/*
  * G.729, G729 Annex D postfilter
  * Copyright (c) 2008 Vladimir Voroshilov
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 #include <inttypes.h>
 #include <limits.h>
 
 #include "avcodec.h"
 #include "g729.h"
 #include "acelp_pitch_delay.h"
 #include "g729postfilter.h"
 #include "celp_math.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
 #include "celp_filters.h"
 
 #define FRAC_BITS 15
 #include "mathops.h"
 
 /**
  * short interpolation filter (of length 33, according to spec)
  * for computing signal with non-integer delay
  */
 static const int16_t ff_g729_interp_filt_short[(ANALYZED_FRAC_DELAYS+1)*SHORT_INT_FILT_LEN] = {
       0, 31650, 28469, 23705, 18050, 12266,  7041,  2873,
       0, -1597, -2147, -1992, -1492,  -933,  -484,  -188,
 };
 
 /**
  * long interpolation filter (of length 129, according to spec)
  * for computing signal with non-integer delay
  */
 static const int16_t ff_g729_interp_filt_long[(ANALYZED_FRAC_DELAYS+1)*LONG_INT_FILT_LEN] = {
    0, 31915, 29436, 25569, 20676, 15206,  9639,  4439,
    0, -3390, -5579, -6549, -6414, -5392, -3773, -1874,
    0,  1595,  2727,  3303,  3319,  2850,  2030,  1023,
    0,  -887, -1527, -1860, -1876, -1614, -1150,  -579,
    0,   501,   859,  1041,  1044,   892,   631,   315,
    0,  -266,  -453,  -543,  -538,  -455,  -317,  -156,
    0,   130,   218,   258,   253,   212,   147,    72,
    0,   -59,  -101,  -122,  -123,  -106,   -77,   -40,
 };
 
 /**
  * formant_pp_factor_num_pow[i] = FORMANT_PP_FACTOR_NUM^(i+1)
  */
 static const int16_t formant_pp_factor_num_pow[10]= {
   /* (0.15) */
   18022, 9912, 5451, 2998, 1649, 907, 499, 274, 151, 83
 };
 
 /**
  * formant_pp_factor_den_pow[i] = FORMANT_PP_FACTOR_DEN^(i+1)
  */
 static const int16_t formant_pp_factor_den_pow[10] = {
   /* (0.15) */
   22938, 16057, 11240, 7868, 5508, 3856, 2699, 1889, 1322, 925
 };
 
 /**
  * \brief Residual signal calculation (4.2.1 if G.729)
  * \param out [out] output data filtered through A(z/FORMANT_PP_FACTOR_NUM)
  * \param filter_coeffs (3.12) A(z/FORMANT_PP_FACTOR_NUM) filter coefficients
  * \param in input speech data to process
  * \param subframe_size size of one subframe
  *
  * \note in buffer must contain 10 items of previous speech data before top of the buffer
  * \remark It is safe to pass the same buffer for input and output.
  */
 static void residual_filter(int16_t* out, const int16_t* filter_coeffs, const int16_t* in,
                             int subframe_size)
 {
     int i, n;
 
     for (n = subframe_size - 1; n >= 0; n--) {
         int sum = 0x800;
         for (i = 0; i < 10; i++)
             sum += filter_coeffs[i] * in[n - i - 1];
 
         out[n] = in[n] + (sum >> 12);
     }
 }
 
 /**
  * \brief long-term postfilter (4.2.1)
  * \param dsp initialized DSP context
  * \param pitch_delay_int integer part of the pitch delay in the first subframe
  * \param residual filtering input data
  * \param residual_filt [out] speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter
  * \param subframe_size size of subframe
  *
  * \return 0 if long-term prediction gain is less than 3dB, 1 -  otherwise
  */

 static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,

                                 const int16_t* residual, int16_t *residual_filt,
                                 int subframe_size)
 {

     int i, k, tmp, tmp2;

     int sum;
     int L_temp0;
     int L_temp1;
     int64_t L64_temp0;
     int64_t L64_temp1;
     int16_t shift;
     int corr_int_num, corr_int_den;
 
