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>
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>
* commit 'e96406eda4f143f101bd44372f7b2d542183000a':
rtsp: Add support for depacketizing RTP data via custom IO
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
The data does not contain timing or trailing line breaks anymore. In
addition to being less idiotic, it is consistent with other codecs and
thus allows more switches between formats and codecs. It also fixes the
issue of the trailing line returns being simple \n instead of CRLF in
the ASS rectangle dialogue (this is the reason of the FATE update).
* qatar/master:
rtmp: Add support for limelight authentication
rtmp: Add support for adobe authentication
Conflicts:
Changelog
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Limelight is a not too uncommon CDN. The authentication scheme is
pretty similar to the adobe authentication, but is even closer to
normal http digest authentication (but not close enough to warrant
sharing code) than the adobe version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly used to authenticate the client when publishing.
Tested with wowza and akamai.
Some but not all servers support resending a new connect invoke
within the same connection, so always reconnect for sending a new
connection attempt. This matches what other applications do as well.
The authentication scheme is structurally pretty similar to http
digest authentication, but uses base64 instead of hex strings.
Signed-off-by: Martin Storsjö <martin@martin.st>
Current MicroDVD AVPackets contain timing information and trailing line
breaks. The data is now only composed of the markup data. Doing this
consistently between text subtitles decoders allows to use different
codec for various formats. For instance, MicroDVD markup is sometimes
found in some VPlayer files. Also, generally speaking, the subtitles
text decoders have no use of these timings (and they must not use them
since it would break any user timing adjustment).
Technically, this is a major ABI break. In practice, a mismatching
lavf/lavc will now error out for MicroDVD decoding. Supporting both
formats requires unnecessary complex and fragile code.
FATE needs update because line breaks in the ASS file were "\n" (because
that's what is used in the original file). ASS format expect "\r\n" line
breaks; this commit fixes this issue. Also note that this "\r\n"
trailing need to be moved at some point from the decoders to the ASS
muxer.
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
The new options reset the timestamps at each new segment, so that the
generated segments will have timestamps starting close to 0.
It is meant to address trac ticket #1425.
Gif demuxer is capable of extracting multiple frames from gif file.
In conjunction with gif decoder it implements support for reading
animated gifs.
Demuxer has two options available to user: default_delay and min_delay.
These options are for protection from too rapid gif animations. In practice
it is standard approach to slow down rendering of this kind of gifs. If you try to
play gif with delay between frames of one hundredth of second (100fps) using
one of major web browsers, you get significantly slower playback,
around 10 fps. This is because browser detects that delay value is less than some
threshold (usually 2 hundredths of second) and reset it to default value (usually 10
hundredths of second, which corresponds to 10fps). Manipulating these options user
can achieve the same effect during conversion to some video format. Otherwise user
can set them to not protect from rapid animations at all.
The other case when these options necessary is for gif images encoded according to
gif87a standard since prior to gif89a there was no delay information included in file.
Bump lavf minor version.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
* commit 'b522000e9b2ca36fe5b2751096b9a5f5ed8f87e6':
avio: introduce avio_closep
mpegtsenc: set muxing type notification to verbose
vc1dec: Use correct spelling of "opposite"
a64multienc: change mc_frame_counter to unsigned
arm: call arm-specific rv34dsp init functions under if (ARCH_ARM)
svq1: Drop a bunch of useless parentheses
parseutils-test: do not print numerical error codes
svq1: K&R formatting cosmetics
Conflicts:
doc/APIchanges
libavcodec/svq1dec.c
libavcodec/svq1enc.c
libavformat/version.h
libavutil/parseutils.c
tests/ref/fate/parseutils
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dwt: Drop unused functions spatial_compose{53|97}i()
nutdec: Remove unused and broken debug function stub
avcodec: Drop long-deprecated imgconvert.h header
Add Opus support to the Ogg muxer.
Add Opus codec id and codec description.
avformat: Identify anonymous AVIO typedef structs.
Conflicts:
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/imgconvert.h
libavcodec/version.h
libavformat/oggenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
nutdec: const correctness for get_v_trace/get_s_trace function arguments
truemotion2: Request samples for old TM2 headers
rtpdec: Remove a useless ff_ prefix from a static symbol
rtpdec: Support depacketizing speex
rtpenc: Add support for packetizing speex
Conflicts:
libavformat/rtpdec.c
libavformat/sdp.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This packetization scheme simply places the full packets into the
RTP packet without any extra header bytes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtp: Packetization of JPEG (RFC 2435)
smoothstreamingenc: Copy the SAR on the AVStreams as well
Conflicts:
Changelog
libavformat/rtpenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3f7fd59d151a2773f0e2e93e56b6b13ec6e5334b':
avformat: fix typo in avformat_close_input
mp3enc: write Xing TOC
mp3enc: support MPEG-2 and MPEG-2.5 in Xing header.
