celp_math: Replace duplicate ff_dot_productf() by ff_scalarproduct_c()

libavcodec/Makefile

@@ -84,11 +84,11 @@ OBJS-$(CONFIG_ALAC_DECODER)            += alac.o
 OBJS-$(CONFIG_ALAC_ENCODER)            += alacenc.o
 OBJS-$(CONFIG_ALS_DECODER)             += alsdec.o bgmc.o mpeg4audio.o
 OBJS-$(CONFIG_AMRNB_DECODER)           += amrnbdec.o celp_filters.o   \
-                                          celp_math.o acelp_filters.o \
+                                          acelp_filters.o             \
                                           acelp_vectors.o             \
                                           acelp_pitch_delay.o
 OBJS-$(CONFIG_AMRWB_DECODER)           += amrwbdec.o celp_filters.o   \
-                                          celp_math.o acelp_filters.o \
+                                          acelp_filters.o             \
                                           acelp_vectors.o             \
                                           acelp_pitch_delay.o
 OBJS-$(CONFIG_AMV_DECODER)             += sp5xdec.o mjpegdec.o mjpeg.o
@@ -298,7 +298,7 @@ OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o pnm.o
 OBJS-$(CONFIG_PRORES_DECODER)          += proresdec.o proresdata.o proresdsp.o
 OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc.o proresdata.o proresdsp.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
-OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o celp_math.o         \
+OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
                                           celp_filters.o acelp_vectors.o \
                                           acelp_filters.o
 OBJS-$(CONFIG_QDM2_DECODER)            += qdm2.o
@@ -311,7 +311,7 @@ OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o \
                                           audio_frame_queue.o
-OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
+OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_filters.o
 OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
@@ -332,7 +332,7 @@ OBJS-$(CONFIG_SGI_DECODER)             += sgidec.o
 OBJS-$(CONFIG_SGI_ENCODER)             += sgienc.o rle.o
 OBJS-$(CONFIG_SHORTEN_DECODER)         += shorten.o
 OBJS-$(CONFIG_SIPR_DECODER)            += sipr.o acelp_pitch_delay.o \
-                                          celp_math.o acelp_vectors.o \
+                                          acelp_vectors.o \
                                           acelp_filters.o celp_filters.o \
                                           sipr16k.o
 OBJS-$(CONFIG_SMACKAUD_DECODER)        += smacker.o
@@ -408,7 +408,7 @@ OBJS-$(CONFIG_WMAV1_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
 OBJS-$(CONFIG_WMAV2_DECODER)           += wmadec.o wma.o wma_common.o aactab.o
 OBJS-$(CONFIG_WMAV2_ENCODER)           += wmaenc.o wma.o wma_common.o aactab.o
 OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
-                                          celp_math.o celp_filters.o \
+                                          celp_filters.o \
                                           acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o        \

libavcodec/acelp_pitch_delay.c

@@ -25,7 +25,6 @@
 #include "avcodec.h"
 #include "dsputil.h"
 #include "acelp_pitch_delay.h"
-#include "celp_math.h"
 
 int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
 {
@@ -120,7 +119,7 @@ float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
     // Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
     float val = fixed_gain_factor *
         exp2f(M_LOG2_10 * 0.05 *
-              (ff_dot_productf(pred_table, prediction_error, 4) +
+              (ff_scalarproduct_float_c(pred_table, prediction_error, 4) +
                energy_mean)) /
         sqrtf(fixed_mean_energy);
 

libavcodec/acelp_vectors.c

@@ -24,8 +24,8 @@
 
 #include "libavutil/common.h"
 #include "avcodec.h"
+#include "dsputil.h"
 #include "acelp_vectors.h"
-#include "celp_math.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_dot_productf(in, in, size);
+    float postfilter_energ = ff_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_dot_productf(in, in, n);
+    float scalefactor = ff_scalarproduct_float_c(in, in, n);
     if (scalefactor)
         scalefactor = sqrt(sum_of_squares / scalefactor);
     for (i = 0; i < n; i++)

libavcodec/amrnbdec.c

@@ -44,8 +44,8 @@
 #include <math.h>
 
 #include "avcodec.h"
+#include "dsputil.h"
 #include "libavutil/common.h"
-#include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
@@ -784,8 +784,8 @@ static int synthesis(AMRContext *p, float *lpc,
 
