* 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>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: make compare() function compatible with POSIX bc
Update Janne's email address.
APIchanges: Replace Subversion revision numbers by Git hashes.
bytestream: Eliminate one level of pointless macro indirection.
xwd: convert to bytestream2.
vqavideo: port to bytestream2 API
Read preset files with suffix .avpreset
prores: allow user to set fixed quantiser
lavf: remove some disabled code.
lavf: only set average frame rate for video.
lavf: remove a pointless check.
avcodec: add XBM encoder
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vqavideo.c
libavformat/img2enc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also add bbox.h and bbox.c files, based on the remove-logo filter by
Robert Edele. These files are useful for sharing code with the pending
removelogo port.
All colorspaces are supported.
Renamed libutvideo.cpp to libutvideodec.cpp.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (29 commits)
amrwb: remove duplicate arguments from extrapolate_isf().
amrwb: error out early if mode is invalid.
h264: change underread for 10bit QPEL to overread.
matroska: check buffer size for RM-style byte reordering.
vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
vp8: change int stride to ptrdiff_t stride.
wma: fix invalid buffer size assumptions causing random overreads.
Windows Media Audio Lossless decoder
rv10/20: Fix slice overflow with checked bitstream reader.
h263dec: Disallow width/height changing with frame threads.
rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
rmdec: Honor .RMF tag size rather than assuming 18.
g722: Fix the QMF scaling
r3d: don't set codec timebase.
electronicarts: set timebase for tgv video.
electronicarts: parse the framerate for cmv video.
ogg: don't set codec timebase
electronicarts: don't set codec timebase
avs: don't set codec timebase
wavpack: Fix an integer overflow
...
Conflicts:
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/fraps.c
libavcodec/h264.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/msmpeg4.c
libavcodec/pnmdec.c
libavcodec/qpeg.c
libavcodec/rawenc.c
libavcodec/ulti.c
libavcodec/vcr1.c
libavcodec/version.h
libavcodec/wmalosslessdec.c
libavformat/electronicarts.c
libswscale/ppc/yuv2rgb_altivec.c
tests/ref/acodec/g722
tests/ref/fate/ea-cmv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Decodes 16-bit WMA Lossless encoded files. 24-bit is not supported yet.
Bitstream parser written by Andreas Öman with contributions from
Baptiste Coudurier and Ulion.
Includes a number of bug-fixes from Benjamin Larsson, Michael Niedermayer and
Konstantin Shishkov, shine and polish by Diego Biurrun.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This library does not fit into Libav as a whole and its code is just a
maintenance burden. Furthermore it is now available as an external project,
which completely obviates any reason to keep it around.
URL: http://git.videolan.org/?p=libpostproc.git
* qatar/master: (22 commits)
als: prevent infinite loop in zero_remaining().
cook: prevent div-by-zero if channels is zero.
pamenc: switch to encode2().
svq1enc: switch to encode2().
dvenc: switch to encode2().
dpxenc: switch to encode2().
pngenc: switch to encode2().
v210enc: switch to encode2().
xwdenc: switch to encode2().
ttadec: use branchless unsigned-to-signed unfolding
avcodec: add a Sun Rasterfile encoder
sunrast: Move common defines to a new header file.
cdxl: fix video decoding for some files
cdxl: fix audio for some samples
apetag: add proper support for binary tags
ttadec: remove dead code
swscale: make access to filter data conditional on filter type.
swscale: update context offsets after removal of AlpMmxFilter.
prores: initialise encoder and decoder parts only when needed
swscale: make monowhite/black RGB-independent.
...
Conflicts:
Changelog
libavcodec/alsdec.c
libavcodec/dpxenc.c
libavcodec/golomb.h
libavcodec/pamenc.c
libavcodec/pngenc.c
libavformat/img2.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
CDXL demuxer and decoder
hls: Re-add legacy applehttp name to preserve interface compatibility.
hlsproto: Rename the functions and context
hlsproto: Encourage users to try the hls demuxer instead of the proto
doc: Move the hls protocol section into the right place
libavformat: Rename the applehttp protocol to hls
hls: Rename the functions and context
libavformat: Rename the applehttp demuxer to hls
rtpdec: Support H263 in RFC 2190 format
rv30: check block type validity
ttadec: CRC checking
movenc: Support muxing VC1
avconv: Don't split out inline sequence headers when stream copying VC1
rv34: handle size changes during frame multithreading
rv40: prevent undefined signed overflow in rv40_loop_filter()
rv34: use AVERROR return values in ff_rv34_decode_frame()
rv34: use uint16_t for RV34DecContext.deblock_coefs
librtmp: Add "lib" prefix to librtmp URLProtocol declarations.
movenc: Use defines instead of hardcoded numbers for RTCP types
smjpegdec: implement seeking
...
Conflicts:
Changelog
doc/general.texi
libavcodec/avcodec.h
libavcodec/rv30.c
libavcodec/tta.c
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
wma: Clip WMA1 and WMA2 frame length to 11 bits.
movenc: Don't require frame_size to be set for modes other than mov
doc: Update APIchanges with info on muxer flushing
movenc: Reindent a block
tools: Remove some unnecessary #undefs.
rv20: prevent calling ff_h263_decode_mba() with unset height/width
tools: K&R reformatting cosmetics
Ignore generated aviocat and ismindex tools.
build: Automatically include architecture-specific library Makefile snippets.
indeo5: prevent null pointer dereference on broken files
pktdumper: Use usleep instead of sleep
cosmetics: Remove some unnecessary block braces.