     int ener;
     int16_t sh_ener;
 
     int16_t gain_num,gain_den; //selected signal's gain numerator and denominator
     int16_t sh_gain_num, sh_gain_den;
     int gain_num_square;
 
     int16_t gain_long_num,gain_long_den; //filtered through long interpolation filter signal's gain numerator and denominator
     int16_t sh_gain_long_num, sh_gain_long_den;
 
     int16_t best_delay_int, best_delay_frac;
 
     int16_t delayed_signal_offset;
     int lt_filt_factor_a, lt_filt_factor_b;
 
     int16_t * selected_signal;
     const int16_t * selected_signal_const; //Necessary to avoid compiler warning
 
     int16_t sig_scaled[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
     int16_t delayed_signal[ANALYZED_FRAC_DELAYS][SUBFRAME_SIZE+1];
     int corr_den[ANALYZED_FRAC_DELAYS][2];
 
     tmp = 0;
     for(i=0; i<subframe_size + RES_PREV_DATA_SIZE; i++)
         tmp |= FFABS(residual[i]);
 
     if(!tmp)
         shift = 3;
     else
         shift = av_log2(tmp) - 11;
 
     if (shift > 0)
         for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
             sig_scaled[i] = residual[i] >> shift;
     else
         for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
             sig_scaled[i] = residual[i] << -shift;
 
     /* Start of best delay searching code */
     gain_num = 0;
 

     ener = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,

                                     sig_scaled + RES_PREV_DATA_SIZE,

                                     subframe_size);

     if (ener) {
         sh_ener = FFMAX(av_log2(ener) - 14, 0);
         ener >>= sh_ener;
         /* Search for best pitch delay.
 
                        sum{ r(n) * r(k,n) ] }^2
            R'(k)^2 := -------------------------
                        sum{ r(k,n) * r(k,n) }
 
 
            R(T)    :=  sum{ r(n) * r(n-T) ] }
 
 
            where
            r(n-T) is integer delayed signal with delay T
            r(k,n) is non-integer delayed signal with integer delay best_delay
            and fractional delay k */
 
         /* Find integer delay best_delay which maximizes correlation R(T).
 
            This is also equals to numerator of R'(0),
            since the fine search (second step) is done with 1/8
            precision around best_delay. */
         corr_int_num = 0;
         best_delay_int = pitch_delay_int - 1;
         for (i = pitch_delay_int - 1; i <= pitch_delay_int + 1; i++) {

             sum = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,

                                            sig_scaled + RES_PREV_DATA_SIZE - i,

                                            subframe_size);

             if (sum > corr_int_num) {
                 corr_int_num = sum;
                 best_delay_int = i;
             }
         }
         if (corr_int_num) {
             /* Compute denominator of pseudo-normalized correlation R'(0). */

             corr_int_den = adsp->scalarproduct_int16(sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,

                                                     sig_scaled - best_delay_int + RES_PREV_DATA_SIZE,

                                                     subframe_size);

 
             /* Compute signals with non-integer delay k (with 1/8 precision),
                where k is in [0;6] range.
                Entire delay is qual to best_delay+(k+1)/8
                This is archieved by applying an interpolation filter of
                legth 33 to source signal. */
             for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
                 ff_acelp_interpolate(&delayed_signal[k][0],
                                      &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int],
                                      ff_g729_interp_filt_short,
                                      ANALYZED_FRAC_DELAYS+1,
                                      8 - k - 1,
                                      SHORT_INT_FILT_LEN,
                                      subframe_size + 1);
             }
 
             /* Compute denominator of pseudo-normalized correlation R'(k).
 
                  corr_den[k][0] is square root of R'(k) denominator, for int(T) == int(T0)
                  corr_den[k][1] is square root of R'(k) denominator, for int(T) == int(T0)+1
 
