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Commit Graph

163 Commits

Author SHA1 Message Date
Michael Niedermayer
53bf6b155c Merge commit 'e3ec6fe7bb2a622a863e3912181717a659eb1bad'
* commit 'e3ec6fe7bb2a622a863e3912181717a659eb1bad':
  rtsp: Add a buffer_size option

Conflicts:
	libavformat/rtsp.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2015-04-01 21:34:20 +02:00
Luca Barbato
e3ec6fe7bb rtsp: Add a buffer_size option
And forward it to rtp and udp.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2015-04-01 14:26:35 +02:00
Gilles Chanteperdrix
4438d1c6ed rtsp: parse lang attribute in SDP
Signed-off-by: Martin Storsjö <martin@martin.st>
2015-02-21 23:37:24 +02:00
Gilles Chanteperdrix
c7ad1f562b avformat/rtsp: parse lang attribute in SDP
Reviewed-by: Thomas Volkert <silvo@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-02-13 00:49:08 +01:00
Michael Niedermayer
c9791925a1 Merge commit '8b2e9636c57b22582143467a8a06b509b47b92f9'
* commit '8b2e9636c57b22582143467a8a06b509b47b92f9':
  rtsp: Support tls-encapsulated RTSP

Conflicts:
	libavformat/rtsp.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2014-10-10 21:18:41 +02:00
Luca Barbato
8b2e9636c5 rtsp: Support tls-encapsulated RTSP 2014-10-10 16:29:06 +02:00
Andrey Utkin
bc764d786f Add "prefer_tcp" flag to "rtsp_flags"
If set, and if TCP is available as RTSP RTP transport, then TCP will be
tried first as RTP transport.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-03-04 22:54:13 +01:00
Michael Niedermayer
1295377f0a Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtspenc: Make sure BYE packets are sent before TEARDOWN

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-11-01 19:40:20 +01:00
Martin Storsjö
50aef03b24 rtspenc: Make sure BYE packets are sent before TEARDOWN
Also make sure the BYE packets are sent at all when using
TCP interleaved transport.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-11-01 09:57:06 +02:00
Michael Niedermayer
20904518e9 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  sdp: Add an option for sending RTCP packets to the source of the last packets

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-08-14 12:42:44 +02:00
Martin Storsjö
b56fc18b20 sdp: Add an option for sending RTCP packets to the source of the last packets
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.

With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-14 11:21:33 +03:00
Michael Niedermayer
870f506cfe Merge commit '1f57d60129b0e297cd197c6031c4439b30a6b503'
* commit '1f57d60129b0e297cd197c6031c4439b30a6b503':
  rtsp: Support RFC4570 (source specific multicast) more properly.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-30 11:57:43 +02:00
Ed Torbett
1f57d60129 rtsp: Support RFC4570 (source specific multicast) more properly.
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-29 22:58:56 +03:00
Michael Niedermayer
4835332537 Merge commit '36fb0d02a1faa11eaee51de01fb4061ad6092af9'
* commit '36fb0d02a1faa11eaee51de01fb4061ad6092af9':
  rtsp: Support multicast source filters (RFC 4570)
  rtpproto: Check the source IP if one single source has been specified
  rtpproto: Support IGMPv3 source specific multicast inclusion

Conflicts:
	libavformat/rtpproto.c
	libavformat/rtsp.c
	libavformat/rtsp.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-20 10:39:53 +02:00
Ed Torbett
36fb0d02a1 rtsp: Support multicast source filters (RFC 4570)
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-07-19 12:02:13 +03:00
Ed Torbett
7203dbde39 avformat/rt*p: Joining a SSM multicast group using an SDP (Issue #2171)
Passes Source-Specific Multicast parameters read from an sdp file through to the UDP socket code,
allowing source-specific multicast streams to be correctly received. As an integral part of this
change, additional checking (currently only enabled in the case of SSM streams, but probably
useful in similar scenarios) has been added to the RTP protocol handler to distinguish UDP packets
arriving from multiple sources to the same port and process only the expected packets
(those transmitted from the expected UDP source address). This resolves an issue identified
when multiple instances of FFmpeg subscribe to different Source-Specific Multicast streams
but with each sharing the same destination port.

