This allows getting rid of quite a bit of boilerplate in depacketizers.
The default value (initializing need_parsing to 0, aka
AVSTREAM_PARSE_NONE) is the same as it is initialized to by default
in AVStream.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'cdcc370293a159c321e41af7f0eef141c62d698d':
rtsp: punch holes again after pause
See: 22bb5bd7a3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.
Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '604c9b1196c70d79bbbc1f23e75f6a8253a74da3':
rtsp: move the CONFIG_ macros to the beginning of the check
Conflicts:
libavformat/rtsp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, AVERROR(EIO) was returned. Now the value is passed from
lower level, thus it is possible to distinguish ECONNREFUSED, ETIMEDOUT,
ENETUNREACH etc.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The were wrongly being exported and used by libavdevice
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f797b134cad4d248b1c8955659997980d0668bc3':
rtpenc_chain: Don't copy the time base to the source stream by default
See: 1fe40e1b05
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.
This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.
This reverts abb810db but achieves the same effect for the other
muxers.
Signed-off-by: Martin Storsjö <martin@martin.st>
For muxing, it accepts
both 0 and AV_NOPTS_VALUE. For demuxing, it will present
AV_NOPTS_VALUE when start_time_realtime is unknown.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
* commit 'f4d371b9737c0405b3bc46d7ca0c856c0a8616b1':
rtsp: Don't include the listen flag in the SDP demuxer flags
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It's only relevant for the RTSP demuxer. Similarly, the custom_io
flag is only present in the SDP demuxer options list.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'b7e6da988bfd5def40ccf3476eb8ce2f98a969a5':
rtpproto: Move rtpproto specific function declarations to a separate header
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '1f57d60129b0e297cd197c6031c4439b30a6b503':
rtsp: Support RFC4570 (source specific multicast) more properly.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
* 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>
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>
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>
* commit '1dd1b2332ebbac710d8e0214cec7595e118f2105':
rtsp: Include an User-Agent header field in all requests
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b3ea76624ad1baab0b6bcc13f3f856be2f958110':
vf_aspect: use the name 's' for the pointer to the private context
Remove commented-out debug #define cruft
Conflicts:
libavcodec/4xm.c
libavcodec/dvdsubdec.c
libavcodec/ituh263dec.c
libavcodec/mpeg12.c
libavfilter/avfilter.c
libavfilter/vf_aspect.c
libavfilter/vf_fieldorder.c
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e926b5ceb1962833f0c884a328382bc2eca67aff':
avformat: Drop unnecessary ff_ name prefixes from static functions
Conflicts:
libavformat/audiointerleave.c
libavformat/mux.c
libavformat/mxfenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
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>
* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
rtsp: Remove an outdated comment
rtsp: Remove references to weirdly named variables in other files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is unclear what the bug exactly was and if it ever was fixed,
and we don't even support decoding via faad any longer. The
comment has been present since d0deedcb in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
One of them is renamed now, but mentioning it by name serves
no purpose here. The other table mentioned ceased to exist
under that name in 4934884a1 in 2006.
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 '8729698d50739524665090e083d1bfdf28235724':
rtsp: Recheck the reordering queue if getting a new packet
lavr: log channel conversion description for any-to-any functions
lavr: mix: reduce the mixing matrix when possible
lavr: cosmetics: reindent
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If we timed out and consumed a packet from the reordering queue,
but didn't return a packet to the caller, recheck the queue status.
Otherwise, we could end up in an infinite loop, trying to consume
a queued packet that has already been consumed.
