cod_ld8a.c

具有浮点指令的g.729语音压缩编码

 *       * quantize and code the LSPs                                     *
 *       * find the interpolated LSPs and convert to a[] for the 2        *
 *         subframes (both quantized and unquantized)                     *
 *------------------------------------------------------------------------*/
   {
      /* Temporary vectors */
     FLOAT r[MP1];                    /* Autocorrelations       */
     FLOAT rc[M];                     /* Reflexion coefficients */
     FLOAT lsp_new[M];                /* lsp coefficients       */
     FLOAT lsp_new_q[M];              /* Quantized lsp coeff.   */

     /* LP analysis */

     autocorr(p_window, M, r);             /* Autocorrelations */
     lag_window(M, r);                     /* Lag windowing    */
     levinson(r, Ap_t, rc);                /* Levinson Durbin  */
     az_lsp(Ap_t, lsp_new, lsp_old);       /* Convert A(z) to lsp */

     /* LSP quantization */

     qua_lsp(lsp_new, lsp_new_q, ana);
     ana += 2;                        /* Advance analysis parameters pointer */

    /*--------------------------------------------------------------------*
     * Find interpolated LPC parameters in all subframes                  *
     * The interpolated parameters are in array Aq_t[].                   *
     *--------------------------------------------------------------------*/

    int_qlpc(lsp_old_q, lsp_new_q, Aq_t);

    /* Compute A(z/gamma) */

    weight_az(&Aq_t[0],   GAMMA1, M, &Ap_t[0]);
    weight_az(&Aq_t[MP1], GAMMA1, M, &Ap_t[MP1]);

    /* update the LSPs for the next frame */

    copy(lsp_new,   lsp_old,   M);
    copy(lsp_new_q, lsp_old_q, M);
  }

   /*----------------------------------------------------------------------*
    * - Find the weighted input speech w_sp[] for the whole speech frame   *
    * - Find the open-loop pitch delay for the whole speech frame          *
    * - Set the range for searching closed-loop pitch in 1st subframe      *
    *----------------------------------------------------------------------*/

   residu(&Aq_t[0],   &speech[0],       &exc[0],       L_SUBFR);
   residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);

  {
     FLOAT Ap1[MP1];

     Ap = Ap_t;
     Ap1[0] = (F)1.0;
     for(i=1; i<=M; i++)
       Ap1[i] = Ap[i] - (F)0.7 * Ap[i-1];
     syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);

     Ap += MP1;
     for(i=1; i<=M; i++)
       Ap1[i] = Ap[i] - (F)0.7 * Ap[i-1];
     syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
   }


   /* Find open loop pitch lag for whole speech frame */

   T_op = pitch_ol_fast(wsp, L_FRAME);

   /* Range for closed loop pitch search in 1st subframe */

   T0_min = T_op - 3;
   if (T0_min < PIT_MIN) T0_min = PIT_MIN;
   T0_max = T0_min + 6;
   if (T0_max > PIT_MAX)
   {
      T0_max = PIT_MAX;
      T0_min = T0_max - 6;
   }

   /*------------------------------------------------------------------------*
    *          Loop for every subframe in the analysis frame                 *
    *------------------------------------------------------------------------*
    *  To find the pitch and innovation parameters. The subframe size is     *
    *  L_SUBFR and the loop is repeated L_FRAME/L_SUBFR times.               *
    *     - find the weighted LPC coefficients                               *
    *     - find the LPC residual signal                                     *
    *     - compute the target signal for pitch search                       *
    *     - compute impulse response of weighted synthesis filter (h1[])     *
    *     - find the closed-loop pitch parameters                            *
    *     - encode the pitch delay                                           *
    *     - find target vector for codebook search                           *
    *     - codebook search                                                  *
    *     - VQ of pitch and codebook gains                                   *
    *     - update states of weighting filter                                *
    *------------------------------------------------------------------------*/

