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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-04-27 03:51:04 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-04-27 03:51:04 +0200 |
| commit | d7e5aebae7652ac766034f1d90e5a4f62677fb3c (patch) | |
| tree | b77ee45f34455cf9aa6e28105a7533ecc204b898 /libavcodec/binkaudio.c | |
| parent | 93c28a55fd84280d97c3c0dd7b0d546043242c34 (diff) | |
| parent | 79ee8977c25eee2408ef7b2822f377a983e4d65b (diff) | |
Merge remote branch 'qatar/master'
* 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>
Diffstat (limited to 'libavcodec/binkaudio.c')
| -rw-r--r-- | libavcodec/binkaudio.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavcodec/binkaudio.c b/libavcodec/binkaudio.c index d879efc2e0..bf1d412ed1 100644 --- a/libavcodec/binkaudio.c +++ b/libavcodec/binkaudio.c @@ -55,7 +55,7 @@ typedef struct { int num_bands; unsigned int *bands; float root; - DECLARE_ALIGNED(16, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; + DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE]; DECLARE_ALIGNED(16, short, previous)[BINK_BLOCK_MAX_SIZE / 16]; ///< coeffs from previous audio block float *coeffs_ptr[MAX_CHANNELS]; ///< pointers to the coeffs arrays for float_to_int16_interleave union { |