dec_ld8a.c

具有浮点指令的g.729语音压缩编码

/* ITU-T G.729 Software Package Release 2 (November 2006) */
/*
   ITU-T G.729 Annex C - Reference C code for floating point
                         implementation of G.729 Annex A
                         Version 1.01 of 15.September.98
*/

/*
----------------------------------------------------------------------
                    COPYRIGHT NOTICE
----------------------------------------------------------------------
   ITU-T G.729 Annex C ANSI C source code
   Copyright (C) 1998, AT&T, France Telecom, NTT, University of
   Sherbrooke.  All rights reserved.

----------------------------------------------------------------------
*/

/*-----------------------------------------------------------------*
 *   Functions init_decod_ld8a  and decod_ld8a                     *
 *-----------------------------------------------------------------*/

#include "typedef.h"
#include "ld8a.h"

/*---------------------------------------------------------------*
 *   Decoder constant parameters (defined in "ld8a.h")           *
 *---------------------------------------------------------------*
 *   L_FRAME     : Frame size.                                   *
 *   L_SUBFR     : Sub-frame size.                               *
 *   M           : LPC order.                                    *
 *   MP1         : LPC order+1                                   *
 *   PIT_MIN     : Minimum pitch lag.                            *
 *   PIT_MAX     : Maximum pitch lag.                            *
 *   L_INTERPOL  : Length of filter for interpolation            *
 *   PRM_SIZE    : Size of vector containing analysis parameters *
 *---------------------------------------------------------------*/

/*--------------------------------------------------------*
 *         Static memory allocation.                      *
 *--------------------------------------------------------*/

        /* Excitation vector */

 static FLOAT old_exc[L_FRAME+PIT_MAX+L_INTERPOL];
 static FLOAT *exc;

        /* Lsp (Line spectral pairs) */

 static FLOAT lsp_old[M]={
       (F)0.9595,  (F)0.8413,  (F)0.6549,  (F)0.4154,  (F)0.1423,
      (F)-0.1423, (F)-0.4154, (F)-0.6549, (F)-0.8413, (F)-0.9595};

 static FLOAT mem_syn[M];       /* Synthesis filter's memory          */
 static FLOAT sharp;            /* pitch sharpening of previous frame */
 static int old_T0;             /* integer delay of previous frame    */
 static FLOAT gain_code;        /* Code gain                          */
 static FLOAT gain_pitch;       /* Pitch gain                         */


/*-----------------------------------------------------------------*
 *   Function init_decod_ld8a                                      *
 *            ~~~~~~~~~~~~~~~                                      *
 *                                                                 *
 *   ->Initialization of variables for the decoder section.        *
 *                                                                 *
 *-----------------------------------------------------------------*/

void init_decod_ld8a(void)
{

  /* Initialize static pointer */

  exc = old_exc + PIT_MAX + L_INTERPOL;

  /* Static vectors to zero */

  set_zero(old_exc, PIT_MAX+L_INTERPOL);
  set_zero(mem_syn, M);

  sharp  = SHARPMIN;
  old_T0 = 60;
  gain_code = (F)0.0;
  gain_pitch = (F)0.0;

  lsp_decw_reset() ;

  return;
}

/*-----------------------------------------------------------------*
 *   Function decod_ld8a                                           *
 *           ~~~~~~~~~~                                            *
 *   ->Main decoder routine.                                       *
 *                                                                 *
 *-----------------------------------------------------------------*/

void decod_ld8a(
  int      parm[],      /* (i)   : vector of synthesis parameters
                                  parm[0] = bad frame indicator (bfi)  */
  FLOAT   synth[],     /* (o)   : synthesis speech                     */
  FLOAT   A_t[],       /* (o)   : decoded LP filter in 2 subframes     */
  int     *T2          /* (o)   : decoded pitch lag in 2 subframes     */
)
{
  FLOAT   *Az;                  /* Pointer on A_t   */
  FLOAT   lsp_new[M];           /* Decoded LSP's    */
  FLOAT   code[L_SUBFR];        /* ACELP codevector */