Signed-off-by: Edward Torbett <ed.torbett@simulation-systems.co.uk>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-18 18:01:31 +02:00
Carl Eugen Hoyos
0fff7f039c Supply a User-Agent header when opening rtsp streams.
Some rtsp servers like the IP Cam IcyBox IB-CAM2002 need it.
Fixes ticket #2761.
Reported, analyzed and tested by trac user imavra.
2013-07-11 23:05:53 +02:00
Michael Niedermayer
0678c388ba rtsp: add option to set the socket timeout of the lower protocol.
Fixes Ticket2294

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-08 17:45:13 +02:00
Michael Niedermayer
b52925d2cd Merge commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05'
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
  lavf: Add a protocol for SRTP encryption/decryption
  rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)

Conflicts:
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-01-15 16:05:34 +01:00
Martin Storsjö
424da30830 rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.

If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-15 11:54:40 +02:00
Michael Niedermayer
34c1c08c66 Merge commit '86d9181cf41edc3382bf2481f95a2fb321058689'
* commit '86d9181cf41edc3382bf2481f95a2fb321058689':
  rtpdec: Support sending RTCP feedback packets

Conflicts:
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-01-09 11:48:14 +01:00
Martin Storsjö
86d9181cf4 rtpdec: Support sending RTCP feedback packets
This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.

This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).

The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.

The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-08 17:48:14 +02:00
Michael Niedermayer
8d0b2aae71 Merge commit 'e96406eda4f143f101bd44372f7b2d542183000a'
* commit 'e96406eda4f143f101bd44372f7b2d542183000a':
  rtsp: Add support for depacketizing RTP data via custom IO

Conflicts:
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-01-04 13:23:19 +01:00
Martin Storsjö
e96406eda4 rtsp: Add support for depacketizing RTP data via custom IO
To use this, set sdpflags=custom_io to the sdp demuxer. During
the avformat_open_input call, the SDP is read from the AVFormatContext
AVIOContext (ctx->pb) - after the avformat_open_input call,
during the av_read_frame() calls, the same ctx->pb is used for reading
packets (and sending back RTCP RR packets).

Normally, one would use this with a read-only AVIOContext for the
SDP during the avformat_open_input call, then close that one and
replace it with a read-write one for the packets after the
avformat_open_input call has returned.

This allows using the RTP depacketizers as "pure" demuxers, without
having them tied to the libavformat network IO.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-03 15:15:27 +02:00
Michael Niedermayer
e3a91c51f7 Merge commit 'c3e15f7b39aac2012f09ee4ca86d2bc674ffdbd4'
* commit 'c3e15f7b39aac2012f09ee4ca86d2bc674ffdbd4':
  rtpdec: Don't pass a non-AVClass pointer as log context
  rtsp: Update a comment to the current filename scheme
  avcodec: handle AVERROR_EXPERIMENTAL
  avutil: Add AVERROR_EXPERIMENTAL
  avcodec: prefer decoders without CODEC_CAP_EXPERIMENTAL

Conflicts:
	doc/APIchanges
	ffmpeg.c
	libavcodec/utils.c
	libavformat/rtpdec.c
	libavutil/error.c
	libavutil/error.h
	libavutil/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-22 14:39:12 +02:00
Martin Storsjö
e0d5ac6ae3 rtsp: Update a comment to the current filename scheme
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-22 01:46:10 +03:00
Michael Niedermayer
81ff0c24ef Merge commit '1cd432e167b1a80853760c89a33606e2b5f229c2'
* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
  configure: fix libcdio check
  rtsp: Allow setting the reordering buffer size via an AVOption
  rtsp: Vertically align a constant definition
  rtp: Update the check for distinguishing between RTP and RTCP
  aac: fix build with hardcoded tables
  fate: dependencies for screen codec tests
  riff: Move functions around to be covered by appropriate #ifdefs