CC: libav-stable@libav.org
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>
* commit '3f95f0dda55fca74b646937095a02a8fa9776622':
rtpdec: Move the URLContext used for RTCP RR out from the context, to a parameter
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>
* qatar/master:
bgmc: Fix av_malloc checks in ff_bgmc_init()
rtp: set the payload type as stream id
Conflicts:
libavformat/rtpenc_chain.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '381dc1a5ec0925b281c573457c413ae643567086':
fate: ac3: Place E-AC-3 tests and AC-3 tests in different groups
fate: Add shorthands for acodec PCM and ADPCM tests
avconv: Drop unused function argument from do_video_stats()
cmdutils: Conditionally compile libswscale-related bits
aacenc: Drop some unused function arguments
rtsp: Avoid a cast when calling strtol
nut: support textual data
nutenc: verbosely report unsupported negative pts
Conflicts:
cmdutils.c
ffmpeg.c
libavformat/nut.c
libavformat/nutenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This gets rid of this warning:
libavformat/rtsp.c: In function ‘rtsp_parse_transport’:
libavformat/rtsp.c:794: warning: cast discards qualifiers from pointer target type
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'c9ef43215c7d68c2cdcdbe02287aa114f27a32ed':
fate-vc1: add dependencies
ARM: fix overreads in neon h264 chroma mc
rtsp: Make sure the ret variable is initialized in ff_rtsp_fetch_packet
gitignore: ignore files created by msvc
fate: Add proper dependencies for the tests in video.mak
configure: Disable Snow decoder and encoder by default
lzo: Drop obsolete fast_memcpy reference
build: Drop OBJS declaration for non-existing PCM_DVD encoder
mpeg4videodec: Disable frame multithreading for GMC, its not implemented at all
Conflicts:
libavcodec/mpegvideo.c
libavformat/rtsp.c
tests/fate/microsoft.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* commit 'd58dd4b5b5d31cfd4092e38a5f2c894eee2ab078':
avopt: Store defaults for AV_OPT_TYPE_FLAGS in the i64 union member
Conflicts:
libavcodec/libvpxenc.c
libavcodec/options_table.h
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '124134e42455763b28cc346fed1d07017a76e84e':
avopt: Store defaults for AV_OPT_TYPE_CONST in the i64 union member
Conflicts:
libavcodec/aacenc.c
libavcodec/libopenjpegenc.c
libavcodec/options_table.h
libavdevice/bktr.c
libavdevice/v4l2.c
libavdevice/x11grab.c
libavfilter/af_amix.c
libavfilter/vf_drawtext.c
libavformat/movenc.c
libavformat/options_table.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vf_hqdn3d: Don't declare the loop variable within the for loop
huffyuv: update to current coding style
huffman: update to current coding style
rtsp: Free the rtpdec context properly
build: fft: x86: Drop unused YASM-OBJS-FFT- variable
Conflicts:
libavcodec/huffman.c
libavcodec/huffyuv.c
libavcodec/x86/Makefile
libavfilter/vf_hqdn3d.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegvideo_enc: don't use deprecated avcodec_encode_video().
cmdutils: refactor -codecs option.
avconv: make -shortest a per-output file option.
lavc: add avcodec_descriptor_get_by_name().
lavc: add const to AVCodec* function parameters.
swf(dec): replace CODEC_ID with AV_CODEC_ID
dvenc: don't use deprecated AVCODEC_MAX_AUDIO_FRAME_SIZE
rtmpdh: Do not generate the same private key every time when using libnettle
rtp: remove ff_rtp_get_rtcp_file_handle().
rtsp.c: use ffurl_get_multi_file_handle() instead of ff_rtp_get_rtcp_file_handle()
avio: add (ff)url_get_multi_file_handle() for getting more than one fd
h264: vdpau: fix crash with unsupported colorspace
amrwbdec: Decode the fr_quality bit properly
Conflicts:
Changelog
cmdutils.c
cmdutils_common_opts.h
doc/ffmpeg.texi
ffmpeg.c
ffmpeg.h
ffmpeg_opt.c
libavcodec/h264.c
libavcodec/options.c
libavcodec/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
avformat: Drop pointless "format" from container long names
swscale: bury one more piece of inline asm under HAVE_INLINE_ASM.
wv: K&R formatting cosmetics
configure: Add missing descriptions to help output
h264_ps: declare array of colorspace strings on its own line.
fate: amix: specify f32 sample format for comparison
tiny_psnr: support 32-bit float samples
eamad/eatgq/eatqi: call special EA IDCT directly
eamad: remove use of MpegEncContext
mpegvideo: remove unnecessary inclusions of faandct.h
af_asyncts: avoid overflow in out_size with large delta values
af_asyncts: add first_pts option
Conflicts:
configure
libavcodec/eamad.c
libavcodec/h264_ps.c
libavformat/crcenc.c
libavformat/ffmdec.c
libavformat/ffmenc.c
libavformat/framecrcenc.c
libavformat/md5enc.c
libavformat/nutdec.c
libavformat/rawenc.c
libavformat/yuv4mpeg.c
tests/tiny_psnr.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
x86: swscale: Place inline assembly code under appropriate #ifdefs
rtsp: remove terminal comma in FF_RTP_FLAG_OPTS macro.
configure: Remove redundant RTMPT/RTMPTS dependencies
configure: add filtering of host cflags/ldflags
configure: initialise all flag filters at the same place
configure: add filtering of linker flags
configure: name some variables more consistently
configure: remove filter_cppflags
configure: set icc_version where it is needed
mpegenc: remove disabled code
Conflicts:
configure
libavformat/movenc.c
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* qatar/master:
MS Screen 1 decoder
aacdec: Fix popping channel layouts.
