* 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
