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📄 enc_gain.c

📁 关于AMR-WB+语音压缩编码的实现代码
💻 C
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   for (i = 4; i > 0; i--)
   {    
      old_ol_lag[i] = old_ol_lag[i-1];
   }    
   old_ol_lag[0] = prev_ol_lag;
   for (i = 0; i < 5; i++)
   {
      tmp[i+1] = old_ol_lag[i];
   }
   E_GAIN_sort(5, tmp);
   return tmp[3];
}
/*
 * E_GAIN_norm_corr
 *
 * Parameters:
 *    exc            I: excitation buffer
 *    xn             I: target signal
 *    h              I: weighted synthesis filter impulse response (Q15)
 *    t0_min         I: minimum value in the searched range
 *    t0_max         I: maximum value in the searched range
 *    corr_norm      O: normalized correlation (Q15)
 *
 * Function:
 *    Find the normalized correlation between the target vector and the
 *    filtered past excitation (correlation between target and filtered
 *    excitation divided by the square root of energy of filtered excitation)
 *    Size of subframe = L_SUBFR.
 *
 * Returns:
 *    void
 */
static void E_GAIN_norm_corr(Float32 exc[], Float32 xn[], Float32 h[],
                             Word32 t_min, Word32 t_max, Float32 corr_norm[])
{
   Float32 excf[L_SUBFR];  /* filtered past excitation */
   Float32 alp, ps, norm;
   Word32 t, j, k;
   k = - t_min;
   /* compute the filtered excitation for the first delay t_min */
   E_UTIL_f_convolve(&exc[k], h, excf);
   /* loop for every possible period */
   for (t = t_min; t <= t_max; t++)
   {
      /* Compute correlation between xn[] and excf[] */
      ps = 0.0F;
      alp = 0.01F;
      for (j = 0; j < L_SUBFR; j++)
      {
         ps += xn[j] * excf[j];
         alp += excf[j] * excf[j];
      }
      /* Compute 1/sqrt(energie of excf[]) */
      norm = (Float32)(1.0F / sqrt(alp));
      /* Normalize correlation = correlation * (1/sqrt(energy)) */
      corr_norm[t] = ps * norm;
      /* update the filtered excitation excf[] for the next iteration */
      if (t != t_max)
      {
         k--;
         for (j = L_SUBFR - 1; j > 0; j--)
         {
            excf[j] = excf[j - 1] + exc[k] * h[j];
         }
         excf[0] = exc[k];
      }
   }
   return;
}
/*
 * E_GAIN_norm_corr_interpolate
 *
 * Parameters:
 *    x           I: input vector
 *    frac        I: fraction (-4..+3)
 *
 * Function:
 *    Interpolating the normalized correlation
 *
 * Returns:
 *    interpolated value
 */
static Float32 E_GAIN_norm_corr_interpolate(Float32 *x, Word32 frac)
{
   Float32 s, *x1, *x2;
   const Float32 *c1, *c2;
   if (frac < 0)
   {
      frac += 4;
      x--;                                          
   }
   x1 = &x[0];
   x2 = &x[1];
   c1 = &E_ROM_inter4_1[frac];
   c2 = &E_ROM_inter4_1[4 - frac];
   s = x1[0] * c1[0] + x2[0] * c2[0];
   s += x1[-1] * c1[4] + x2[1] * c2[4];
   s += x1[-2] * c1[8] + x2[2] * c2[8];
   s += x1[-3] * c1[12] + x2[3] * c2[12];
   return s;
}
/*
 * E_GAIN_closed_loop_search
 *
 * Parameters:
 *    exc            I: excitation buffer
 *    xn             I: target signal
 *    h              I: weighted synthesis filter impulse response
 *    t0_min         I: minimum value in the searched range
 *    t0_max         I: maximum value in the searched range
 *    pit_frac       O: chosen fraction
 *    i_subfr        I: flag to first subframe
 *    t0_fr2         I: minimum value for resolution 1/2
 *    t0_fr1         I: minimum value for resolution 1
 *
 * Function:
 *    Find the closed loop pitch period with 1/4 subsample resolution.
 *
 * Returns:
 *    chosen integer pitch lag
 */
Word32 E_GAIN_closed_loop_search(Float32 exc[], Float32 xn[], Float32 h[],
                             Word32 t0_min, Word32 t0_max, Word32 *pit_frac,
                             Word32 i_subfr, Word32 t0_fr2, Word32 t0_fr1)
{
   Float32 corr_v[15 + 2 * L_INTERPOL1 + 1];
   Float32 cor_max, max, temp;
   Float32 *corr;
   Word32 i, fraction, step;
   Word32 t0, t_min, t_max;
   /* Find interval to compute normalized correlation */
   t_min = t0_min - L_INTERPOL1;
   t_max = t0_max + L_INTERPOL1;
   /* allocate memory to normalized correlation vector */
   corr = &corr_v[-t_min];      /* corr[t_min..t_max] */
   /* Compute normalized correlation between target and filtered excitation */
   E_GAIN_norm_corr(exc, xn, h, t_min, t_max, corr);
   /*  find integer pitch */
   max = corr[t0_min];
   t0  = t0_min;
   for(i = t0_min + 1; i <= t0_max; i++)
   {
      if( corr[i] > max)
      {
         max = corr[i];
         t0 = i;
      }
   }
   /* If first subframe and t0 >= t0_fr1, do not search fractionnal pitch */
   if((i_subfr == 0) & (t0 >= t0_fr1))
   {
      *pit_frac = 0;
      return(t0);
   }
   /*
    * Search fractionnal pitch with 1/4 subsample resolution.