   Aq = Aq_t;    /* pointer to interpolated quantized LPC parameters */
   Ap = Ap_t;    /* pointer to weighted LPC coefficients             */

   for (i_subfr = 0;  i_subfr < L_FRAME; i_subfr += L_SUBFR)
   {
      /*---------------------------------------------------------------*
       * Compute impulse response, h1[], of weighted synthesis filter  *
       *---------------------------------------------------------------*/

      h1[0] = (F)1.0;
      set_zero(&h1[1], L_SUBFR-1);
      syn_filt(Ap, h1, h1, L_SUBFR, &h1[1], 0);

      /*-----------------------------------------------*
       * Find the target vector for pitch search:      *
       *----------------------------------------------*/

      syn_filt(Ap, &exc[i_subfr], xn, L_SUBFR, mem_w0, 0);

      /*-----------------------------------------------------------------*
       *    Closed-loop fractional pitch search                          *
       *-----------------------------------------------------------------*/

      T0 = pitch_fr3_fast(&exc[i_subfr], xn, h1, L_SUBFR, T0_min, T0_max,
                    i_subfr, &T0_frac);

      index = enc_lag3(T0, T0_frac, &T0_min, &T0_max, PIT_MIN, PIT_MAX,
                            i_subfr);

      *ana++ = index;

      if (i_subfr == 0)
        *ana++ = parity_pitch(index);

      /*-----------------------------------------------------------------*
       *   - find filtered pitch exc                                     *
       *   - compute pitch gain and limit between 0 and 1.2              *
       *   - update target vector for codebook search                    *
       *   - find LTP residual.                                          *
       *-----------------------------------------------------------------*/

      syn_filt(Ap, &exc[i_subfr], y1, L_SUBFR, mem_zero, 0);

      gain_pit = g_pitch(xn, y1, g_coeff, L_SUBFR);

      /* clip pitch gain if taming is necessary */

      taming = test_err(T0, T0_frac);

      if( taming == 1){
        if (gain_pit > GPCLIP) {
          gain_pit = GPCLIP;
      }
    }

    for (i = 0; i < L_SUBFR; i++)
        xn2[i] = xn[i] - y1[i]*gain_pit;

      /*-----------------------------------------------------*
       * - Innovative codebook search.                       *
       *-----------------------------------------------------*/

      index = ACELP_code_A(xn2, h1, T0, sharp, code, y2, &i);

      *ana++ = index;           /* Positions index */
      *ana++ = i;               /* Signs index     */

      /*------------------------------------------------------*
       *  - Compute the correlations <y2,y2>, <xn,y2>, <y1,y2>*
       *  - Vector quantize gains.                            *
       *------------------------------------------------------*/

      corr_xy2(xn, y1, y2, g_coeff);
       *ana++ =qua_gain(code, g_coeff, L_SUBFR, &gain_pit, &gain_code,
                                    taming);

      /*------------------------------------------------------------*
       * - Update pitch sharpening "sharp" with quantized gain_pit  *
       *------------------------------------------------------------*/

      sharp = gain_pit;
      if (sharp > SHARPMAX) sharp = SHARPMAX;
      if (sharp < SHARPMIN) sharp = SHARPMIN;

      /*------------------------------------------------------*
       * - Find the total excitation                          *
       * - update filters' memories for finding the target    *
       *   vector in the next subframe  (mem_w0[])            *
       *------------------------------------------------------*/

      for (i = 0; i < L_SUBFR;  i++)
        exc[i+i_subfr] = gain_pit*exc[i+i_subfr] + gain_code*code[i];

      update_exc_err(gain_pit, T0);

      for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
        mem_w0[j]  = xn[i] - gain_pit*y1[i] - gain_code*y2[i];

      Aq += MP1;           /* interpolated LPC parameters for next subframe */
      Ap += MP1;
   }

   /*--------------------------------------------------*
    * Update signal for next frame.                    *
    * -> shift to the left by L_FRAME:                 *
    *     speech[], wsp[] and  exc[]                   *
    *--------------------------------------------------*/

   copy(&old_speech[L_FRAME], &old_speech[0], L_TOTAL-L_FRAME);
   copy(&old_wsp[L_FRAME], &old_wsp[0], PIT_MAX);
   copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);

   return;
}