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author | Michael Niedermayer <michaelni@gmx.at> | 2012-03-01 01:13:16 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-03-01 03:17:11 +0100 |
commit | 79ae084e9b930f8b53ae0499c6a06636d194574d (patch) | |
tree | e7d829e566b01ef7e84a12b06a2bcb87a8164059 /libavcodec/mpegaudiodec.c | |
parent | a77c8ade2ee20fc6149e4c689a3f196f53e85273 (diff) | |
parent | 882abda5a26ffb8e3d1c5852dfa7cdad0a291d2d (diff) | |
download | ffmpeg-79ae084e9b930f8b53ae0499c6a06636d194574d.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/mpegaudiodec.c')
-rw-r--r-- | libavcodec/mpegaudiodec.c | 7 |
1 files changed, 7 insertions, 0 deletions
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 75a3c5d372..00565875fc 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -31,6 +31,7 @@ #include "get_bits.h" #include "mathops.h" #include "mpegaudiodsp.h" +#include "dsputil.h" /* * TODO: @@ -82,6 +83,7 @@ typedef struct MPADecodeContext { int err_recognition; AVCodecContext* avctx; MPADSPContext mpadsp; + DSPContext dsp; AVFrame frame; } MPADecodeContext; @@ -434,6 +436,7 @@ static av_cold int decode_init(AVCodecContext * avctx) s->avctx = avctx; ff_mpadsp_init(&s->mpadsp); + ff_dsputil_init(&s->dsp, avctx); avctx->sample_fmt= OUT_FMT; s->err_recognition = avctx->err_recognition; @@ -1155,6 +1158,9 @@ found2: /* ms stereo ONLY */ /* NOTE: the 1/sqrt(2) normalization factor is included in the global gain */ +#if CONFIG_FLOAT + s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576); +#else tab0 = g0->sb_hybrid; tab1 = g1->sb_hybrid; for (i = 0; i < 576; i++) { @@ -1163,6 +1169,7 @@ found2: tab0[i] = tmp0 + tmp1; tab1[i] = tmp0 - tmp1; } +#endif } } |