mp3enc: downgrade some errors in writing Xing frame to warnings
lavf: flush the output AVIOContext in av_write_trailer().
lavf: cosmetics, reformat av_write_trailer().
avio: flush the internal buffer in avio_close()
Enhance doc on asyncts audiofilter
cmdutils: avoid setting data pointers to invalid values in alloc_buffer()
libavcodec: remove av_destruct_packet_nofree()
Conflicts:
libavcodec/avpacket.c
libavformat/mp3enc.c
libavformat/nutenc.c
libavformat/utils.c
libavformat/version.h
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '07584eaf4a95db3f11d3bc411f9786932829e82b':
mpegts: check substreams before discarding
Add a smooth streaming segmenter muxer
file: Add an avoption for disabling truncating existing files on open
img2dec: always close AVIOContexts
rtpdec_jpeg: Error out on other unsupported type values as well
rtpdec_jpeg: Disallow using the reserved q values
rtpdec_jpeg: Fold the default qtables case into an existing if statement
rtpdec_jpeg: Store and reuse old qtables for q values 128-254
rtpdec_jpeg: Simplify the calculation of the number of qtables
rtpdec_jpeg: Add more comments about the fields in the SOF0 section
rtpdec_jpeg: Clarify where the subsampling magic numbers come from
rtpdec_jpeg: Don't use a bitstream writer for the EOI marker
rtpdec_jpeg: Don't needlessly use a bitstream writer for the header
rtpdec_jpeg: Simplify writing of the jpeg header
rtpdec_jpeg: Merge two if statements
rtpdec_jpeg: Write the DHT section properly
Conflicts:
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This muxer splits the output from the ismv muxer into individual
files, in realtime.
The same can also be done by the standalone tool ismindex, but this
muxer is needed for doing it in realtime (especially for live
streams that need extra handling for updating the lookahead fields
in the fragment headers).
Using this muxer, one can deliver live smooth streaming from a
normal static file web server. (Using ismindex, one can deliver
premade smooth streaming files from a static file web server,
or prepare files for serving with IIS.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow to specify options affecting the segment list generation.
In particular: add +live and +cache flags.
For a full discussion read trac ticket #1642:
http://ffmpeg.org/trac/ffmpeg/ticket/1642
Also add live M3U8 generation example.
* qatar/master:
x86: dsputil: Only compile motion_est code when encoders are enabled
mem: fix typo in check for __ICC
fate: mp3: drop redundant CMP setting
rtp: Depacketization of JPEG (RFC 2435)
Rename ff_put_string to avpriv_put_string
mjpeg: Rename some symbols to avpriv_* instead of ff_*
yadif: cosmetics
Conflicts:
Changelog
libavcodec/mjpegenc.c
libavcodec/x86/Makefile
libavfilter/vf_yadif.c
libavformat/version.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make internal small_strptime() function public, and use it in place of
strptime().
This allows to avoid a dependency on strptime() on systems which do not
support it.
In particular, fix trac ticket #992.
* qatar/master:
libvpxenc: use the default bitrate if not set
utvideo: Rename utvideo.c to utvideodec.c
doc: Fix syntax errors in sample Emacs config
mjpegdec: more meaningful return values
configure: clean up Altivec detection
getopt: Remove an unnecessary define
rtmp: Use int instead of ssize_t
getopt: Add missing includes
rtmp: Add support for receiving incoming streams
Add missing includes for code relying on external libraries
Conflicts:
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/mjpegdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add support for SWFVerification
api-example: use new video encoding API.
x86: avcodec: Appropriately name files containing only init functions
mpegvideo_mmx_template: drop some commented-out cruft
libavresample: add mix level normalization option
w32pthreads: Add missing #includes to make header compile standalone
rtmp: Gracefully ignore _checkbw errors by tracking them
rtmp: Do not send _checkbw calls as notifications
prores: interlaced ProRes encoding
Conflicts:
doc/examples/decoding_encoding.c
libavcodec/proresenc_kostya.c
libavcodec/w32pthreads.h
libavcodec/x86/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifies how the server verifies client SWF files before allowing the
files to connect to an application. Verifying SWF files is a security
measure that prevents someone from creating their own SWF files that can
attempt to stream your resources.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow to override the default 'glob_sequence' value, which is deprecated
in favor of the new 'glob' and 'sequence' options.
The new pattern types should be easier on the user since they are more
predictable than 'glob_sequence', and do not require awkward escaping.
* qatar/master:
flvdec: remove spurious use of stream id
lavf: deprecate r_frame_rate.
lavf: round estimated average fps to a "standard" fps.
Conflicts:
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/flvdec.c
libavformat/rawdec.c
libavformat/utils.c
tests/ref/fate/iv8-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.
An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.
Additional changes by Josh Allmann <joshua.allmann@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.
Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