av_gettime: support Win32 without gettimeofday()
Use av_gettime() in various places
Move av_gettime() to libavutil
dct-test: use emms_c() from libavutil instead of duplicating it
mov: fix operator precedence bug
mathematics.h: remove a couple of math defines
Remove unnecessary inclusions of [sys/]time.h
lavf: remove unnecessary inclusions of unistd.h
bfin: libswscale: add const where appropriate to fix warnings
bfin: libswscale: remove unnecessary #includes
udp: Properly check for invalid sockets
tcp: Check the return value from getsockopt
network: Use av_strerror for getting error messages
udp: Properly print error from getnameinfo
mmst: Use AVUNERROR() to convert error codes to the right range for strerror
network: Pass pointers of the right type to get/setsockopt/ioctlsocket on windows
rtmp: Reduce the number of idle posts sent by sleeping 50ms
Conflicts:
Changelog
configure
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dct-test.c
libavcodec/version.h
libavformat/riff.c
libavformat/udp.c
libavutil/Makefile
libswscale/bfin/yuv2rgb_bfin.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (24 commits)
flvdec: remove incomplete, disabled seeking code
mem: add support for _aligned_malloc() as found on Windows
lavc: Extend the documentation for avcodec_init_packet
flvdec: remove incomplete, disabled seeking code
http: replace atoll() with strtoll()
mpegts: remove unused/incomplete/broken seeking code
af_amix: allow float planar sample format as input
af_amix: use AVFloatDSPContext.vector_fmac_scalar()
float_dsp: add x86-optimized functions for vector_fmac_scalar()
float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
lavr: Add x86-optimized function for flt to s32 conversion
lavr: Add x86-optimized function for flt to s16 conversion
lavr: Add x86-optimized functions for s32 to flt conversion
lavr: Add x86-optimized functions for s32 to s16 conversion
lavr: Add x86-optimized functions for s16 to flt conversion
lavr: Add x86-optimized function for s16 to s32 conversion
rtpenc: Support packetizing iLBC
rtpdec: Add a depacketizer for iLBC
Implement the iLBC storage file format
mov: Support muxing/demuxing iLBC
...
Conflicts:
Changelog
configure
libavcodec/avcodec.h
libavcodec/dsputil.c
libavcodec/version.h
libavformat/movenc.c
libavformat/mpegts.c
libavformat/version.h
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This seems to be the correct mode to send, according to the
original RTSP RFC, and matches the method RECORD which is
sent later when starting to send data.
Darwin Streaming Server works fine with either of them.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avprobe: restore pseudo-INI old style format for compatibility.
avprobe: fix formatting.
log: make colored output more colorful.
rtsp: Check for dynamic payload handlers if no static payload mapping was found
Conflicts:
Changelog
doc/ffprobe.texi
ffprobe.c
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
opt: Add av_opt_set_bin()
avconv: Display the error returned by avformat_write_header
rtpenc_chain: Return an error code instead of just a plain pointer
rtpenc_chain: Free the URLContext on failure
rtpenc: Expose the ssrc as an avoption
avprobe: display the codec profile in show_stream()
avprobe: fix function prototype
cosmetics: Fix indentation
avprobe: changelog entry
avprobe: update documentation
avprobe: provide JSON output
avprobe: output proper INI format
avprobe: improve formatting
rtmp: fix url parsing
fate: document TARGET_EXEC and its usage
Conflicts:
doc/APIchanges
doc/fate.texi
doc/ffprobe.texi
ffprobe.c
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some systems abuse the static payload types 35 or 36 (which
according to IANA are unassigned) for H264.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
* qatar/master: (28 commits)
dfa: use more meaningful return codes
eatgv: check vector_bits
eatgv: check motion vectors
Mark a number of variables only used in av_dlog() calls as av_unused.
dvdec: drop const qualifier from variable to eliminate a warning
avcodec: Improve comment for thread_safe_callbacks to avoid misinterpretation.
tests/utils: don't ignore the return value of fwrite()
lavfi/formats: use sizeof(var) instead of sizeof(type).
lavfi: remove avfilter_default_config_input_link() declaration
lavfi: always enable the scale filter and depend on sws.
vf_split: support user-specifiable number of outputs.
avconv: remove stray useless comment.
mpegmux: add stuffing to avoid incomplete PCM frames
rtsp: avoid const warnings from strtol() call
avserver: check return value of ftruncate()
lagarith: make offset array type unsigned
dfa: add some checks to ensure that decoder won't write past frame end
aacps: NEON optimisations
aacps: align some arrays
aacps: move some loops to function pointers
...