     // emphasize pitch vector contribution
     if (p->pitch_gain[4] > 0.5 && !overflow) {
-        float energy = ff_dot_productf(excitation, excitation,
-                                       AMR_SUBFRAME_SIZE);
+        float energy = ff_scalarproduct_float_c(excitation, excitation,
+                                                AMR_SUBFRAME_SIZE);
         float pitch_factor =
             p->pitch_gain[4] *
             (p->cur_frame_mode == MODE_12k2 ?
@@ -861,8 +861,8 @@ static float tilt_factor(float *lpc_n, float *lpc_d)
     ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
                                  LP_FILTER_ORDER);
 
-    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
-    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
+    rh0 = ff_scalarproduct_float_c(hf, hf,     AMR_TILT_RESPONSE);
+    rh1 = ff_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1);
 
     // The spec only specifies this check for 12.2 and 10.2 kbit/s
     // modes. But in the ref source the tilt is always non-negative.
@@ -882,8 +882,8 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
     int i;
     float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
 
-    float speech_gain       = ff_dot_productf(samples, samples,
-                                              AMR_SUBFRAME_SIZE);
+    float speech_gain       = ff_scalarproduct_float_c(samples, samples,
+                                                       AMR_SUBFRAME_SIZE);
 
     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
     const float *gamma_n, *gamma_d;                       // Formant filter factor table
@@ -988,8 +988,10 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
 
         p->fixed_gain[4] =
             ff_amr_set_fixed_gain(fixed_gain_factor,
-                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
-                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
+                                  ff_scalarproduct_float_c(p->fixed_vector,
+                                                           p->fixed_vector,
+                                                           AMR_SUBFRAME_SIZE) /
+                                  AMR_SUBFRAME_SIZE,
                        p->prediction_error,
                        energy_mean[p->cur_frame_mode], energy_pred_fac);
 

libavcodec/amrwbdec.c

@@ -28,8 +28,8 @@
 #include "libavutil/lfg.h"
 
 #include "avcodec.h"
+#include "dsputil.h"
 #include "lsp.h"
-#include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
@@ -585,10 +585,12 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
 static float voice_factor(float *p_vector, float p_gain,
                           float *f_vector, float f_gain)
 {
-    double p_ener = (double) ff_dot_productf(p_vector, p_vector,
-                                             AMRWB_SFR_SIZE) * p_gain * p_gain;
-    double f_ener = (double) ff_dot_productf(f_vector, f_vector,
-                                             AMRWB_SFR_SIZE) * f_gain * f_gain;
+    double p_ener = (double) ff_scalarproduct_float_c(p_vector, p_vector,
+                                                      AMRWB_SFR_SIZE) *
+                    p_gain * p_gain;
+    double f_ener = (double) ff_scalarproduct_float_c(f_vector, f_vector,
+                                                      AMRWB_SFR_SIZE) *
+                    f_gain * f_gain;
 
     return (p_ener - f_ener) / (p_ener + f_ener);
 }
@@ -756,8 +758,8 @@ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
     /* emphasize pitch vector contribution in low bitrate modes */
     if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
         int i;
-        float energy = ff_dot_productf(excitation, excitation,
-                                       AMRWB_SFR_SIZE);
+        float energy = ff_scalarproduct_float_c(excitation, excitation,
+                                                AMRWB_SFR_SIZE);
 
         // XXX: Weird part in both ref code and spec. A unknown parameter
         // {beta} seems to be identical to the current pitch gain
@@ -816,8 +818,9 @@ static void upsample_5_4(float *out, const float *in, int o_size)
         i++;
 
         for (k = 1; k < 5; k++) {
-            out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
-                                     UPS_MEM_SIZE);
+            out[i] = ff_scalarproduct_float_c(in0 + int_part,
+                                              upsample_fir[4 - frac_part],
+                                              UPS_MEM_SIZE);
             int_part++;
             frac_part--;
             i++;
@@ -843,8 +846,8 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
     if (ctx->fr_cur_mode == MODE_23k85)
         return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
 
-    tilt = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
-           ff_dot_productf(synth, synth, AMRWB_SFR_SIZE);
+    tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
+           ff_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
 
     /* return gain bounded by [0.1, 1.0] */
     return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
@@ -863,7 +866,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
                                  const float *synth_exc, float hb_gain)
 {
     int i;
-    float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
+    float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE);
 