Drop unnecessary prefix from *sink* variable and struct names.
Add a tool for creating smooth streaming manifests
movdec: Calculate an average bit rate for fragmented streams, too
movenc: Write the sample rate instead of time scale in the stsd atom
movenc: Add a separate ismv/isma (smooth streaming) muxer
movenc: Allow the caller to decide on fragmentation
libavformat: Add a flag for muxers that support write_packet(NULL) for flushing
movenc: Add support for writing fragmented mov files
...
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
ffmpeg.c
ffplay.c
libavfilter/Makefile
libavformat/Makefile
libavformat/avformat.h
libavformat/movenc.c
libavformat/movenc.h
libavformat/version.h
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
aacenc: Fix identification padding when the bitstream is already aligned.
aacenc: Write correct length for long identification strings.
aud: remove unneeded field, audio_stream_index from context
aud: fix time stamp calculation for ADPCM IMA WS
aud: simplify header parsing
aud: set pts_wrap_bits to 64.
cosmetics: indentation
aud: support Westwood SND1 audio in AUD files.
adpcm_ima_ws: fix stereo decoding
avcodec: add a new codec_id for CRYO APC IMA ADPCM.
vqa: remove unused context fields, audio_samplerate and audio_bits
vqa: clean up audio header parsing
vqa: set time base to frame rate as coded in the header.
vqa: set packet duration.
vqa: use 1/sample_rate as the audio stream time base
vqa: set stream start_time to 0.
lavc: postpone the removal of AVCodecContext.request_channels.
lavf: postpone removing av_close_input_file().
lavc: postpone removing old audio encoding and decoding API
avplay: remove the -er option.
...
Conflicts:
Changelog
libavcodec/version.h
libavdevice/v4l.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Remove ffmpeg.
aacenc: Simplify windowing
aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
aacenc: Deinterleave input samples before processing.
aacenc: Store channel count in AACEncContext.
aacenc: Move Q^3/4 calculation to it's own table
aacenc: Request normalized float samples instead of converting s16 samples to float.
aacpsy: Replace an if with FFMAX in LAME windowing.
aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
aacenc: cosmetics: move init() and end() to the bottom of the file.
aacenc: aac_encode_init() cleanup
XWD encoder and decoder
vc1: don't read the interpfrm and bfraction elements for interlaced frames
mxfdec: fix memleak on mxf_read_close()
westwood: split the AUD and VQA demuxers into separate files.
Conflicts:
.gitignore
Changelog
Makefile
configure
doc/ffmpeg.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/aacenc.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/img2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous implementation assumed that a new picture would always
supersede the previous picture. Similarly, presentation segments
were assumed to pertain to the most-recently-read picture.
However, each presentation segment may refer to 0 or more pictures
by their ID. Picture IDs may repeat, and a repeated picture ID
indicates that the old picture for that ID is no longer needed
and may be discarded.
The new implementation allocates a buffer with one slot for each
possible picture ID (the picture ID is a 16-bit field) and
properly decodes presentation segments so that all relevant
pictures are output upon encountering a display segment.
Given that most PGS streams are unlikely to use more than a small
fraction of the available picture IDs, it would probably be better
to use a more memory-efficient data structure. I'm lazy though, so
I leave this to a more motivated individual.
I've tested the code with MKV files in VLC (a recent revision from
their git repo) and with HandBrake (a version that I hacked up to
use ffmpeg's PGS subtitle decoder).
Review-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rv34: add NEON rv34_idct_add
rv34: 1-pass inter MB reconstruction
add SMJPEG muxer
avformat: split out common SMJPEG code
pictordec: Use bytestream2 functions
avconv: use avcodec_encode_audio2()
pcmenc: use AVCodec.encode2()
avcodec: bump minor version and add APIChanges for the new audio encoding API
avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio()
avcodec: add a public function, avcodec_fill_audio_frame().
rv34: Intra 16x16 handling
rv34: Inter/intra MB code split
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/pictordec.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/rv34dsp.asm
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fft: init functions with INIT_XMM/YMM.
pcmenc: set frame_size to 0.
gsm demuxer: use generic seeking instead of a gsm-specific function.
gsm demuxer: return packets with only 1 gsm block at a time.
avcodec: add GSM parser
doc: Replace ffmpeg references in avserver config file by avconv.
doc: Fix names of av_log color environment variables.
Fix a bunch of platform name and other typos.
Add some missing changelog entries and release 0.8_beta2
No longer build libpostproc by default
wtv: fix memleaks during normal operation
threads: add CODEC_CAP_AUTO_THREADS for libvpx and xavs
Conflicts:
Changelog
RELEASE
cmdutils.c
configure
doc/ffserver.conf
doc/platform.texi
ffplay.c
libavcodec/Makefile
libavcodec/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The WAVE demuxer returns packets with many blocks per frame, which needs to be
parsed into single blocks. This has a side-effect of fixing the timestamps.
