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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-05-20 07:42:04 +0400 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-05-20 07:48:22 +0400 |
| commit | 80d156d7fdc44b09783ba242fe2681a6d4cc8df5 (patch) | |
| tree | 7881b70297c87daa2f6d6f4790afaf438c53b3aa /libavcodec/mpegaudiodec_float.c | |
| parent | 6efb29686fc9a7f76480405df8fe7eaa7a9dd4cf (diff) | |
| parent | 984ece7503597d30e6f3bdeb67e337ea1616f880 (diff) | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
qdm2: Use floating point synthesis filter.
h264: correct border check.
h264: fix loopfilter with threading at slice boundaries.
Fix ff_mpa_synth_filter_fixed() prototype
Rename costablegen.c ---> cos_tablegen.c.
Collapse tableprint.c into tableprint.h.
Simplify trig table rules
Remove potentially unstable filenames from comments in generated files.
Ignore generated tables and generated table generator programs.
Simplify CLEANFILES make variable by using wildcards.
Remove silly insults from avformat_version() Doxygen documentation.
mpegaudiodsp: fix x86 and ppc makefiles
configure: Adjust AVX assembler check.
mpegaudio: remove unused version of SAME_HEADER_MASK
mpegaudio: remove useless #undef at end of file
asfdec: add missing #include for av_bswap32()
mpegaudio: merge two #if CONFIG_FLOAT blocks
mpegaudio: move some struct definitions from mpegaudio.h
Move some mpegaudio functions to new mpegaudiodsp subsystem
Conflicts:
libavcodec/h264.c
libavcodec/x86/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/mpegaudiodec_float.c')
| -rw-r--r-- | libavcodec/mpegaudiodec_float.c | 19 |
1 files changed, 0 insertions, 19 deletions
diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c index 183e5540c2..26454190f5 100644 --- a/libavcodec/mpegaudiodec_float.c +++ b/libavcodec/mpegaudiodec_float.c @@ -22,25 +22,6 @@ #define CONFIG_FLOAT 1 #include "mpegaudiodec.c" -void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr, - int *synth_buf_offset, - float *window, int *dither_state, - float *samples, int incr, - float sb_samples[SBLIMIT]) -{ - float *synth_buf; - int offset; - - offset = *synth_buf_offset; - synth_buf = synth_buf_ptr + offset; - - s->dct.dct32(synth_buf, sb_samples); - s->apply_window_mp3(synth_buf, window, dither_state, samples, incr); - - offset = (offset - 32) & 511; - *synth_buf_offset = offset; -} - static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g) { |