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authorRonald S. Bultje <rsbultje@gmail.com>2013-01-20 15:41:52 -0800
committerRonald S. Bultje <rsbultje@gmail.com>2013-01-22 11:55:42 -0800
commitd56668bd80075615b89aff652fe8a576bf853ceb (patch)
tree9da3ed036b716dbaf33f5c9869578bedb6e393a2 /libavcodec/acelp_vectors.c
parent5959bfaca396ecaf63a8123055f499688b79cae3 (diff)
downloadffmpeg-d56668bd80075615b89aff652fe8a576bf853ceb.tar.gz
floatdsp: move scalarproduct_float from dsputil to avfloatdsp.
This makes the aac decoder and all voice codecs independent of dsputil.
Diffstat (limited to 'libavcodec/acelp_vectors.c')
-rw-r--r--libavcodec/acelp_vectors.c6
1 files changed, 3 insertions, 3 deletions
diff --git a/libavcodec/acelp_vectors.c b/libavcodec/acelp_vectors.c
index b50c5f3ffe..a85e45f4c7 100644
--- a/libavcodec/acelp_vectors.c
+++ b/libavcodec/acelp_vectors.c
@@ -23,8 +23,8 @@
#include <inttypes.h>
#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
-#include "dsputil.h"
#include "acelp_vectors.h"
const uint8_t ff_fc_2pulses_9bits_track1[16] =
@@ -183,7 +183,7 @@ void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem)
{
int i;
- float postfilter_energ = ff_scalarproduct_float_c(in, in, size);
+ float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
float gain_scale_factor = 1.0;
float mem = *gain_mem;
@@ -204,7 +204,7 @@ void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n)
{
int i;
- float scalefactor = ff_scalarproduct_float_c(in, in, n);
+ float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)