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authorMichael Niedermayer <michaelni@gmx.at>2012-11-27 13:39:52 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-11-27 13:39:52 +0100
commit2684d2e3ea8a1a2863ae9842d072341b44b09829 (patch)
tree3d116023d439f92f473480c8f68f6d27f9f62344 /libavcodec/libmp3lame.c
parent257196209fe7d27ad22e18bf5757ffcad47dce6b (diff)
parent284ea790d89441fa1e6b2d72d3c1ed6d61972f0b (diff)
downloadffmpeg-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.c14
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;