语音压缩

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COD_LD8A.C：源码内容
							/*
   ITU-T G.729A Speech Coder    ANSI-C Source Code
   Version 1.1    Last modified: September 1996
   Copyright (c) 1996,
   AT&T, France Telecom, NTT, Universite de Sherbrooke
   All rights reserved.
*/
/*-----------------------------------------------------------------*
 *   Functions Coder_ld8a and Init_Coder_ld8a                      *
 *             ~~~~~~~~~~     ~~~~~~~~~~~~~~~                      *
 *                                                                 *      
 *  Init_Coder_ld8a(void);                                         *
 *                                                                 *
 *   ->Initialization of variables for the coder section.          *
 *                                                                 *
 *                                                                 *
 *  Coder_ld8a(Word16 ana[]);                                      *
 *                                                                 *
 *   ->Main coder function.                                        *
 *                                                                 *
 *                                                                 *
 *  Input:                                                         *
 *                                                                 *
 *    80 speech data should have beee copy to vector new_speech[]. *
 *    This vector is global and is declared in this function.      *
 *                                                                 *
 *  Ouputs:                                                        *
 *                                                                 *
 *    ana[]      ->analysis parameters.                            *
 *                                                                 *
 *-----------------------------------------------------------------*/
#include <stdio.h>
#include <stdlib.h>
#include "typedef.h"
#include "basic_op.h"
#include "ld8a.h"
/*-----------------------------------------------------------*
 *    Coder constant parameters (defined in "ld8a.h")        *
 *-----------------------------------------------------------*
 *   L_WINDOW    : LPC analysis window size.                 *
 *   L_NEXT      : Samples of next frame needed for autocor. *
 *   L_FRAME     : Frame size.                               *
 *   L_SUBFR     : Sub-frame size.                           *
 *   M           : LPC order.                                *
 *   MP1         : LPC order+1                               *
 *   L_TOTAL     : Total size of speech buffer.              *
 *   PIT_MIN     : Minimum pitch lag.                        *
 *   PIT_MAX     : Maximum pitch lag.                        *
 *   L_INTERPOL  : Length of filter for interpolation        *
 *-----------------------------------------------------------*/
/*--------------------------------------------------------*
 *         Static memory allocation.                      *
 *--------------------------------------------------------*/
        /* Speech vector */
 static Word16 old_speech[L_TOTAL];
 static Word16 *speech, *p_window;
 Word16 *new_speech;                    /* Global variable */
        /* Weighted speech vector */
 static Word16 old_wsp[L_FRAME+PIT_MAX];
 static Word16 *wsp;
        /* Excitation vector */
 static Word16 old_exc[L_FRAME+PIT_MAX+L_INTERPOL];
 static Word16 *exc;
        /* Lsp (Line spectral pairs) */
 static Word16 lsp_old[M]={
              30000, 26000, 21000, 15000, 8000, 0, -8000,-15000,-21000,-26000};
 static Word16 lsp_old_q[M];
        /* Filter's memory */
 static Word16  mem_w0[M], mem_w[M], mem_zero[M];
 static Word16  sharp;
/*-----------------------------------------------------------------*
 *   Function  Init_Coder_ld8a                                     *
 *            ~~~~~~~~~~~~~~~                                      *
 *                                                                 *
 *  Init_Coder_ld8a(void);                                         *
 *                                                                 *
 *   ->Initialization of variables for the coder section.          *
 *       - initialize pointers to speech buffer                    *
 *       - initialize static  pointers                             *
 *       - set static vectors to zero                              *
 *                                                                 *
 *-----------------------------------------------------------------*/
void Init_Coder_ld8a(void)
{
  /*----------------------------------------------------------------------*
  *      Initialize pointers to speech vector.                            *
  *                                                                       *
  *                                                                       *
