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author | Michael Niedermayer <michaelni@gmx.at> | 2012-11-27 13:39:52 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-11-27 13:39:52 +0100 |
commit | 2684d2e3ea8a1a2863ae9842d072341b44b09829 (patch) | |
tree | 3d116023d439f92f473480c8f68f6d27f9f62344 /libavcodec/libmp3lame.c | |
parent | 257196209fe7d27ad22e18bf5757ffcad47dce6b (diff) | |
parent | 284ea790d89441fa1e6b2d72d3c1ed6d61972f0b (diff) | |
download | ffmpeg-2684d2e3ea8a1a2863ae9842d072341b44b09829.tar.gz |
Merge commit '284ea790d89441fa1e6b2d72d3c1ed6d61972f0b'
* commit '284ea790d89441fa1e6b2d72d3c1ed6d61972f0b':
dsputil: move vector_fmul_scalar() to AVFloatDSPContext in libavutil
aacenc: use the correct output buffer
aacdec: fix signed overflows in lcg_random()
base64: fix signed overflow in shift
Conflicts:
libavcodec/dsputil.c
libavutil/base64.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/libmp3lame.c')
-rw-r--r-- | libavcodec/libmp3lame.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index a0f39bb5f5..1447d6a10e 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -28,12 +28,12 @@ #include "libavutil/channel_layout.h" #include "libavutil/common.h" +#include "libavutil/float_dsp.h" #include "libavutil/intreadwrite.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "avcodec.h" #include "audio_frame_queue.h" -#include "dsputil.h" #include "internal.h" #include "mpegaudio.h" #include "mpegaudiodecheader.h" @@ -50,7 +50,7 @@ typedef struct LAMEContext { int reservoir; float *samples_flt[2]; AudioFrameQueue afq; - DSPContext dsp; + AVFloatDSPContext fdsp; } LAMEContext; @@ -168,7 +168,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext *avctx) if (ret < 0) goto error; - ff_dsputil_init(&s->dsp, avctx); + avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); return 0; error: @@ -206,10 +206,10 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, return AVERROR(EINVAL); } for (ch = 0; ch < avctx->channels; ch++) { - s->dsp.vector_fmul_scalar(s->samples_flt[ch], - (const float *)frame->data[ch], - 32768.0f, - FFALIGN(frame->nb_samples, 8)); + s->fdsp.vector_fmul_scalar(s->samples_flt[ch], + (const float *)frame->data[ch], + 32768.0f, + FFALIGN(frame->nb_samples, 8)); } ENCODE_BUFFER(lame_encode_buffer_float, float, s->samples_flt); break; |