Conflicts:
configure
doc/filters.texi
libavcodec/dfa.c
libavcodec/eatgv.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/vf_split.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
rtsp: Don't use av_malloc(0) if there are no streams
rtsp: Don't use uninitialized data if there are no streams
vaapi: mpeg2: fix slice_vertical_position calculation.
hwaccel: mpeg2: decode first field, if requested.
cosmetics: Fix indentation
rtsp: Don't expose the MS-RTSP RTX data stream to the caller
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
asf: only set index_read if the index contained entries.
cabac: add overread protection to BRANCHLESS_GET_CABAC().
cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC().
cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE().
cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC().
h264: add overread protection to get_cabac_bypass_sign_x86().
h264: reindent get_cabac_bypass_sign_x86().
h264: use struct offsets in get_cabac_bypass_sign_x86().
h264: fix overreads in cabac reader.
wmall: fix seeking.
lagarith: fix buffer overreads.
dvdec: drop unnecessary dv_tablegen.h #include
build: fix doc generation errors in parallel builds
Replace memset(0) by zero initializations.
faandct: Remove FAAN_POSTSCALE define and related code.
dvenc: print allowed profiles if the video doesn't conform to any of them.
avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size.
FATE: add a test for vp8 with changing frame size.
fate: add kgv1 fate test.
oggdec: calculate correct timestamps in Ogg/FLAC
Conflicts:
libavcodec/4xm.c
libavcodec/cook.c
libavcodec/dvdata.c
libavcodec/dvdsubdec.c
libavcodec/lagarith.c
libavcodec/lagarithrac.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc/general: update supported devices table.
doc/general: add missing @tab to codecs table.
h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
avconv: reindent
avconv: link '-passlogfile' option to libx264 'stats' AVOption.
libx264: add 'stats' private option for setting 2pass stats filename.
libx264: fix help text for slice-max-size option.
http: Clear the auth state on redirects
http: Retry auth if it failed due to being stale
rtsp: Resend new keepalive commands if they used stale auth
rtsp: Retry authentication if failed due to being stale
httpauth: Parse the stale field in digest auth
dxva2_vc1: pass the overlap flag to the decoder
dxva2_vc1: fix decoding of BI frames
FATE: add shorthand to wavpack test
dfa: convert to bytestream2 API
anm decoder: move buffer allocation from decode_init() to decode_frame()
h264: improve parsing of broken AVC SPS
Conflicts:
ffmpeg.c
libavcodec/anm.c
libavcodec/dfa.c
libavcodec/h264.c
libavcodec/h264_direct.c
libavcodec/h264_ps.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
pcm-mpeg: convert to bytestream2 API
Revert "h264: clear trailing bits in partially parsed NAL units"
remove iwmmxt optimizations
mimic: do not continue if swap_buf_size is 0
mimic: convert to bytestream2 API
frwu: use MKTAG to check marker instead of AV_RL32
txd: port to bytestream2 API
c93: convert to bytestream2 API
iff: make .long_name more descriptive
FATE: add test for cdxl demuxer
rtsp: Fix a typo
Conflicts:
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/dsputil_iwmmxt_rnd_template.c
libavcodec/arm/mpegvideo_iwmmxt.c
libavcodec/c93.c
libavcodec/txd.c
libavutil/arm/cpu.c
tests/fate/demux.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix a bunch of common typos.
build: Skip compiling xvmc.h under the correct condition.
configure: darwin: Change dylib install names to include major version.
mpegts: Always honor a registration descriptor if present and there is no other codec information.
aacdec: Fix SCE parity check.
aacdec: Fix out of array writes (stack).
rtsp: Only set the ttl parameter if the server actually gave a value
udp: Set ttl for read-write streams, too, not only for write-only ones
udp: Only bind to the multicast address if in read-only mode
udp: Clarify the comment about binding the multicast address
udp: Reorder comments
Conflicts:
libavcodec/aacdec.c
tools/patcheck
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avcodec_default_reget_buffer(): fix compilation in DEBUG mode
fate: Overhaul WavPack coverage
h264: fix mmxext chroma deblock to use correct TC values.
flvdec: Remove the now redundant check for known broken metadata creator
flvdec: Validate index entries added from metadata while reading
rtsp: Handle requests from server to client
movenc: use timestamps instead of frame_size for samples-per-packet
movenc: use the first cluster duration as the tfhd default duration
movenc: factorize calculation of cluster duration into a separate function
doc/APIchanges: fill in missing dates and hashes.
lavc: reorder AVCodecContext fields.
lavc: reorder AVFrame fields.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavformat/flvdec.c
libavformat/movenc.c
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.
Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.
Signed-off-by: Martin Storsjö <martin@martin.st>