  *   |--------------------|-------------|-------------|------------|     *
  *     previous speech           sf1           sf2         L_NEXT        *
  *                                                                       *
  *   <----------------  Total speech vector (L_TOTAL)   ----------->     *
  *   <----------------  LPC analysis window (L_WINDOW)  ----------->     *
  *   |                   <-- present frame (L_FRAME) -->                 *
  * old_speech            |              <-- new speech (L_FRAME) -->     *
  * p_window              |              |                                *
  *                     speech           |                                *
  *                             new_speech                                *
  *-----------------------------------------------------------------------*/
  new_speech = old_speech + L_TOTAL - L_FRAME;         /* New speech     */
  speech     = new_speech - L_NEXT;                    /* Present frame  */
  p_window   = old_speech + L_TOTAL - L_WINDOW;        /* For LPC window */
  /* Initialize static pointers */
  wsp    = old_wsp + PIT_MAX;
  exc    = old_exc + PIT_MAX + L_INTERPOL;
  /* Static vectors to zero */
  Set_zero(old_speech, L_TOTAL);
  Set_zero(old_exc, PIT_MAX+L_INTERPOL);
  Set_zero(old_wsp, PIT_MAX);
  Set_zero(mem_w,   M);
  Set_zero(mem_w0,  M);
  Set_zero(mem_zero, M);
  sharp = SHARPMIN;
  /* Initialize lsp_old_q[] */
  Copy(lsp_old, lsp_old_q, M);
  Lsp_encw_reset();
  Init_exc_err();
 return;
}
/*-----------------------------------------------------------------*
 *   Functions Coder_ld8a                                          *
 *            ~~~~~~~~~~                                           *
 *  Coder_ld8a(Word16 ana[]);                                      *
 *                                                                 *
 *   ->Main coder function.                                        *
 *                                                                 *
 *                                                                 *
 *  Input:                                                         *
 *                                                                 *
 *    80 speech data should have beee copy to vector new_speech[]. *
 *    This vector is global and is declared in this function.      *
 *                                                                 *
 *  Ouputs:                                                        *
 *                                                                 *
 *    ana[]      ->analysis parameters.                            *
 *                                                                 *
 *-----------------------------------------------------------------*/
void Coder_ld8a(
     Word16 ana[]       /* output  : Analysis parameters */
)
{
  /* LPC analysis */
  Word16 Aq_t[(MP1)*2];         /* A(z)   quantized for the 2 subframes */
  Word16 Ap_t[(MP1)*2];         /* A(z/gamma)       for the 2 subframes */
  Word16 *Aq, *Ap;              /* Pointer on Aq_t and Ap_t             */
  /* Other vectors */
  Word16 h1[L_SUBFR];            /* Impulse response h1[]              */
  Word16 xn[L_SUBFR];            /* Target vector for pitch search     */
  Word16 xn2[L_SUBFR];           /* Target vector for codebook search  */
  Word16 code[L_SUBFR];          /* Fixed codebook excitation          */
  Word16 y1[L_SUBFR];            /* Filtered adaptive excitation       */
  Word16 y2[L_SUBFR];            /* Filtered fixed codebook excitation */
  Word16 g_coeff[4];             /* Correlations between xn & y1       */
  Word16 g_coeff_cs[5];
  Word16 exp_g_coeff_cs[5];      /* Correlations between xn, y1, & y2
                                     <y1,y1>, -2<xn,y1>,
                                          <y2,y2>, -2<xn,y2>, 2<y1,y2> */
  /* Scalars */
  Word16 i, j, k, i_subfr;
  Word16 T_op, T0, T0_min, T0_max, T0_frac;
  Word16 gain_pit, gain_code, index;
  Word16 temp, taming;
  Word32 L_temp;
/*------------------------------------------------------------------------*
 *  - Perform LPC analysis:                                               *
 *       * autocorrelation + lag windowing                                *
 *       * Levinson-durbin algorithm to find a[]                          *
 *       * convert a[] to lsp[]                                           *
 *       * quantize and code the LSPs                                     *
 *       * find the interpolated LSPs and convert to a[] for the 2        *
 *         subframes (both quantized and unquantized)                     *
 *------------------------------------------------------------------------*/
  {
     /* Temporary vectors */
    Word16 r_l[MP1], r_h[MP1];       /* Autocorrelations low and hi          */
    Word16 rc[M];                    /* Reflection coefficients.             */