Conflicts:
	configure
	tests/fate/screen.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-19 13:58:14 +02:00
Martin Storsjö
3f055f8f5f rtsp: Allow setting the reordering buffer size via an AVOption
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-18 23:10:48 +03:00
Martin Storsjö
1c37744963 rtsp: Vertically align a constant definition
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-10-18 23:10:42 +03:00
Michael Niedermayer
9f088a1ed4 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mpegvideo: reduce excessive inlining of mpeg_motion()
  mpegvideo: convert mpegvideo_common.h to a .c file
  build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
  Move MASK_ABS macro to libavcodec/mathops.h
  x86: move MANGLE() and related macros to libavutil/x86/asm.h
  x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
  aacdec: Don't fall back to the old output configuration when no old configuration is present.
  rtmp: Add message tracking
  rtsp: Support mpegts in raw udp packets
  rtsp: Support receiving plain data over UDP without any RTP encapsulation
  rtpdec: Remove an unused include
  rtpenc: Remove an av_abort() that depends on user-supplied data
  vsrc_movie: discourage its use with avconv.
  avconv: allow no input files.
  avconv: prevent invalid reads in transcode_init()
  avconv: rename OutputStream.is_past_recording_time to finished.

Conflicts:
	configure
	doc/filters.texi
	ffmpeg.c
	ffmpeg.h
	libavcodec/Makefile
	libavcodec/aacdec.c
	libavcodec/mpegvideo.c
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-08-09 19:31:56 +02:00
Martin Storsjö
1243c72251 rtsp: Support mpegts in raw udp packets
This is basically the same way as mpegts packets are parsed in
rtpdec.c.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-09 00:25:57 +03:00
Martin Storsjö
df8cf076c8 rtsp: Support receiving plain data over UDP without any RTP encapsulation
EvoStream Media Server can serve data in this format, and
VLC/live555 already supports it.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-08-09 00:25:15 +03:00
Michael Niedermayer
9ca27df52f Merge remote-tracking branch 'qatar/master'
* qatar/master:
  configure: Check for the math function rint
  TechSmith Screen Codec 2 decoder
  rtsp: Add listen mode
  rtsp: Make rtsp_open_transport_ctx() non-static
  rtsp: Move rtsp_read_close
  rtsp: Parse the mode=receive/record parameter in transport lines

Conflicts:
	Changelog
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-11 23:57:11 +02:00
Jordi Ortiz
a8ad6ffafe rtsp: Add listen mode
This makes the RTSP demuxer act as a server, listening for an
incoming connection.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 22:00:28 +03:00
Jordi Ortiz
6e71c1202b rtsp: Make rtsp_open_transport_ctx() non-static
This is required for the upcoming RTSP listen mode.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 21:21:19 +03:00
Jordi Ortiz
45b068580b rtsp: Parse the mode=receive/record parameter in transport lines
We need to support the nonstandard mode=receive, for compatibility
with older libavformat clients.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-07-10 21:20:04 +03:00
Michael Niedermayer
61930bd0d7 Merge remote-tracking branch 'qatar/master'
* qatar/master: (27 commits)
  libxvid: Give more suitable names to libxvid-related files.
  libxvid: Separate libxvid encoder from libxvid rate control code.
  jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
  fate: cosmetics: lowercase some comments
  fate: Give more consistent names to some RealVideo/RealAudio tests.
  lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
  lavfi: add extended_data to AVFilterBuffer.
  lavc: check that extended_data is properly set in avcodec_encode_audio2().
  lavc: pad last audio frame with silence when needed.
  samplefmt: add a function for filling a buffer with silence.
  samplefmt: add a function for copying audio samples.
  lavr: do not try to copy to uninitialized output audio data.
  lavr: make avresample_read() with NULL output discard samples.
  fate: split idroq audio and video into separate tests
  fate: improve dependencies
  fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
  fate: split some combined tests into separate audio and video tests
  fate: fix dependencies for probe tests
  mips: intreadwrite: fix inline asm for gcc 4.8
  mips: intreadwrite: remove unnecessary inline asm
  ...

