Instead of this it would be possible to change the code to support
larger K but that would make the code slower and probably still need
checks on K.
This variant is choosen as there are no known valid files that need
larger K
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '205a95f7b5178362874bc1e65eae9866723491c1':
wmaenc: alloc/free coded_frame instead of keeping it in the WMACodecContext
wma: decode directly to the user-provided AVFrame
wmapro: decode directly to the user-provided AVFrame
wavpack: decode directly to the user-provided AVFrame
Conflicts:
libavcodec/wavpack.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Wavpack format documentation mentions that 131072 is
max number of samples.
This fixes huge memory allocations in sample from ticket #1889.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Integer Overflow Checker detected an integer
overflow while FATE was running.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (40 commits)
swf: check return values for av_get/new_packet().
wavpack: Don't shift minclip/maxclip
rtpenc: Expose the max packet size via an avoption
rtpenc: Move max_packet_size to a context variable
rtpenc: Add an option for not sending RTCP packets
lavc: drop encode() support for video.
snowenc: switch to encode2().
snowenc: don't abuse input picture for storing information.
a64multienc: switch to encode2().
a64multienc: don't write into output buffer when there's no output.
libxvid: switch to encode2().
tiffenc: switch to encode2().
tiffenc: properly forward error codes in encode_frame().
lavc: drop libdirac encoder.
gifenc: switch to encode2().
libvpxenc: switch to encode2().
flashsvenc: switch to encode2().
Remove libpostproc.
lcl: don't overwrite input memory.
swscale: take first/lastline over/underflows into account for MMX.
...
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/APIchanges
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdiracenc.c
libavcodec/libxvidff.c
libavcodec/qtrleenc.c
libavcodec/tiffenc.c
libavcodec/utils.c
libavformat/mov.c
libavformat/movenc.c
libpostproc/Makefile
libpostproc/postprocess.c
libpostproc/postprocess.h
libpostproc/postprocess_altivec_template.c
libpostproc/postprocess_internal.h
libpostproc/postprocess_template.c
libswscale/swscale.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Since we are clipping before we shift the values to
16 or 32 bits, we should not shift the min/max clip
values to compensate.
Fixes 8 and 24 bit lossy decoding.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (38 commits)
v210enc: remove redundant check for pix_fmt
wavpack: allow user to disable CRC checking
v210enc: Use Bytestream2 functions
cafdec: Check return value of avio_seek and avoid modifying state if it fails
yop: Check return value of avio_seek and avoid modifying state if it fails
tta: Check return value of avio_seek and avoid modifying state if it fails
tmv: Check return value of avio_seek and avoid modifying state if it fails
r3d: Check return value of avio_seek and avoid modifying state if it fails
nsvdec: Check return value of avio_seek and avoid modifying state if it fails
mpc8: Check return value of avio_seek and avoid modifying state if it fails
jvdec: Check return value of avio_seek and avoid modifying state if it fails
filmstripdec: Check return value of avio_seek and avoid modifying state if it fails
ffmdec: Check return value of avio_seek and avoid modifying state if it fails
dv: Check return value of avio_seek and avoid modifying state if it fails
bink: Check return value of avio_seek and avoid modifying state if it fails
Check AVCodec.pix_fmts in avcodec_open2()
svq3: Prevent illegal reads while parsing extradata.
remove ParseContext1
vc1: use ff_parse_close
mpegvideo parser: move specific fields into private context
...
Conflicts:
libavcodec/4xm.c
libavcodec/aacdec.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/mpeg4video_parser.c
libavcodec/svq3.c
libavcodec/v210enc.c
libavformat/cafdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts e6e7bfc1 and 365e1ec2.
The code may be incorrect both before and after the revert, but we
do not have any samples that were fixed by the original commits.
Fixes ticket #871.
* qatar/master:
fate: split ADPCM and DPCM test references into separate files.
mov, mxfdec: Employ more meaningful return values.
lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer()
wavpack: fix clipping for 32-bit lossy mode
vb: Use bytestream2 functions
Conflicts:
libavcodec/utils.c
libavcodec/vb.c
libavformat/mxfdec.c
tests/fate/dpcm.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Reference decoder clips data before shifting it to final range and also
forces 32-bit lossy mode to be actually 24-bit lossy mode in order to be
able to perform proper clipping.
* qatar/master:
mpegenc: use avctx->slices as number of slices
v410enc: fix undefined signed left shift caused by integer promotion
Release notes: mention cleaned up header includes
fix Changelog file
Fix a bunch of typos.
Drop some pointless void* return value casts from av_malloc() invocations.
wavpack: fix typos in previous cosmetic clean-up commit
wavpack: cosmetics: K&R pretty-printing
avconv: remove the 'codec framerate is different from stream' warning
wavpack: determine sample_fmt before requesting a buffer
bmv audio: implement new audio decoding API
mpegaudiodec: skip all channels when skipping granules
mpegenc: simplify muxrate calculation
Conflicts:
Changelog
avconv.c
doc/RELEASE_NOTES
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpegaudiodec.c
libavcodec/mpegvideo.c
libavformat/mpegenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
build: fix standalone compilation of OMA muxer
build: fix standalone compilation of Microsoft XMV demuxer
build: fix standalone compilation of Core Audio Format demuxer
kvmc: fix invalid reads
4xm: Add a check in decode_i_frame to prevent buffer overreads
adpcm: fix IMA SMJPEG decoding
options: set minimum for "threads" to zero
bsd: use number of logical CPUs as automatic thread count
windows: use number of CPUs as automatic thread count
linux: use number of CPUs as automatic thread count
pthreads: reset active_thread_type when slice thread_init returrns early
v410dec: include correct headers
Drop ALT_ prefix from BITSTREAM_READER_LE name.