    Word16 lsp_new[M], lsp_new_q[M]; /* LSPs at 2th subframe                 */
    /* LP analysis */
    Autocorr(p_window, M, r_h, r_l);              /* Autocorrelations */
    Lag_window(M, r_h, r_l);                      /* Lag windowing    */
    Levinson(r_h, r_l, Ap_t, rc);                 /* Levinson Durbin  */
    Az_lsp(Ap_t, lsp_new, lsp_old);               /* From 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                                     *
  *----------------------------------------------------------------------*/
  Residu(&Aq_t[0], &speech[0], &exc[0], L_SUBFR);
  Residu(&Aq_t[MP1], &speech[L_SUBFR], &exc[L_SUBFR], L_SUBFR);
  {
    Word16 Ap1[MP1];
    Ap = Ap_t;
    Ap1[0] = 4096;
    for(i=1; i<=M; i++)    /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
       Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
    Syn_filt(Ap1, &exc[0], &wsp[0], L_SUBFR, mem_w, 1);
    Ap += MP1;
    for(i=1; i<=M; i++)    /* Ap1[i] = Ap[i] - 0.7 * Ap[i-1]; */
       Ap1[i] = sub(Ap[i], mult(Ap[i-1], 22938));
    Syn_filt(Ap1, &exc[L_SUBFR], &wsp[L_SUBFR], L_SUBFR, mem_w, 1);
  }
  /* Find open loop pitch lag */
  T_op = Pitch_ol_fast(wsp, PIT_MAX, L_FRAME);
  /* Range for closed loop pitch search in 1st subframe */
  T0_min = sub(T_op, 3);
  if (sub(T0_min,PIT_MIN)<0) {
    T0_min = PIT_MIN;
  }
  T0_max = add(T0_min, 6);
  if (sub(T0_max ,PIT_MAX)>0)
  {
     T0_max = PIT_MAX;
     T0_min = sub(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 2 times.                             *
  *     - find the weighted LPC coefficients                               *
  *     - find the LPC residual signal res[]                               *
  *     - 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] = 4096;
    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                    *
    *-----------------------------------------------------------------*/
    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 (sub(gain_pit, GPCLIP) > 0) {
        gain_pit = GPCLIP;
      }
    }
    /* xn2[i]   = xn[i] - y1[i] * gain_pit  */
    for (i = 0; i < L_SUBFR; i++)
    {
      L_temp = L_mult(y1[i], gain_pit);
      L_temp = L_shl(L_temp, 1);               /* gain_pit in Q14 */
      xn2[i] = sub(xn[i], extract_h(L_temp));
    }
   /*-----------------------------------------------------*
    * - Innovative codebook search.                       *
    *-----------------------------------------------------*/
    index = ACELP_Code_A(xn2, h1, T0, sharp, code, y2, &i);
    *ana++ = index;        /* Positions index */
    *ana++ = i;            /* Signs index     */
   /*-----------------------------------------------------*
    * - Quantization of gains.                            *
    *-----------------------------------------------------*/
    g_coeff_cs[0]     = g_coeff[0];            /* <y1,y1> */
    exp_g_coeff_cs[0] = negate(g_coeff[1]);    /* Q-Format:XXX -> JPN */
    g_coeff_cs[1]     = negate(g_coeff[2]);    /* (xn,y1) -> -2<xn,y1> */
    exp_g_coeff_cs[1] = negate(add(g_coeff[3], 1)); /* Q-Format:XXX -> JPN */
    Corr_xy2( xn, y1, y2, g_coeff_cs, exp_g_coeff_cs );  /* Q0 Q0 Q12 ^Qx ^Q0 */
                         /* g_coeff_cs[3]:exp_g_coeff_cs[3] = <y2,y2>   */
                         /* g_coeff_cs[4]:exp_g_coeff_cs[4] = -2<xn,y2> */
                         /* g_coeff_cs[5]:exp_g_coeff_cs[5] = 2<y1,y2>  */
    *ana++ = Qua_gain(code, g_coeff_cs, exp_g_coeff_cs,
                         L_SUBFR, &gain_pit, &gain_code, taming);
   /*------------------------------------------------------------*
    * - Update pitch sharpening "sharp" with quantized gain_pit  *
    *------------------------------------------------------------*/
    sharp = gain_pit;
    if (sub(sharp, SHARPMAX) > 0) { sharp = SHARPMAX;         }
    if (sub(sharp, SHARPMIN) < 0) { sharp = SHARPMIN;         }
   /*------------------------------------------------------*
    * - Find the total excitation                          *
    * - update filters memories for finding the target     *
    *   vector in the next subframe                        *
    *------------------------------------------------------*/
    for (i = 0; i < L_SUBFR;  i++)
    {
      /* exc[i] = gain_pit*exc[i] + gain_code*code[i]; */
      /* exc[i]  in Q0   gain_pit in Q14               */
      /* code[i] in Q13  gain_cod in Q1                */
      L_temp = L_mult(exc[i+i_subfr], gain_pit);
      L_temp = L_mac(L_temp, code[i], gain_code);
      L_temp = L_shl(L_temp, 1);
      exc[i+i_subfr] = round(L_temp);
    }
    update_exc_err(gain_pit, T0);
    for (i = L_SUBFR-M, j = 0; i < L_SUBFR; i++, j++)
    {
      temp       = extract_h(L_shl( L_mult(y1[i], gain_pit),  1) );
      k          = extract_h(L_shl( L_mult(y2[i], gain_code), 2) );
      mem_w0[j]  = sub(xn[i], add(temp, k));
    }
    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;
}

						