Conflicts:
	cmdutils.h
	configure
	doc/APIchanges
	doc/filters.texi
	ffmpeg.c
	ffplay.c
	libavcodec/internal.h
	libavcodec/jpeglsdec.c
	libavcodec/libschroedingerdec.c
	libavcodec/libxvid.c
	libavcodec/libxvid_rc.c
	libavcodec/utils.c
	libavcodec/version.h
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/buffersink.h
	tests/Makefile
	tests/fate/aac.mak
	tests/fate/audio.mak
	tests/fate/demux.mak
	tests/fate/ea.mak
	tests/fate/image.mak
	tests/fate/libavutil.mak
	tests/fate/lossless-audio.mak
	tests/fate/lossless-video.mak
	tests/fate/microsoft.mak
	tests/fate/qt.mak
	tests/fate/real.mak
	tests/fate/screen.mak
	tests/fate/video.mak
	tests/fate/voice.mak
	tests/fate/vqf.mak
	tests/ref/fate/ea-mad
	tests/ref/fate/ea-tqi

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-10 02:25:41 +02:00
Jordi Ortiz
fcd0298c05 rtsp: Add content-type message header parsing
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2012-05-08 10:18:35 -07:00
Michael Niedermayer
feb997577b Merge remote-tracking branch 'qatar/master'
* qatar/master: (25 commits)
  riff: fix invalid av_freep() calls on EOF in ff_read_riff_info
  pam: Fix a typo that broke writing and reading PAM files.
  mxfdec: fix memleak on av_realloc failures
  mxfdec: Do not parse slices or DeltaEntryArrays.
  mxfdec: hybrid demuxing/seeking solution
  mxfdec: Add Avid's essence element key.
  mfxdec: Separate mxf_essence_container_uls for audio and video.
  mxfdec: Compute packet offsets properly.
  mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack.
  mxfdec: use av_dlog() for 'no corresponding source package found'
  mxfdec: Make mxf->partitions sorted by offset.
  mxfdec: parse ThisPartition
  mxfdec: Speed up metadata and index parsing.
  mxfdec: Make sure DataDefinition is consistent between material track and source track.
  mxfdec: add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
  mxfdec: Add hack that adjusts the n_delta calculation when system items are present.
  mxfdec: Parse IndexTableSegments and convert them into AVIndexEntry arrays.
  mxfdec: Move FooterPartition to MXFContext and make sure it is never zero.
  mxfdec: check return value of avio_seek
  mxfdec: skip to end of structural sets
  ...

Conflicts:
	configure
	libavcodec/pnm.c
	libavformat/mxfdec.c
	libavformat/riff.c
	libavformat/rtsp.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-23 01:05:20 +01:00
Martin Storsjö
dbb06b8c0d rtsp: Allow specifying the UDP port range via AVOptions
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-22 01:10:02 +02:00
Michael Niedermayer
b5a69e79c5 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtpdec: Use our own SSRC in the SDES field when sending RRs
  Finalize changelog for 0.8 Release
  Prepare for 0.8 Release
  threads: change the default for threads back to 1
  threads: update slice_count and slice_offset from user context
  aviocat: Remove useless includes
  doc/APIChanges: fill in missing dates and hashes
  Revert "avserver: fix build after the next bump."
  mpegaudiodec: switch error detection check to AV_EF_BUFFER
  lavf: rename fer option and document resulting (f_)err_detect options
  lavc: rename err_filter option to err_detect and document it
  mpegvideo: fix invalid memory access for small video dimensions
  movenc: Reorder entries in the MOVIentry struct, for tigheter packing
  rtsp: Remove extern declarations for variables that don't exist
  aviocat: Flush the output before closing

Conflicts:
	Changelog
	RELEASE
	libavcodec/mpegaudiodec.c
	libavcodec/pthread.c
	libavformat/options.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-21 23:11:27 +01:00
Martin Storsjö
1f712e6a05 rtsp: Remove extern declarations for variables that don't exist
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-21 01:12:07 +02:00
Dmitry Volyntsev
b29ec002bb fix rtp 916 issue
Reviewed-by: Luca Abeni <lucabe72@email.it>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-18 16:36:38 +01:00
Michael Niedermayer
8bc7fe4daf Merge remote-tracking branch 'qatar/master'
* qatar/master:
  doxygen: misc consistency, spelling and wording fixes
  vcr1: drop unnecessary emms_c() calls without MMX code
  Replace all uses of av_close_input_file() with avformat_close_input().
  lavf: add avformat_close_input().
  lavf: deprecate av_close_input_stream().
  lavf doxy: add some basic demuxing documentation.
  lavf doxy: add some general lavf information.
  lavf doxy: add misc utility functions to a group.
  lavf doxy: add av_guess_codec/format to the encoding group.
  lavf doxy: add core functions to a doxy group.
  Add basic libavdevice documentation.
  lavc: convert error_recognition to err_recognition.
  avconv: update -map option help text
  x86: Require 7 registers for the cabac asm
  x86: bswap: remove test for bswap instruction
  bswap: make generic implementation more compiler-friendly
  h264: remove useless cast
  proresdec: fix decode_slice() prototype

