field based on the last packet
  rtsp: Specify unicast for TCP interleaved streams, too
  Set the correct target for mingw64 dlltool
  applehttp: Change the variable for stream position in seconds into int64_t
  ...

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/ac3dec.c
	libavformat/avio.h
	libavformat/id3v2.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-03-23 02:42:56 +01:00
Martin Storsjö
895678f823 rtsp: Specify unicast for TCP interleaved streams, too
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.

According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-03-21 20:58:33 +01:00
Mans Rullgard
2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Nicolas George
c76374c6db Use AVERROR_EXIT with url_interrupt_cb.
Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.

This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).

Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-03-15 08:09:19 -04:00
Anton Khirnov
22a3212e32 avio: rename url_fopen/fclose -> avio_open/close.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 10:18:55 -05:00
Martin Storsjö
28c4741a66 libavformat: Remove FF_NETERRNO()
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.

This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).

This fixes roundup issue 2614, unbreaking blocking network IO on
windows.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 07:21:31 -05:00
Anton Khirnov
b7effd4e83 avio: avio_ prefixes for get_* functions
In the name of consistency:
get_byte           -> avio_r8
get_<type>         -> avio_r<type>
get_buffer         -> avio_read

get_partial_buffer will be made private later

get_strz is left out becase I want to change it later to return
something useful.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-21 11:23:22 -05:00
Anton Khirnov
e731b8d872 avio: move init_put_byte() to a new private header and rename it
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:31 -05:00
Anton Khirnov
ae628ec1fd avio: rename ByteIOContext to AVIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:15 -05:00
Anton Khirnov
9fcae9735e Replace remaining uses of parse_date with av_parse_time.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-16 23:39:57 +00:00
Martin Storsjö
2c35a6bde9 rtsp: udp_read_packet returning 0 doesn't mean success
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).

This fixes issue 2612.
2011-02-17 00:37:00 +01:00
Martin Storsjö
b2dd842d21 rtsp/rdt: Assign the RTSPStream index to AVStream->id
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.

Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-11 16:58:19 -05:00
Martin Storsjö
b22dbb291d Use avformat_free_context for cleaning up muxers
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:39:55 -05:00
Martin Storsjö
1338dc0823 libavformat: Use avcodec_copy_context for chained muxers
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.

This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:28:07 -05:00
Martin Storsjö
ce41c51b0c Free AVStream->info in chained muxers
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 01:03:31 +01:00
Martin Storsjö
d9c0510e22 rtsp: Don't store RTSPStream in AVStream->priv_data
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.

This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.

Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 00:49:15 +01:00
Luca Barbato
ea7f080749 Free the RTSPStreams in ff_rtsp_close_streams
This plugs a small memory leak

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-02-01 20:40:16 +01:00
Luca Barbato
dfd2a005eb Replace dprintf with av_dlog
dprintf clashes with POSIX.1-2008
2011-01-29 23:55:37 +01:00
Luca Barbato
f81c7ac70a rtsp: make ff_sdp_parse return value forwarded
the sdp demuxer did not forward it at all while the rtsp demuxer assumed
a single kind of error
2011-01-28 15:45:19 +01:00
Luca Barbato
a8475bbdb6 os: replace select with poll
Select has limitations on the fd values it could accept and silently
breaks when it is reached.
2011-01-28 15:45:19 +01:00
Diego Elio Pettenò
c6610a216e Prefix all _demuxer, _muxer, _protocol from libavformat and libavdevice.
This also lists the objects from those two libraries as internal (by adding
the ff_ prefix) so that they can then be hidden via linker scripts.
2011-01-26 22:10:09 +00:00
Diego Elio Pettenò
57c4d01ec9 Make ff_rtsp_send_cmd_with_content_async static to rtsp.c.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-25 22:10:36 +01:00
Martin Storsjö
2762a7a28b rtspdec: Retry with TCP if UDP failed
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:49:36 +01:00
Martin Storsjo
aeb2de1c82 rtsp: Use ff_rtsp_undo_setup in the cleanup code in ff_rtsp_make_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:43 +01:00
Martin Storsjo
93e7490ee0 rtsp: Split out a function undoing the setup made by ff_rtsp_make_setup_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:39 +01:00
Martin Storsjo
fef5649a82 rtsp: Make make_setup_request a nonstatic function
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:36 +01:00
Martin Storsjö
a3b058b7ba rtsp: Properly fail if unable to open an input RTP port
Originally committed as revision 26285 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-09 10:47:53 +00:00
Martin Storsjö
a92c30d76e rtsp: Allow requesting of filtering of source packets
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
2011-01-06 15:22:58 +00:00
Martin Storsjö
29db7c3af4 rtsp: Parse RTP-Info headers
Originally committed as revision 26236 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-05 21:23:42 +00:00
Martin Storsjö
d2995eb910 rtsp: Store the Content-Base header value straight to the target
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
2011-01-02 10:11:12 +00:00
Martin Storsjö
77223c5388 rtsp: Pass the method name to ff_rtsp_parse_line
Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:10:12 +00:00
Martin Storsjö
acc9ed1450 rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthState
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
2011-01-02 10:07:56 +00:00
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