floatdsp: move scalarproduct_float from dsputil to avfloatdsp.
[ffmpeg.git] / libavcodec / wmavoice.c
index 08d0600..ba778cd 100644 (file)
@@ -30,8 +30,8 @@
 #include <math.h>
 
 #include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
 #include "libavutil/mem.h"
-#include "dsputil.h"
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
@@ -523,7 +523,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
 
     /* find best fitting point in history */
     do {
-        dot = ff_scalarproduct_float_c(in, ptr, size);
+        dot = avpriv_scalarproduct_float_c(in, ptr, size);
         if (dot > optimal_gain) {
             optimal_gain  = dot;
             best_hist_ptr = ptr;
@@ -532,7 +532,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
 
     if (optimal_gain <= 0)
         return -1;
-    dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
+    dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
     if (dot <= 0) // would be 1.0
         return -1;
 
@@ -562,8 +562,8 @@ static float tilt_factor(const float *lpcs, int n_lpcs)
 {
     float rh0, rh1;
 
-    rh0 = 1.0     + ff_scalarproduct_float_c(lpcs,  lpcs,    n_lpcs);
-    rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
+    rh0 = 1.0     + avpriv_scalarproduct_float_c(lpcs,  lpcs,    n_lpcs);
+    rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
 
     return rh1 / rh0;
 }
@@ -656,7 +656,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
                              -1.8 * tilt_factor(coeffs, remainder - 1),
                              coeffs, remainder);
     }
-    sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
+    sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
+                                                               remainder));
     for (n = 0; n < remainder; n++)
         coeffs[n] *= sq;
 }
@@ -1320,7 +1321,8 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
     /* Calculate gain for adaptive & fixed codebook signal.
      * see ff_amr_set_fixed_gain(). */
     idx = get_bits(gb, 7);
-    fcb_gain = expf(ff_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) -
+    fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
+                                                 gain_coeff, 6) -
                     5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
     acb_gain = wmavoice_gain_codebook_acb[idx];
     pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],