* release/0.7: (290 commits)
nuv: Fix combination of size changes and LZO compression.
av_lzo1x_decode: properly handle negative buffer length.
Do not call parse_keyframes_index with NULL stream.
update versions for 0.7 branch
Version numbers for 0.8.6
snow: emu edge support Fixes Ticket592
imc: validate channel count
imc: check for ff_fft_init() failure (cherry picked from commit 95fee70d67)
libgsmdec: check output buffer size before decoding (cherry picked from commit b03761b130)
configure: fix arch x86_32
mp3enc: avoid truncating id3v1 tags by one byte
asfdec: Check packet_replic_size earlier
cin audio: validate the channel count
binkaudio: add some buffer overread checks.
atrac1: validate number of channels (cherry picked from commit bff5b2c1ca)
atrac1: check output buffer size before decoding (cherry picked from commit 33684b9c12)
vp3: fix oob read for negative tokens and memleaks on error. (cherry picked from commit 8370e426e4)
apedec: set s->currentframeblocks after validating nblocks
apedec: use unsigned int for 'nblocks' and make sure that it's within int range
apedec: check for data buffer realloc failure (cherry picked from commit 11ca8b2d74)
...
Conflicts:
Changelog
Makefile
RELEASE
configure
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavformat/matroskaenc.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Writing zeros to the high entries in the array need only be
done once as the cutoff position is constant throughout the
loop.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit bf00a73ace)
* khirnov/release/0.7: (64 commits)
rv34: Check for invalid slice offsets
rv34: Fix potential overreads
rv34: Avoid NULL dereference on corrupted bitstream
rv10: Reject slices that does not have the same type as the first one
lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
oggdec: fix out of bound write in the ogg demuxer
Fixed size given to init_get_bits().
smacker: fix a few off by 1 errors
Check for invalid VLC value in smacker decoder.
Check and propagate errors when VLC trees cannot be built in smacker decoder.
Fixed off by one packet size allocation in the smacker demuxer.
Check for invalid packet size in the smacker demuxer.
ape demuxer: fix segfault on memory allocation failure.
xan: Add some buffer checks (cherry picked from commit 0872bb23b4)
Fixed size given to init_get_bits() in xan decoder. (cherry picked from commit 393d5031c6)
smacker demuxer: handle possible av_realloc() failure.
Fixed segfault with wavpack decoder on corrupted decorrelation terms sub-blocks.
cljr: init_get_bits size in bits instead of bytes (cherry picked from commit 0c1f5b93d9)
indeo2: fail if input buffer too small (cherry picked from commit b7ce4f1d1c)
indeo2: init_get_bits size in bits instead of bytes (cherry picked from commit 68ca330cbd)
...
Conflicts:
ffmpeg.c
libavdevice/alsa-audio.h
libavformat/gxf.c
libswscale/x86/swscale_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Set the frame size when decoding DTS audio.
This has the side effect of fixing the computation of timestamps for DTS-HD in compute_pkt_fields. Since frame_size is
not currently set, the duration of a frame is being guessed based on the streams bitrate. But for DTS-HD, the bitrate
currently used is the rate of the DTS core which is much different than the whole DTS-HD stream and leads to a wildly
inaccurate frame duration estimate.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 49c7006c7e)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master: (36 commits)
ARM: allow unaligned buffer in fixed-point NEON FFT4
fate: test more FFT etc sizes
dca: set AVCodecContext frame_size for DTS audio
YASM: Shut up unused variable compiler warning with --disable-yasm.
x86_32: Fix build on x86_32 with --disable-yasm.
iirfilter: add fate test
doxygen: Add qmul docs.
ogg: propagate return values and return more meaningful error values
H.264: fix overreads of qscale_table
Remove unused static tables and static inline functions.
eval: clear Parser instances before using
dct-test: remove 'ref' function pointer from tables
build: Remove deleted 'check' target from .PHONY list.
oggdec: Abort Ogg header parsing when encountering a data packet.
Add LGPL license boilerplate to files lacking it.
mxfenc: small typo fix
doxygen: Fix documentation for some VP8 functions.
sha: use AV_RB32() instead of assuming buffer can be cast to uint32_t*
des: allow unaligned input and output buffers
aes: allow unaligned input and output buffers
...
Conflicts:
libavcodec/dct-test.c
libavcodec/libvpxenc.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/h264_qpel_mmx.c
libavfilter/x86/gradfun.c
libavformat/oggdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Set the frame size when decoding DTS audio.
This has the side effect of fixing the computation of timestamps for DTS-HD in compute_pkt_fields. Since frame_size is
not currently set, the duration of a frame is being guessed based on the streams bitrate. But for DTS-HD, the bitrate
currently used is the rate of the DTS core which is much different than the whole DTS-HD stream and leads to a wildly
inaccurate frame duration estimate.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
cosmetics: fix some then/than typos
doxygen: Include libavcodec and libavformat examples into the documentation
avutil: elaborate documentation for av_get_random_seed
Add support for aac streams in mp4/mov without extradata.
aes: whitespace cosmetics
adler32: whitespace cosmetics
swscale: fix another yuv range conversion overflow in 16bit scaling.
Fix cpu flags test program
opt-test: Add missing braces to silence compiler warnings.
build: Eliminate obsolete test targets.
udp: Fix a compilation warning
swscale: Unbreak build with --enable-small
base64: add fate test
aes: improve test program and add fate test
adler32: make test program more useful and add fate test
swscale: fix yuv range correction when using 16-bit scaling.
aacenc: Make chan_map const correct
Conflicts:
Makefile
doc/examples/muxing-example.c
libavformat/udp.c
libavutil/random_seed.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
APIchanges: fill in date and commit for request_sample_fmt
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
Add support for request_sample_format in ffmpeg and ffplay.
Add APIchanges entry for request_sample_fmt.
Add request_sample_fmt field to AVCodecContext.
Add float_interleave() to FmtConvertContext with x86-optimized versions.
Remove unused make variable SEEK_REFFILE
fate: remove redundant aref and vref references
fate: remove do_ffmpeg_nocheck function
fate: do not collect -benchmark output
mpegaudiodec: remove decode_end() function
fate: run aref and vref as regular tests
mpegaudio: sanitise compute_antialias_* names
mpeg12: add slice-threading checks to slice-threading initializers.
h264: copy pixel_shift between slice threading contexts.
mdec: enable frame-level multithreading.
mdec.c: fix overread.
Conflicts:
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/dca.c
libavcodec/h264.c
libavcodec/mdec.c
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/version.h
libavcodec/vorbisdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
They use now code identical to the AAC decoder.
The AC3 decoder previously did not check the data_size and
the dca decoder checked against and set wrong values for float.
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 7e06e0ede3)
This avoids the core substream extensions scan when the EXT_AUDIO_ID
field indicates no extensions or only unsupported extensions. The scan
is done only if the value of EXT_AUDIO_ID is unknown or indicates a
present XCh extension which we can decode.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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>
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672)
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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)
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 8f4a5d225c)
DTS-HD HRA streams do not always have an XBR extension in the extension
substream. Instead they can have only XXCh and X96 extensions in
there and still be considered DTS-HD HRA.
This is also confirmed with Onkyo TX-SR607 receiver which recognizes
such a stream as HiRes Audio.
Signed-off-by: Mans Rullgard <mans@mansr.com>