               Also compute maximum value of above denominators over all k. */
             tmp = corr_int_den;
             for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {

                 sum = adsp->scalarproduct_int16(&delayed_signal[k][1],

                                                &delayed_signal[k][1],

                                                subframe_size - 1);

                 corr_den[k][0] = sum + delayed_signal[k][0            ] * delayed_signal[k][0            ];
                 corr_den[k][1] = sum + delayed_signal[k][subframe_size] * delayed_signal[k][subframe_size];
 
                 tmp = FFMAX3(tmp, corr_den[k][0], corr_den[k][1]);
             }
 
             sh_gain_den = av_log2(tmp) - 14;
             if (sh_gain_den >= 0) {
 
                 sh_gain_num =  FFMAX(sh_gain_den, sh_ener);
                 /* Loop through all k and find delay that maximizes
                    R'(k) correlation.
                    Search is done in [int(T0)-1; intT(0)+1] range
                    with 1/8 precision. */
                 delayed_signal_offset = 1;
                 best_delay_frac = 0;
                 gain_den = corr_int_den >> sh_gain_den;
                 gain_num = corr_int_num >> sh_gain_num;
                 gain_num_square = gain_num * gain_num;
                 for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
                     for (i = 0; i < 2; i++) {
                         int16_t gain_num_short, gain_den_short;
                         int gain_num_short_square;
                         /* Compute numerator of pseudo-normalized
                            correlation R'(k). */

                         sum = adsp->scalarproduct_int16(&delayed_signal[k][i],

                                                        sig_scaled + RES_PREV_DATA_SIZE,

                                                        subframe_size);

                         gain_num_short = FFMAX(sum >> sh_gain_num, 0);
 
                         /*
                                       gain_num_short_square                gain_num_square
                            R'(T)^2 = -----------------------, max R'(T)^2= --------------
                                            den                                 gain_den
                         */
                         gain_num_short_square = gain_num_short * gain_num_short;
                         gain_den_short = corr_den[k][i] >> sh_gain_den;
 
                         tmp = MULL(gain_num_short_square, gain_den, FRAC_BITS);
                         tmp2 = MULL(gain_num_square, gain_den_short, FRAC_BITS);
 
                         // R'(T)^2 > max R'(T)^2
                         if (tmp > tmp2) {
                             gain_num = gain_num_short;
                             gain_den = gain_den_short;
                             gain_num_square = gain_num_short_square;
                             delayed_signal_offset = i;
                             best_delay_frac = k + 1;
                         }
                     }
                 }
 
                 /*
                        R'(T)^2
                   2 * --------- < 1
                         R(0)
                 */
                 L64_temp0 =  (int64_t)gain_num_square  << ((sh_gain_num << 1) + 1);
                 L64_temp1 = ((int64_t)gain_den * ener) << (sh_gain_den + sh_ener);
                 if (L64_temp0 < L64_temp1)
                     gain_num = 0;
             } // if(sh_gain_den >= 0)
         } // if(corr_int_num)
     } // if(ener)
     /* End of best delay searching code  */
 
     if (!gain_num) {
         memcpy(residual_filt, residual + RES_PREV_DATA_SIZE, subframe_size * sizeof(int16_t));
 
         /* Long-term prediction gain is less than 3dB. Long-term postfilter is disabled. */
         return 0;
     }
     if (best_delay_frac) {
         /* Recompute delayed signal with an interpolation filter of length 129. */
         ff_acelp_interpolate(residual_filt,
                              &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int + delayed_signal_offset],
                              ff_g729_interp_filt_long,
                              ANALYZED_FRAC_DELAYS + 1,
                              8 - best_delay_frac,
                              LONG_INT_FILT_LEN,
                              subframe_size + 1);
         /* Compute R'(k) correlation's numerator. */

         sum = adsp->scalarproduct_int16(residual_filt,

                                        sig_scaled + RES_PREV_DATA_SIZE,

                                        subframe_size);

 
         if (sum < 0) {
             gain_long_num = 0;
             sh_gain_long_num = 0;
         } else {
             tmp = FFMAX(av_log2(sum) - 14, 0);
             sum >>= tmp;
             gain_long_num = sum;
             sh_gain_long_num = tmp;
         }
 