   /* Scalars */

   int i, i_subfr;
   int T0, T0_frac, index;
   int  bfi, bad_pitch;
   extern int bad_lsf;        /* bad LSF indicator   */

   /* Test bad frame indicator (bfi) */

   bfi = *parm++;

   /* Decode the LSPs */

   d_lsp(parm, lsp_new, bfi+bad_lsf );
   parm += 2;                        /* Advance synthesis parameters pointer */

  /*
  Note: "bad_lsf" is introduce in case the standard is used with
         channel protection.
  */

   /* Interpolation of LPC for the 2 subframes */

   int_qlpc(lsp_old, lsp_new, A_t);

   /* update the LSFs for the next frame */

   copy(lsp_new, lsp_old, M);

   /*------------------------------------------------------------------------*
    *          Loop for every subframe in the analysis frame                 *
    *------------------------------------------------------------------------*
    * The subframe size is L_SUBFR and the loop is repeated L_FRAME/L_SUBFR  *
    *  times                                                                 *
    *     - decode the pitch delay                                           *
    *     - decode algebraic code                                            *
    *     - decode pitch and codebook gains                                  *
    *     - find the excitation and compute synthesis speech                 *
    *------------------------------------------------------------------------*/

   Az = A_t;                     /* pointer to interpolated LPC parameters */

   for (i_subfr = 0; i_subfr < L_FRAME; i_subfr += L_SUBFR) {

      /*-------------------------------------------------*
       *  - Find the adaptive codebook vector.            *
       *--------------------------------------------------*/

      index = *parm++;            /* pitch index */

      if(i_subfr == 0)
      {

        i = *parm++; /* get parity check result */
        bad_pitch = bfi + i;
        if( bad_pitch == 0)
        {
            dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
            old_T0 = T0;
        }
        else        /* Bad frame, or parity error */
        {
          T0  =  old_T0;
          T0_frac = 0;
          old_T0++;
          if( (old_T0 - PIT_MAX) > 0)
            old_T0 = PIT_MAX;
        }

      }
      else                  /* second subframe */
      {
        if( bfi == 0)
        {
          dec_lag3(index, PIT_MIN, PIT_MAX, i_subfr, &T0, &T0_frac);
          old_T0 = T0;
        }
        else
        {
          T0  =  old_T0;
          T0_frac = 0;
          old_T0++;
          if( (old_T0 - PIT_MAX) > 0)
            old_T0 = PIT_MAX;
        }
      }
      *T2++ = T0;

     /*-------------------------------------------------*
      * - Find the adaptive codebook vector.            *
      *-------------------------------------------------*/

      pred_lt_3(&exc[i_subfr], T0, T0_frac, L_SUBFR);

      /*-------------------------------------------------------*
       * - Decode innovative codebook.                         *
       * - Add the fixed-gain pitch contribution to code[].    *
       *-------------------------------------------------------*/

      if(bfi != 0)        /* Bad frame */
      {
        parm[0] = random_g729() & (INT16)0x1fff;     /* 13 bits random */
        parm[1] = random_g729() & (INT16)0x000f;     /*  4 bits random */
      }

      decod_ACELP(parm[1], parm[0], code);
      parm +=2;

      for (i = T0; i < L_SUBFR; i++)   code[i] += sharp * code[i-T0];

      /*-------------------------------------------------*
       * - Decode pitch and codebook gains.              *
       *-------------------------------------------------*/

      index = *parm++;          /* index of energy VQ */

      dec_gain(index, code, L_SUBFR, bfi, &gain_pitch, &gain_code);

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

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

      /*-------------------------------------------------------*
       * - Find the total excitation.                          *
       * - Find synthesis speech corresponding to exc[].       *
       *-------------------------------------------------------*/

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

      syn_filt(Az, &exc[i_subfr], &synth[i_subfr], L_SUBFR, mem_syn, 1);

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

  /*--------------------------------------------------*
   * Update signal for next frame.                    *
   * -> shift to the left by L_FRAME  exc[]           *
   *--------------------------------------------------*/

   copy(&old_exc[L_FRAME], &old_exc[0], PIT_MAX+L_INTERPOL);

   return;
}