lavfi: always build vsrc_buffer.
ra144enc: zero the reflection coeffs if the filter is unstable
sws: readd PAL8 to isPacked()
mov: Don't stick the QuickTime field ordering atom in extradata.
truespeech: fix invalid reads in truespeech_apply_twopoint_filter()
Conflicts:
configure
libavcodec/4xm.c
libavcodec/avcodec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavformat/Makefile
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
get_bits: remove A32 variant
avconv: support stream specifiers in -metadata and -map_metadata
wavpack: Fix 32-bit clipping
wavpack: Clip samples after shifting
h264: don't drop B-frames after next keyframe on POC reset.
get_bits: remove useless pointer casts
configure: refactor lists of tests and components into variables
rv40: NEON optimised weak loop filter
mpegts: replace some magic numbers with the existing define
swscale: add unscaled packed 16 bit per component endianess conversion
Conflicts:
libavcodec/get_bits.h
libavcodec/h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In the case that (frame_flags & 0x03) == 3, hybrid_maxclip
may have had a signed integer overflow.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It doesn't make much sense to clip pre-shift,
nor is it correct for proper decoding.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
When decoding lossy WavPack samples, they are supposed
to be clipped, in order to be decoded correctly.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Move id3v2 tag writing to a separate file.
swscale: add missing colons to x86 assembly yuv2planeX.
g722: split decoder and encoder into separate files
cosmetics: remove extra spaces before end-of-statement semi-colons
vorbisdec: check output buffer size before writing output
wavpack: calculate bpp using av_get_bytes_per_sample()
ac3enc: Set max value for mode options correctly
lavc: move get_b_cbp() from h263.h to mpeg4videoenc.c
mpeg12: move closed_gop from MpegEncContext to Mpeg1Context
mpeg12: move full_pel from MpegEncContext to Mpeg1Context
mpeg12: move Mpeg1Context from mpeg12.c to mpeg12.h
mpegvideo: remove some unused variables from MpegEncContext.
Conflicts:
libavcodec/mpeg12.c
libavformat/mp3enc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
wavpack_decode_block() supposes that it is called back with the exact
same buffer unless it has returned with an error. With multi-channels
files, wavpack_decode_frame() was breaking this assumption.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
wavpack_decode_block() supposes that it is called back with the exact
same buffer unless it has returned with an error. With multi-channels
files, wavpack_decode_frame() was breaking this assumption.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
avconv: move audio_channels to the options context.
avconv: move *_disable to options context.
avconv: remove -[vas]lang options.
avconv: move codec tags to options context.
cljr: init_get_bits size in bits instead of bytes
indeo2: fail if input buffer too small
indeo2: init_get_bits size in bits instead of bytes
ffv1: Fixed size given to init_get_bits() in decoder.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Employ FF_ARRAY_ELEMS instead of manually calculating array length.
Fixed invalid access in wavpack decoder on corrupted bitstream.
Fixed invalid writes in wavpack decoder on corrupted bitstreams.
Fixed invalid access in wavpack decoder on corrupted extra bits sub-blocks.
rtpdec_asf: Fix integer underflow that could allow remote code execution
Conflicts:
libavformat/rtpdec_asf.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This function should return number of samples decoded, not number of bytes
decoded.
Spotted by Uoti Urpala.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This fixes improper flushing in the cases when the same frame is decoded in
several iterations (for being too large to fit into output buffer) and flush is
called mid-decoding and it also resets context in case of decoding errors.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
WavPack demuxer: do not rely on index when timestamp is not in indexed range.
WavPack demuxer: store position of the first block in index.
WavPack decoder: implement flush function
avconv: Separate initialization from the main transcode loop.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mxfdec: Include FF_INPUT_BUFFER_PADDING_SIZE when allocating extradata.
H.264: tweak some other x86 asm for Atom
probe: Fix insane flow control.
mpegts: remove invalid error check
s302m: use nondeprecated audio sample format API
lavc: use designated initialisers for all codecs.
x86: cabac: add operand size suffixes missing from 6c32576
Conflicts:
libavcodec/ac3enc_float.c
libavcodec/flacenc.c
libavcodec/frwu.c
libavcodec/pictordec.c
libavcodec/qtrleenc.c
libavcodec/v210enc.c
libavcodec/wmv2dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is pretty hopeless that other considerable projects will adopt
libavutil alone in other projects. Projects that need small footprint
are better off with more specialized libraries such as gnulib or rather
just copy the necessary parts that they need. With this in mind, nobody
is helped by having libavutil and libavcore split. In order to ease
maintenance inside and around FFmpeg and to reduce confusion where to
put common code, avcore's functionality is merged (back) to avutil.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
None of these symbols should be accessed directly, so declare them as
hidden.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit d36beb3f69)
Passing an explicit filename to this command is only necessary if the
documentation in the @file block refers to a file different from the
one the block resides in.
Originally committed as revision 22921 to svn://svn.ffmpeg.org/ffmpeg/trunk
hold, decode it in several iterations outputting as many samples as possible.
Originally committed as revision 21894 to svn://svn.ffmpeg.org/ffmpeg/trunk
per AVPacket
No longer print "Multiple frames in a packet" error message
when CODEC_CAP_SUBFRAMES is set (wmapro, wavpack)
Originally committed as revision 19881 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Laurent Aimar (fenrir at `antonym of 'audio'+antonym of 'WAN'` dot org)
Originally committed as revision 18754 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch by Laurent Aimar <fenrir at VLCsite>
Thread: [PATCH] Fixed wavpack 9-15 bits support
Originally committed as revision 18666 to svn://svn.ffmpeg.org/ffmpeg/trunk