         /* Compute R'(k) correlation's denominator. */

         sum = adsp->scalarproduct_int16(residual_filt, residual_filt, subframe_size);

 
         tmp = FFMAX(av_log2(sum) - 14, 0);
         sum >>= tmp;
         gain_long_den = sum;
         sh_gain_long_den = tmp;
 
         /* Select between original and delayed signal.
            Delayed signal will be selected if it increases R'(k)
            correlation. */
         L_temp0 = gain_num * gain_num;
         L_temp0 = MULL(L_temp0, gain_long_den, FRAC_BITS);
 
         L_temp1 = gain_long_num * gain_long_num;
         L_temp1 = MULL(L_temp1, gain_den, FRAC_BITS);
 
         tmp = ((sh_gain_long_num - sh_gain_num) << 1) - (sh_gain_long_den - sh_gain_den);
         if (tmp > 0)
             L_temp0 >>= tmp;
         else
             L_temp1 >>= -tmp;
 
         /* Check if longer filter increases the values of R'(k). */
         if (L_temp1 > L_temp0) {
             /* Select long filter. */
             selected_signal = residual_filt;
             gain_num = gain_long_num;
             gain_den = gain_long_den;
             sh_gain_num = sh_gain_long_num;
             sh_gain_den = sh_gain_long_den;
         } else
             /* Select short filter. */
             selected_signal = &delayed_signal[best_delay_frac-1][delayed_signal_offset];
 
         /* Rescale selected signal to original value. */
         if (shift > 0)
             for (i = 0; i < subframe_size; i++)
                 selected_signal[i] <<= shift;
         else
             for (i = 0; i < subframe_size; i++)
                 selected_signal[i] >>= -shift;
 
         /* necessary to avoid compiler warning */
         selected_signal_const = selected_signal;
     } // if(best_delay_frac)
     else
         selected_signal_const = residual + RES_PREV_DATA_SIZE - (best_delay_int + 1 - delayed_signal_offset);
 #ifdef G729_BITEXACT
     tmp = sh_gain_num - sh_gain_den;
     if (tmp > 0)
         gain_den >>= tmp;
     else
         gain_num >>= -tmp;
 
     if (gain_num > gain_den)
         lt_filt_factor_a = MIN_LT_FILT_FACTOR_A;
     else {
         gain_num >>= 2;
         gain_den >>= 1;
         lt_filt_factor_a = (gain_den << 15) / (gain_den + gain_num);
     }
 #else

     L64_temp0 = (((int64_t)gain_num) << sh_gain_num) >> 1;

     L64_temp1 = ((int64_t)gain_den) << sh_gain_den;
     lt_filt_factor_a = FFMAX((L64_temp1 << 15) / (L64_temp1 + L64_temp0), MIN_LT_FILT_FACTOR_A);
 #endif
 
     /* Filter through selected filter. */
     lt_filt_factor_b = 32767 - lt_filt_factor_a + 1;
 
     ff_acelp_weighted_vector_sum(residual_filt, residual + RES_PREV_DATA_SIZE,
                                  selected_signal_const,
                                  lt_filt_factor_a, lt_filt_factor_b,
                                  1<<14, 15, subframe_size);
 
     // Long-term prediction gain is larger than 3dB.
     return 1;
 }
 
 /**
  * \brief Calculate reflection coefficient for tilt compensation filter (4.2.3).
  * \param dsp initialized DSP context
  * \param lp_gn (3.12) coefficients of A(z/FORMANT_PP_FACTOR_NUM) filter
  * \param lp_gd (3.12) coefficients of A(z/FORMANT_PP_FACTOR_DEN) filter
  * \param speech speech to update
  * \param subframe_size size of subframe
  *
  * \return (3.12) reflection coefficient
  *
  * \remark The routine also calculates the gain term for the short-term
  *         filter (gf) and multiplies the speech data by 1/gf.
  *
  * \note All members of lp_gn, except 10-19 must be equal to zero.
  */

 static int16_t get_tilt_comp(AudioDSPContext *adsp, int16_t *lp_gn,

                              const int16_t *lp_gd, int16_t* speech,
                              int subframe_size)
 {
     int rh1,rh0; // (3.12)
     int temp;
     int i;
     int gain_term;
 
     lp_gn[10] = 4096; //1.0 in (3.12)
 
