| Commit message (Collapse) | Author | Age |
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Still be less verbose about channel count change from 0 channels.
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Only inform the user that the number of channels changed if it was set
before.
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(cherry picked from commit 42e78d2296752293cd08c5727f76c3b37c68fb24)
Review-by: Benjamin Larsson
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 08e3dea3f7f69309574dafc0af6671615e909720)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* 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>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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* 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>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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* 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>
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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>
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* 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.
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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>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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* 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
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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>
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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>
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* 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>
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* qatar/master: (28 commits)
Replace usages of av_get_bits_per_sample_fmt() with av_get_bytes_per_sample().
x86: cabac: fix register constraints for 32-bit mode
cabac: move x86 asm to libavcodec/x86/cabac.h
x86: h264: cast pointers to intptr_t rather than int
x86: h264: remove hardcoded edi in decode_significance_8x8_x86()
x86: h264: remove hardcoded esi in decode_significance[_8x8]_x86()
x86: h264: remove hardcoded edx in decode_significance[_8x8]_x86()
x86: h264: remove hardcoded eax in decode_significance[_8x8]_x86()
x86: cabac: change 'a' constraint to 'r' in get_cabac_inline()
x86: cabac: remove hardcoded esi in get_cabac_inline()
x86: cabac: remove hardcoded edx in get_cabac_inline()
x86: cabac: remove unused macro parameter
x86: cabac: remove hardcoded ebx in inline asm
x86: cabac: remove hardcoded struct offsets from inline asm
cabac: remove inline asm under #if 0
cabac: remove BRANCHLESS_CABAC_DECODER switch
cabac: remove #if 0 cascade under never-set #ifdef ARCH_X86_DISABLED
document libswscale bump
error_resilience: skip last-MV predictor step if MVs are not available.
error_resilience: actually add counter when adding a MV predictor.
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Conflicts:
Changelog
libavcodec/error_resilience.c
libavfilter/defaults.c
libavfilter/vf_drawtext.c
libswscale/swscale.h
tests/ref/vsynth1/error
tests/ref/vsynth2/error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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av_get_bits_per_sample_fmt() is deprecated.
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* qatar/master: (25 commits)
Replace custom DEBUG preprocessor trickery by the standard one.
vorbis: Remove non-compiling debug statement.
vorbis: Remove pointless DEBUG #ifdef around debug output macros.
cook: Remove non-compiling debug output.
Remove pointless #ifdefs around function declarations in a header.
Replace #ifdef + av_log() combinations by av_dlog().
Replace custom debug output functions by av_dlog().
cook: Remove unused debug functions.
Remove stray extra arguments from av_dlog() invocations.
targa: fix big-endian build
v4l2: remove one forgotten use of AVFormatParameters.pix_fmt.
vfwcap: add a framerate private option.
v4l2: add a framerate private option.
libdc1394: add a framerate private option.
fbdev: add a framerate private option.
bktr: add a framerate private option.
oma: check avio_read() return value
nutdec: remove unused variable
Remove unused variables
swscale: allocate larger buffer to handle altivec overreads.
...
Conflicts:
ffmpeg.c
libavcodec/dca.c
libavcodec/dirac.c
libavcodec/error_resilience.c
libavcodec/h264.c
libavcodec/mpeg12.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/mpegvideo_enc.c
libavcodec/pthread.c
libavcodec/rv10.c
libavcodec/s302m.c
libavcodec/shorten.c
libavcodec/truemotion2.c
libavcodec/utils.c
libavdevice/dv1394.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/v4l2.c
libavformat/4xm.c
libavformat/apetag.c
libavformat/asfdec.c
libavformat/avidec.c
libavformat/mmf.c
libavformat/mpeg.c
libavformat/mpegenc.c
libavformat/mpegts.c
libavformat/oggdec.c
libavformat/oggparseogm.c
libavformat/rl2.c
libavformat/rmdec.c
libavformat/rpl.c
libavformat/rtpdec_latm.c
libavformat/sauce.c
libavformat/sol.c
libswscale/utils.c
tests/ref/vsynth1/error
tests/ref/vsynth2/error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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* 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>
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decoders.
Based on patches by clsid2 in ffdshow-tryout.
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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.
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* qatar/master: (23 commits)
ac3enc: correct the flipped sign in the ac3_fixed encoder
Eliminate pointless '#if 1' statements without matching '#else'.
Add AVX FFT implementation.
Increase alignment of av_malloc() as needed by AVX ASM.
Update x86inc.asm from x264 to allow AVX emulation using SSE and MMX.
mjpeg: Detect overreads in mjpeg_decode_scan() and error out.
documentation: extend documentation for ffmpeg -aspect option
APIChanges: update commit hashes for recent additions.
lavc: deprecate FF_*_TYPE macros in favor of AV_PICTURE_TYPE_* enums
aac: add headers needed for log2f()
lavc: remove FF_API_MB_Q cruft
lavc: remove FF_API_RATE_EMU cruft
lavc: remove FF_API_HURRY_UP cruft
pad: make the filter parametric
vsrc_movie: add key_frame and pict_type.
vsrc_movie: fix leak in request_frame()
lavfi: add key_frame and pict_type to AVFilterBufferRefVideo.
vsrc_buffer: add sample_aspect_ratio fields to arguments.
lavfi: add fieldorder filter
scale: make the filter parametric
...
Conflicts:
Changelog
doc/filters.texi
ffmpeg.c
libavcodec/ac3dec.h
libavcodec/dsputil.c
libavfilter/avfilter.h
libavfilter/vf_scale.c
libavfilter/vf_yadif.c
libavfilter/vsrc_buffer.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Signed-off-by: Reinhard Tartler <siretart@tauware.de>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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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>
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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>
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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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>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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DSPContext.vector_fmul_window()
DCADSPContext.lfe_fir()
SynthFilterContext.synth_filter_float()
Signed-off-by: Mans Rullgard <mans@mansr.com>
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None of these symbols should be accessed directly, so declare them as
hidden.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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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>
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git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
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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 7e06e0ede3b798f591634b277e8dfa6507b196de)
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Done to keep ABI compatible. Otherwise this is just silly
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Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit c2fcd0a7a4d0bda1a3306e40b70ce281a987df60)
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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 c73d99e672329c8f2df290736ffc474c360ac4ae)
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Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit e86e858111501650bb9ce8e39282e20c57bac913)
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DSPContext.vector_fmul_window()
DCADSPContext.lfe_fir()
SynthFilterContext.synth_filter_float()
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 80ba1ddb58b5923b9f36a6acd542affc4ca722eb)
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(cherry picked from commit b5ec6383432c907bed175f20e39af1a99dc75cfb)
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