* qatar/master:
adtsenc: Check frame size.
txd: Fix order of operations.
APIchanges: fill in some blanks
timer: fix misspelling of "decicycles"
Eliminate pointless 0/NULL initializers in AVCodec and similar declarations.
indeo3: cosmetics
md5proto: Fix order of operations.
dca: Replace oversized unused get_bits() with skip_bits_long().
Conflicts:
doc/APIchanges
libavformat/mmsh.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It makes more sense for a bit mask to use an unsigned type.
The change should be source and binary compatible on all
supported systems, hence micro version bump.
Fixes a few invalid shifts.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is a hand-tuned version of the code with impossible parts of
the FASTDIV function ommitted.
2-5% faster overall on Cortex-A8.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (51 commits)
cin audio: use sign_extend() instead of casting to int16_t
cin audio: restructure decoding loop to avoid a separate counter variable
cin audio: use local variable for delta value
cin audio: remove unneeded cast from void*
cin audio: validate the channel count
cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
dsicin: fix several audio-related fields in the CIN demuxer
flacdec: use av_get_bytes_per_sample() to get sample size
dca: handle errors from dca_decode_block()
dca: return error if the frame header is invalid
dca: return proper error codes instead of -1
utvideo: handle empty Huffman trees
binkaudio: change short to int16_t
binkaudio: only decode one block at a time.
binkaudio: store interleaved overlap samples in BinkAudioContext.
binkaudio: pre-calculate quantization factors
binkaudio: add some buffer overread checks.
atrac3: support float or int16 output using request_sample_fmt
atrac3: add CODEC_CAP_SUBFRAMES capability
atrac3: return appropriate error codes instead of -1
...
Conflicts:
libavcodec/atrac1.c
libavcodec/dca.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 08e3dea3f7f69309574dafc0af6671615e909720)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: fix signed overflow in avformat_find_stream_info()
vp8: fix signed overflows
motion_est: fix some signed overflows
dca: fix signed overflow in shift
aacdec: fix undefined shifts
bink: Check for various out of bound writes
bink: Check for out of bound writes when building tree
put_bits: fix invalid shift by 32 in flush_put_bits()
Conflicts:
libavcodec/bink.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (34 commits)
dpcm: return error if packet is too small
dpcm: use smaller data types for static tables
dpcm: use sol_table_16 directly instead of through the DPCMContext.
dpcm: replace short with int16_t
dpcm: check to make sure channels is 1 or 2.
dpcm: misc pretty-printing
dpcm: remove unnecessary variable by using bytestream functions.
dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
dpcm: consistently use the variable name 'n' for the next input byte.
dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
dpcm: calculate and check actual output data size prior to decoding.
dpcm: factor out the stereo flag calculation
dpcm: cosmetics: rename channel_number to ch
avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
lavf: Avoid using av_malloc(0) in av_dump_format
dxva2_h264: pass the correct 8x8 scaling lists
dca: NEON optimised high freq VQ decoding
avcodec: reject audio packets with NULL data and non-zero size
dxva: Add ability to enable workaround for older ATI cards
latmenc: Set latmBufferFullness to largest value to indicate it is not used
...
Conflicts:
libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dca: clear inactive subbands only once in qmf_32_subbands()
vf_unsharp: set default chroma size value to 5x5
vf_unsharp: fix out-of-buffer read
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>
* qatar/master: (23 commits)
avconv: Reformat s16 volume adjustment.
ARM: NEON optimised vector_fmac_scalar()
dca: use vector_fmac_scalar from dsputil
dsputil: add vector_fmac_scalar()
latmenc: Fix private options
vf_unsharp: store hsub/vsub in the filter context
vf_unsharp: adopt a more natural order of params in apply_unsharp()
vf_unsharp: rename method "unsharpen" to "apply_unsharp"
vf_scale: apply the same transform to the aspect during init that is applied per frame
vf_pad: fix "vsub" variable value computation
vf_scale: add a "sar" variable
lavfi: fix realloc size computation in avfilter_add_format()
vsrc_color: use internal timebase
lavfi: fix signature for avfilter_graph_parse() and avfilter_graph_config()
graphparser: prefer void * over AVClass * for log contexts
avfiltergraph: use meaningful error codes
avconv: Initialize return value for codec copy path.
fate: use 'run' helper for seek-test
fate: remove seek-mpeg2reuse test
Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
...
Conflicts:
doc/filters.texi
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/vf_scale.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