     /* Apply 1/A(z/FORMANT_PP_FACTOR_DEN) filter to hf. */

     ff_celp_lp_synthesis_filter(lp_gn + 11, lp_gd + 1, lp_gn + 11, 22, 10, 0, 0, 0x800);

     /* Now lp_gn (starting with 10) contains impulse response
        of A(z/FORMANT_PP_FACTOR_NUM)/A(z/FORMANT_PP_FACTOR_DEN) filter. */
 

     rh0 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 10, 20);
     rh1 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 11, 20);

 
     /* downscale to avoid overflow */
     temp = av_log2(rh0) - 14;
     if (temp > 0) {
         rh0 >>= temp;
         rh1 >>= temp;
     }
 
     if (FFABS(rh1) > rh0 || !rh0)
         return 0;
 
     gain_term = 0;
     for (i = 0; i < 20; i++)
         gain_term += FFABS(lp_gn[i + 10]);
     gain_term >>= 2; // (3.12) -> (5.10)
 
     if (gain_term > 0x400) { // 1.0 in (5.10)
         temp = 0x2000000 / gain_term; // 1.0/gain_term in (0.15)
         for (i = 0; i < subframe_size; i++)
             speech[i] = (speech[i] * temp + 0x4000) >> 15;
     }
 
     return -(rh1 << 15) / rh0;
 }
 
 /**
  * \brief Apply tilt compensation filter (4.2.3).
  * \param res_pst [in/out] residual signal (partially filtered)
  * \param k1 (3.12) reflection coefficient
  * \param subframe_size size of subframe
  * \param ht_prev_data previous data for 4.2.3, equation 86
  *
  * \return new value for ht_prev_data
 */
 static int16_t apply_tilt_comp(int16_t* out, int16_t* res_pst, int refl_coeff,
                                int subframe_size, int16_t ht_prev_data)
 {
     int tmp, tmp2;
     int i;
     int gt, ga;
     int fact, sh_fact;
 
     if (refl_coeff > 0) {
         gt = (refl_coeff * G729_TILT_FACTOR_PLUS + 0x4000) >> 15;
         fact = 0x4000; // 0.5 in (0.15)
         sh_fact = 15;
     } else {
         gt = (refl_coeff * G729_TILT_FACTOR_MINUS + 0x4000) >> 15;
         fact = 0x800; // 0.5 in (3.12)
         sh_fact = 12;
     }
     ga = (fact << 15) / av_clip_int16(32768 - FFABS(gt));
     gt >>= 1;
 
     /* Apply tilt compensation filter to signal. */
     tmp = res_pst[subframe_size - 1];
 
     for (i = subframe_size - 1; i >= 1; i--) {
         tmp2 = (res_pst[i] << 15) + ((gt * res_pst[i-1]) << 1);
         tmp2 = (tmp2 + 0x4000) >> 15;
 
         tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact;
         out[i] = tmp2;
     }
     tmp2 = (res_pst[0] << 15) + ((gt * ht_prev_data) << 1);
     tmp2 = (tmp2 + 0x4000) >> 15;
     tmp2 = (tmp2 * ga * 2 + fact) >> sh_fact;
     out[0] = tmp2;
 
     return tmp;
 }
 

 void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voicing,

                      const int16_t *lp_filter_coeffs, int pitch_delay_int,
                      int16_t* residual, int16_t* res_filter_data,
                      int16_t* pos_filter_data, int16_t *speech, int subframe_size)
 {

     int16_t residual_filt_buf[SUBFRAME_SIZE+11];

     int16_t lp_gn[33]; // (3.12)
     int16_t lp_gd[11]; // (3.12)
     int tilt_comp_coeff;
     int i;
 
     /* Zero-filling is necessary for tilt-compensation filter. */
     memset(lp_gn, 0, 33 * sizeof(int16_t));
 