    * Test the fractions around t0 and choose the one which maximizes
    * the interpolated normalized correlation.
    */
   step = 1;                /* 1/4 subsample resolution */
   fraction = -3;
   if (((i_subfr == 0) & (t0 >= t0_fr2)) | (t0_fr2 == PIT_MIN))
   {
      step = 2;              /* 1/2 subsample resolution */
      fraction = -2;
   }
   if (t0 == t0_min)
   {
      fraction = 0;
   }
   cor_max = E_GAIN_norm_corr_interpolate(&corr[t0], fraction);
   for (i = (fraction + step); i <= 3; i += step)
   {
;
      temp = E_GAIN_norm_corr_interpolate(&corr[t0], i);                
      if (temp > cor_max)
      {
         cor_max = temp;
         fraction = i;
      }
   }
   /* limit the fraction value in the interval [0,1,2,3] */
   if (fraction < 0)
   {
      fraction += 4;
      t0 -= 1;
   }
   *pit_frac = fraction;
   return (t0);
}
/*
 * E_GAIN_adaptive_codebook_excitation
 *
 * Parameters:
 *    exc          I/O: excitation buffer
 *    T0             I: integer pitch lag
 *    frac           I: fraction of lag
 *    L_subfr        I: subframe size
 *
 * Function:
 *    Compute the result of Word32 term prediction with fractional
 *    interpolation of resolution 1/4.
 *
 * Returns:
 *    interpolated signal (adaptive codebook excitation)
 */
void E_GAIN_adaptive_codebook_excitation(Word16 exc[], Word16 T0, Word32 frac, Word16 L_subfr)
{
   Word32 i, j, k, L_sum;
   Word16 *x;
   x = &exc[-T0];
   frac = -(frac);
   if (frac < 0)
   {
      frac = (frac + UP_SAMP);
      x--;                                                      
   }
   x = x - L_INTERPOL2 + 1;
   for (j = 0; j < L_subfr; j++)
   {
      L_sum = 0L;
      for (i = 0, k = ((UP_SAMP - 1) - frac); i < 2 * L_INTERPOL2; i++, k += UP_SAMP)
      {
         L_sum = L_sum + (x[i] * E_ROM_inter4_2[k]);
      }
      L_sum = (L_sum + 0x2000) >> 14;
      exc[j] = E_UTIL_saturate(L_sum);
      x++;
   }
   return;
}
/*
 * E_GAIN_pitch_sharpening
 *
 * Parameters:
 *    x            I/O: impulse response (or algebraic code)
 *    pit_lag        I: pitch lag
 *
 * Function:
 *    Performs Pitch sharpening routine for one subframe.
 *    pitch sharpening factor is 0.85
 *
 * Returns:
 *    void
 */
void E_GAIN_pitch_sharpening(Word16 *x, Word16 pit_lag)
{
   Word32 L_tmp, i;
   for (i = pit_lag; i < L_SUBFR; i++)
   {
      L_tmp = x[i] << 15;
      L_tmp += x[i - pit_lag] * PIT_SHARP;
      x[i] = (Word16)((L_tmp + 0x4000) >> 15);
   }
   return;
}
void E_GAIN_f_pitch_sharpening(Float32 *x, Word32 pit_lag)
{
   Word32 i;
   for (i = pit_lag; i < L_SUBFR; i++)
   {
      x[i] += x[i - pit_lag] * F_PIT_SHARP;
   }
   return;
}
/*
 * E_GAIN_voice_factor
 *
 * Parameters:
 *    exc            I: pitch excitation (Q_exc)
 *    Q_exc          I: exc format
 *    gain_pit       I: gain of pitch (Q14)
 *    code           I: Fixed codebook excitation (Q9)
 *    gain_code      I: gain of code (Q0)
 *
 *
 * Function:
 *    Find the voicing factor (1=voice to -1=unvoiced)
 *    Subframe length is L_SUBFR
 *
 * Returns:
 *    factor (-1=unvoiced to 1=voiced) (Q15)
 */
Word32 E_GAIN_voice_factor(Word16 exc[], Word16 Q_exc, Word16 gain_pit,
                          Word16 code[], Word16 gain_code)
{
   Word32 i, L_tmp, tmp, exp, ener1, exp1, ener2, exp2;
   ener1 = E_UTIL_dot_product12(exc, exc, L_SUBFR, &exp1) >> 16;
   exp1 = exp1 - (Q_exc + Q_exc);
   L_tmp = (gain_pit * gain_pit) << 1;
   exp = E_UTIL_norm_l(L_tmp);
   tmp = (L_tmp << exp) >> 16;
   ener1 = (ener1 * tmp) >> 15;
   exp1 = (exp1 - exp) - 10;        /* 10 -> gain_pit Q14 to Q9 */
   ener2 = E_UTIL_dot_product12(code, code, L_SUBFR, &exp2) >> 16;
   exp = E_UTIL_norm_s(gain_code);
   tmp = gain_code << exp;
   tmp = (tmp * tmp) >> 15;
   ener2 = (ener2 * tmp) >> 15;
   exp2 = exp2 - (exp + exp);
   i = exp1 - exp2;
   if (i >= 0)
   {
      ener1 = ener1 >> 1;
      ener2 = ener2 >> (i + 1);
   }
   else
   {
      i = 1 - i;
      if (i < 32)
      {
         ener1 = ener1 >> i;
      }
      else
      {
         ener1 = 0;
      }
      ener2 = ener2 >> 1;
   }
   tmp = ener1 - ener2;
   ener1 = (ener1 + ener2) + 1;
   tmp = (tmp << 15) / ener1;
   return (tmp);
}

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