www.pudn.com > AVS_M_ver10.rar > deci12k8.c

/* 
*********************************************************************** 
* COPYRIGHT AND WARRANTY INFORMATION 
* 
* Copyright 2007  Audio Video Coding Standard, Part ¢ú 
* 
* This software module was developed by AVS Audio sub-group 
* 
* DISCLAIMER OF WARRANTY 
* 
* These software programs are available to the users without any 
* license fee or royalty on an "as is" basis. The AVS disclaims 
* any and all warranties, whether express, implied, or statutory, 
* including any implied warranties of merchantability or of fitness 
* for a particular purpose. In no event shall the contributors or  
* the AVS be liable for any incidental, punitive, or consequential 
* damages of any kind whatsoever arising from the use of this program. 
* 
* This disclaimer of warranty extends to the user of this program 
* and user's customers, employees, agents, transferees, successors, 
* and assigns. 
* 
* The AVS does not represent or warrant that the program furnished 
* hereunder are free of infringement of any third-party patents. 
* Commercial implementations of AVS, including shareware, may be 
* subject to royalty fees to patent holders. Information regarding 
* the AVS patent policy is available from the AVS Web site at 
* http://www.avs.org.cn 
* 
* THIS IS NOT A GRANT OF PATENT RIGHTS - SEE THE AVS PATENT POLICY. 
************************************************************************ 
*/ 
 
/*-------------------------------------------------------------------* 
 * Function decim_12k8() and oversamp_12k8()                         * 
 * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~                         * 
 * decim_12k8    : decimation from fs to 12.8kHz.                    * 
 * oversamp_12k8 : oversampling from 12.8kHz to fs.                  * 
 *                                                                   * 
 * fs = 16/22/24/28.8/32/44/48 kHz.                                  * 
 *-------------------------------------------------------------------*/ 
#include  
#include  
#include  
#include "../include/amr_plus.h" 
#define L_FRAME_MAX   (L_FRAME24k) 
#define L_FILT_MAX    (L_FILT24k) 
#define L_SUBFR16k   80       /* Subframe size at 16kHz           */ 
static void interpol_add(float *signal, float *signal_int, int L_frame_int, 
                  const float *filter, int nb_coef, int fac_up, int fac_down, float gain); 
void decim_12k8( 
  float sig_fs[],     /* input:  signal to decimate      */ 
  int lg,             /* input:  length of input         */ 
  float sig12k8[],    /* output: decimated signal        */ 
  float mem[],        /* in/out: memory (2*L_FILT_FS)    */ 
  int band)           /* input:  0=0..6.4k, 1=6.4..10.8k */ 
{ 
  int i, ncoef; 
  float signal[(2*L_FILT_MAX)+L_FRAME_MAX]; 
  const float *filter; 
  float gain; 
  int fac_up, fac_down; 
  gain = 1.0; 
  switch (lg) { 
  case L_FRAME16kPLUS: 
    ncoef = NCOEF_32k; 
    fac_up = FAC1_32k; 
    fac_down = FAC2_32k; 
    if (band == 0) { 
      filter = filter_32k; 
    } else { 
      if (lg == L_FRAME16kPLUS) { 
        filter = filter_32k_7k; 
      } else { 
        filter = filter_32k_hf; 
      } 
    }     
    break; 
  case L_FRAME24k: 
    ncoef = NCOEF_48k; 
    fac_up = FAC1_48k; 
    fac_down = FAC2_48k; 
    if (band == 0) { 
      filter = filter_48k; 
    } else { 
      filter = filter_48k_hf; 
    } 
    break; 
  case L_FRAME8k: /* This mode actually upsamples rather than down */ 
    ncoef = NCOEF_32k>>2; 
    fac_up = FAC1_32k*2; 
    fac_down = FAC2_32k; 
    filter = filter_8k; 
    gain = 2.0f; 
    break; 
  default: 
    printf("wrong frame size in decim_12k8!\n"); 
    exit(0); 
    break; 
  }       
  if (lg <= L_FRAME24k) { 
    ncoef = (ncoef+1)>>1; 
    fac_up *= 2; 
    gain *= 2.0; 
  }     
  for (i=0; i<(2*ncoef); i++) { 
    signal[i] = mem[i]; 
  } 
  for (i=0; i fac_up_minus_frac_step) { 
      pos += pos_step_plus_one; 
      frac -= fac_up_minus_frac_step; 
    } 
    else { 
      pos += pos_step; 
      frac += frac_step; 
    } 
  } 
  return; 
} 
static void interpol_add(float *signal, float *signal_int, int L_frame_int, 
                  const float *filter, int nb_coef, int fac_up, int fac_down, float gain) 
{ 
  int i, j, frac, frac_step, pos, pos_step; 
  int pos_step_plus_one, fac_up_minus_frac_step; 
  float s, *x1, *x2, tmp; 
  const float *c1, *c2; 
  pos_step = fac_down / fac_up; 
  pos_step_plus_one = pos_step + 1; 
  frac_step = fac_down - (pos_step*fac_up); 
  fac_up_minus_frac_step = fac_up-frac_step; 
  pos = 0; 
  frac = 0; 
  for(i=0; i fac_up_minus_frac_step) { 
      pos += pos_step_plus_one; 
      frac -= fac_up_minus_frac_step; 
    } 
    else { 
      pos += pos_step; 
      frac += frac_step; 
    } 
  } 
  return; 
}