     /* Generate a white-noise excitation */
     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
@@ -1156,8 +1159,10 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
 
         ctx->fixed_gain[0] =
             ff_amr_set_fixed_gain(fixed_gain_factor,
-                       ff_dot_productf(ctx->fixed_vector, ctx->fixed_vector,
-                                       AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE,
+                                  ff_scalarproduct_float_c(ctx->fixed_vector,
+                                                           ctx->fixed_vector,
+                                                           AMRWB_SFR_SIZE) /
+                                  AMRWB_SFR_SIZE,
                        ctx->prediction_error,
                        ENERGY_MEAN, energy_pred_fac);
 

libavcodec/celp_math.c

@@ -86,14 +86,3 @@ int ff_log2(uint32_t value)
 
     return (power_int << 15) + value;
 }
-
-float ff_dot_productf(const float* a, const float* b, int length)
-{
-    float sum = 0;
-    int i;
-
-    for(i=0; i<length; i++)
-        sum += a[i] * b[i];
-
-    return sum;
-}

libavcodec/celp_math.h

@@ -55,14 +55,4 @@ static inline int bidir_sal(int value, int offset)
     else           return value <<  offset;
 }
 
-/**
- * Return the dot product.
- * @param a input data array
- * @param b input data array
- * @param length number of elements
- *
- * @return dot product = sum of elementwise products
- */
-float ff_dot_productf(const float* a, const float* b, int length);
-
 #endif /* AVCODEC_CELP_MATH_H */

libavcodec/dsputil.c

@@ -2424,7 +2424,7 @@ static void butterflies_float_interleave_c(float *dst, const float *src0,
     }
 }
 
-static float scalarproduct_float_c(const float *v1, const float *v2, int len)
+float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
 {
     float p = 0.0;
     int i;
@@ -2877,7 +2877,7 @@ av_cold void ff_dsputil_init(DSPContext* c, AVCodecContext *avctx)
     c->scalarproduct_and_madd_int16 = scalarproduct_and_madd_int16_c;
     c->apply_window_int16 = apply_window_int16_c;
     c->vector_clip_int32 = vector_clip_int32_c;
-    c->scalarproduct_float = scalarproduct_float_c;
+    c->scalarproduct_float = ff_scalarproduct_float_c;
     c->butterflies_float = butterflies_float_c;
     c->butterflies_float_interleave = butterflies_float_interleave_c;
     c->vector_fmul_scalar = vector_fmul_scalar_c;

libavcodec/dsputil.h

@@ -549,6 +549,17 @@ void ff_dsputil_init(DSPContext* p, AVCodecContext *avctx);
 
 int ff_check_alignment(void);
 
+/**
+ * Return the scalar product of two vectors.
+ *
+ * @param v1  first input vector
+ * @param v2  first input vector
+ * @param len number of elements
+ *
+ * @return sum of elementwise products
+ */
+float ff_scalarproduct_float_c(const float *v1, const float *v2, int len);
+
 /**
  * permute block according to permuatation.
  * @param last last non zero element in scantable order

libavcodec/lsp.c

@@ -27,7 +27,6 @@
 #define FRAC_BITS 14
 #include "mathops.h"
 #include "lsp.h"
-#include "celp_math.h"
 
 void ff_acelp_reorder_lsf(int16_t* lsfq, int lsfq_min_distance, int lsfq_min, int lsfq_max, int lp_order)
 {

libavcodec/qcelpdec.c

@@ -32,10 +32,8 @@
 #include "avcodec.h"
 #include "internal.h"
 #include "get_bits.h"
-
+#include "dsputil.h"
 #include "qcelpdata.h"
-
-#include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
@@ -401,8 +399,9 @@ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
 
     for (i = 0; i < 160; i += 40)
         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
-                                                ff_dot_productf(v_ref + i,
-                                                                v_ref + i, 40),
+                                                ff_scalarproduct_float_c(v_ref + i,
+                                                                         v_ref + i,
+                                                                         40),
                                                 40);
 }
 
@@ -678,8 +677,8 @@ static void postfilter(QCELPContext *q, float *samples, float *lpc)
     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
 
     ff_adaptive_gain_control(samples, pole_out + 10,
-                             ff_dot_productf(q->formant_mem + 10,
-                                             q->formant_mem + 10, 160),
+                             ff_scalarproduct_float_c(q->formant_mem + 10,
+                                                      q->formant_mem + 10, 160),
                              160, 0.9375, &q->postfilter_agc_mem);
 }
 

libavcodec/ra288.c

@@ -25,7 +25,6 @@
 #include "get_bits.h"
 #include "ra288.h"
 #include "lpc.h"
-#include "celp_math.h"
 #include "celp_filters.h"
 