Conflicts:
	configure
	doc/APIchanges
	ffprobe.c
	libavcodec/avcodec.h
	libavcodec/celp_math.h
	libavcodec/h264.c
	libavfilter/src_movie.c
	libavformat/anm.c
	libavformat/avformat.h
	libavformat/version.h
	libavutil/avstring.h
	libavutil/bswap.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-13 00:39:48 +01:00
Diego Biurrun
58c42af722 doxygen: misc consistency, spelling and wording fixes 2011-12-12 23:06:23 +01:00
Michael Niedermayer
988f585fcb Merge remote-tracking branch 'qatar/master'
* qatar/master: (44 commits)
  replacement Indeo 3 decoder
  gsm demuxer: do not allocate packet twice.
  flvenc: use first packet delay as global delay.
  ac3enc: doxygen update.
  imc: return error codes instead of 0 for error conditions.
  imc: return meaningful error codes instead of -1
  imc: do not set channel layout for stereo
  imc: validate channel count
  imc: check for ff_fft_init() failure
  imc: check output buffer size before decoding
  imc: use DSPContext.bswap16_buf() to byte-swap packet data
  rtsp: add allowed_media_types option
  libgsm: add flush function to reset the decoder state when seeking
  libgsm: simplify decoding by using a loop
  gsm: log error message when packet is too small
  libgsmdec: do not needlessly set *data_size to 0
  gsmdec: do not needlessly set *data_size to 0
  gsmdec: add flush function to reset the decoder state when seeking
  libgsmdec: check output buffer size before decoding
  gsmdec: log error message when output buffer is too small.
  ...

Conflicts:
	Changelog
	ffplay.c
	libavcodec/indeo3.c
	libavcodec/mjpeg_parser.c
	libavcodec/vp3.c
	libavformat/cutils.c
	libavformat/id3v2.c
	libavutil/parseutils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-03 02:16:26 +01:00
John Brooks
f011fcd67e rtsp: add allowed_media_types option
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>
2011-11-02 21:37:46 +02:00
Michael Niedermayer
fae714a9fb Merge remote-tracking branch 'qatar/master'
* qatar/master:
  avconv: add presets
  rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
  rtsp: Make the rtsp flags avoptions set via a define
  rtpenc: Set a default video codec
  avoptions: Fix av_opt_flag_is_set
  rtp: Fix ff_rtp_get_payload_type
  doc: Update the documentation on setting options for RTSP
  rtsp: Remove the separate filter_source variable
  rtsp: Accept options via private avoptions instead of URL options
  rtsp: Simplify AVOption definitions
  rtsp: Merge the AVOption lists
  lavfi: port libmpcodecs delogo filter
  lavfi: port boxblur filter from libmpcodecs
  lavfi: add negate filter
  lavfi: add LUT (LookUp Table) generic filters
  AVOptions: don't segfault on NULL parameter in av_set_options_string()
  avio: Check for invalid buffer length.
  mpegenc/mpegtsenc: add muxrate private options.
  lavf: deprecate AVFormatContext.file_size
  mov: add support for TV metadata atoms tves, tvsn and stik

Conflicts:
	Changelog
	doc/filters.texi
	doc/protocols.texi
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/avfilter.h
	libavfilter/formats.c
	libavfilter/internal.h
	libavfilter/vf_boxblur.c
	libavfilter/vf_delogo.c
	libavfilter/vf_lut.c
	libavformat/mpegtsenc.c
	libavformat/utils.c
	libavformat/version.h
	libavutil/opt.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-18 01:54:40 +02:00
Martin Storsjö
9867aea524 rtsp: Remove the separate filter_source variable
Read it as a flag from the flags field instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:49 +03:00
Martin Storsjö
eca4850c6d rtsp: Accept options via private avoptions instead of URL options
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>
2011-10-17 19:57:48 +03:00