     /* Calculate A(z/FORMANT_PP_FACTOR_NUM) filter coefficients. */
     for (i = 0; i < 10; i++)
         lp_gn[i + 11] = (lp_filter_coeffs[i + 1] * formant_pp_factor_num_pow[i] + 0x4000) >> 15;
 
     /* Calculate A(z/FORMANT_PP_FACTOR_DEN) filter coefficients. */
     for (i = 0; i < 10; i++)
         lp_gd[i + 1] = (lp_filter_coeffs[i + 1] * formant_pp_factor_den_pow[i] + 0x4000) >> 15;
 
     /* residual signal calculation (one-half of short-term postfilter) */
     memcpy(speech - 10, res_filter_data, 10 * sizeof(int16_t));
     residual_filter(residual + RES_PREV_DATA_SIZE, lp_gn + 11, speech, subframe_size);
     /* Save data to use it in the next subframe. */
     memcpy(res_filter_data, speech + subframe_size - 10, 10 * sizeof(int16_t));
 
     /* long-term filter. If long-term prediction gain is larger than 3dB (returned value is
        nonzero) then declare current subframe as periodic. */

     *voicing = FFMAX(*voicing, long_term_filter(adsp, pitch_delay_int,

                                                 residual, residual_filt_buf + 10,
                                                 subframe_size));
 
     /* shift residual for using in next subframe */
     memmove(residual, residual + subframe_size, RES_PREV_DATA_SIZE * sizeof(int16_t));
 
     /* short-term filter tilt compensation */

     tilt_comp_coeff = get_tilt_comp(adsp, lp_gn, lp_gd, residual_filt_buf + 10, subframe_size);

 
     /* Apply second half of short-term postfilter: 1/A(z/FORMANT_PP_FACTOR_DEN) */
     ff_celp_lp_synthesis_filter(pos_filter_data + 10, lp_gd + 1,
                                 residual_filt_buf + 10,

                                 subframe_size, 10, 0, 0, 0x800);

     memcpy(pos_filter_data, pos_filter_data + subframe_size, 10 * sizeof(int16_t));
 
     *ht_prev_data = apply_tilt_comp(speech, pos_filter_data + 10, tilt_comp_coeff,
                                     subframe_size, *ht_prev_data);
 }

 
 /**
  * \brief Adaptive gain control (4.2.4)
  * \param gain_before gain of speech before applying postfilters
  * \param gain_after  gain of speech after applying postfilters
  * \param speech [in/out] signal buffer
  * \param subframe_size length of subframe
  * \param gain_prev (3.12) previous value of gain coefficient
  *
  * \return (3.12) last value of gain coefficient
  */

 int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech,

                                    int subframe_size, int16_t gain_prev)
 {
     int gain; // (3.12)
     int n;
     int exp_before, exp_after;
 
     if(!gain_after && gain_before)
         return 0;
 
     if (gain_before) {
 
         exp_before  = 14 - av_log2(gain_before);
         gain_before = bidir_sal(gain_before, exp_before);
 
         exp_after  = 14 - av_log2(gain_after);
         gain_after = bidir_sal(gain_after, exp_after);
 
         if (gain_before < gain_after) {
             gain = (gain_before << 15) / gain_after;
             gain = bidir_sal(gain, exp_after - exp_before - 1);
         } else {
             gain = ((gain_before - gain_after) << 14) / gain_after + 0x4000;
             gain = bidir_sal(gain, exp_after - exp_before);
         }
         gain = (gain * G729_AGC_FAC1 + 0x4000) >> 15; // gain * (1-0.9875)
     } else
         gain = 0;
 
     for (n = 0; n < subframe_size; n++) {
         // gain_prev = gain + 0.9875 * gain_prev
         gain_prev = (G729_AGC_FACTOR * gain_prev + 0x4000) >> 15;
         gain_prev = av_clip_int16(gain + gain_prev);
         speech[n] = av_clip_int16((speech[n] * gain_prev + 0x2000) >> 14);
     }
     return gain_prev;
 }