 #define MAX_BACKWARD_FILTER_ORDER  36
@@ -74,7 +73,7 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx)
 static void convolve(float *tgt, const float *src, int len, int n)
 {
     for (; n >= 0; n--)
-        tgt[n] = ff_dot_productf(src, src - n, len);
+        tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
 
 }
 
@@ -103,7 +102,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
     for (i=0; i < 5; i++)
         buffer[i] = codetable[cb_coef][i] * sumsum;
 
-    sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
+    sum = ff_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.);
 
     sum = FFMAX(sum, 1);
 

libavcodec/sipr.c

@@ -32,7 +32,6 @@
 #include "dsputil.h"
 
 #include "lsp.h"
-#include "celp_math.h"
 #include "acelp_vectors.h"
 #include "acelp_pitch_delay.h"
 #include "acelp_filters.h"
@@ -411,7 +410,7 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
                               SUBFR_SIZE);
 
         avg_energy =
-            (0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
+            (0.01 + ff_scalarproduct_float_c(fixed_vector, fixed_vector, SUBFR_SIZE)) /
                 SUBFR_SIZE;
 
         ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
@@ -453,9 +452,9 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
 
     if (ctx->mode == MODE_5k0) {
         for (i = 0; i < subframe_count; i++) {
-            float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
-                                           ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
-                                           SUBFR_SIZE);
+            float energy = ff_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
+                                                    ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
+                                                    SUBFR_SIZE);
             ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
                                      &synth[i * SUBFR_SIZE], energy,
                                      SUBFR_SIZE, 0.9, &ctx->postfilter_agc);

libavcodec/sipr16k.c

@@ -26,8 +26,9 @@
 #include "sipr.h"
 #include "libavutil/common.h"
 #include "libavutil/mathematics.h"
+#include "dsputil.h"
 #include "lsp.h"
-#include "celp_math.h"
+#include "celp_filters.h"
 #include "acelp_vectors.h"
 #include "acelp_pitch_delay.h"
 #include "acelp_filters.h"
@@ -163,10 +164,10 @@ static float acelp_decode_gain_codef(float gain_corr_factor, const float *fc_v,
                                      int subframe_size, int ma_pred_order)
 {
     mr_energy +=
-        ff_dot_productf(quant_energy, ma_prediction_coeff, ma_pred_order);
+        ff_scalarproduct_float_c(quant_energy, ma_prediction_coeff, ma_pred_order);
 
     mr_energy = gain_corr_factor * exp(M_LN10 / 20. * mr_energy) /
-        sqrt((0.01 + ff_dot_productf(fc_v, fc_v, subframe_size)));
+        sqrt((0.01 + ff_scalarproduct_float_c(fc_v, fc_v, subframe_size)));
     return mr_energy;
 }
 

libavcodec/wmavoice.c

@@ -28,11 +28,12 @@
 #define UNCHECKED_BITSTREAM_READER 1
 
 #include <math.h>
+
+#include "dsputil.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "put_bits.h"
 #include "wmavoice_data.h"
-#include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_vectors.h"
 #include "acelp_filters.h"
@@ -518,7 +519,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
 
     /* find best fitting point in history */
     do {
-        dot = ff_dot_productf(in, ptr, size);
+        dot = ff_scalarproduct_float_c(in, ptr, size);
         if (dot > optimal_gain) {
             optimal_gain  = dot;
             best_hist_ptr = ptr;
@@ -527,7 +528,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
 
     if (optimal_gain <= 0)
         return -1;
-    dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size);
+    dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
     if (dot <= 0) // would be 1.0
         return -1;
 
@@ -557,8 +558,8 @@ static float tilt_factor(const float *lpcs, int n_lpcs)
 {
     float rh0, rh1;
 
-    rh0 = 1.0     + ff_dot_productf(lpcs,  lpcs,    n_lpcs);
-    rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1);
+    rh0 = 1.0     + ff_scalarproduct_float_c(lpcs,  lpcs,    n_lpcs);
+    rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
 
     return rh1 / rh0;
 }
@@ -651,7 +652,7 @@ 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_dot_productf(coeffs, coeffs, remainder));
+    sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
     for (n = 0; n < remainder; n++)
         coeffs[n] *= sq;
 }
@@ -1315,7 +1316,7 @@ 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_dot_productf(s->gain_pred_err, gain_coeff, 6) -
+    fcb_gain = expf(ff_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],