* 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>
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>
This also allows to remove a linking dependency of libavfilter on
libavcodec.
Originally committed as revision 25789 to svn://svn.ffmpeg.org/ffmpeg/trunk
that flag to the dca codec. This capability when set
will make sure the codec will initialize the channel
configuration instead of trusting the container. This
fixes issue 2137 and issue 850.
Originally committed as revision 25320 to svn://svn.ffmpeg.org/ffmpeg/trunk
to verify the sync word the extension fsize field should be compared to
the core data length field.
Patch by nick.nbrereton@net
Originally committed as revision 24054 to svn://svn.ffmpeg.org/ffmpeg/trunk
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
This reorders the lfe_fir tables, and drops the mirrored half,
such that the loops in lfe_interpolation_fir() can be simplified.
The new loop structure should be easier to implement with SIMD.
Static data size is reduced by 2kB.
3% faster on Cortex-A8.
Originally committed as revision 22849 to svn://svn.ffmpeg.org/ffmpeg/trunk
This prevents gcc reloading the value from memory on each iteration
of the loop.
Originally committed as revision 22848 to svn://svn.ffmpeg.org/ffmpeg/trunk
Optimised implementations of the synth filter will require these
arrays 16-byte aligned.
Originally committed as revision 22826 to svn://svn.ffmpeg.org/ffmpeg/trunk
These macros are redundant. All uses are replaced with the generic
DECLARE_ALIGNED macro instead.
Originally committed as revision 22233 to svn://svn.ffmpeg.org/ffmpeg/trunk
It happens when the number of channels defined by DCAContext:acmod is lower
than DCAContext:prim_channels. In this case, dca_subsubframe() will call
qmf_32_subbands() using s->channel_order_tab[] entries equal to -1.
Originally committed as revision 22083 to svn://svn.ffmpeg.org/ffmpeg/trunk
Scaling (i)MDCT output has no runtime overhead and can be used to improve
performance of audio codecs. All the changes are only needed in
'ff_mdct_init' function and slow down initialization a bit.
Originally committed as revision 18855 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVPacket argument rather than a const uint8_t *buf + int buf_size. This allows
passing of packet-specific flags from demuxer to decoder, such as the keyframe
flag, which appears necessary to playback corePNG P-frames.
Patch by Thilo Borgmann thilo.borgmann googlemail com, see also the thread
"Google Summer of Code participation" on the mailinglist.
Originally committed as revision 18351 to svn://svn.ffmpeg.org/ffmpeg/trunk
Otherwise doxygen complains about ambiguous filenames when files exist
under the same name in different subdirectories.
Originally committed as revision 16912 to svn://svn.ffmpeg.org/ffmpeg/trunk
