www.pudn.com > AVS_M_ver10.rar > decim_split.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. 
************************************************************************ 
*/ 
 
#include  
#include  
#include  
#include "../include/amr_plus.h" 
#define L_FRAME_MAX   (L_FRAME48k) 
 
 
 
/* local function */ 
static void interpol_mem( 
  float *signal,  
  float *signal_int,  
  int L_frame_int, 
  float *filter,  
  int nb_coef,  
  int fac_up,  
  int fac_down,  
  float gain, 
  int *mem_frac); 
 
 
int decim_split_12k8( /* number of sample decimated         */ 
  float sig_fs[],     /* input:  signal to decimate         */ 
  int lg_input,       /* input:  2*L_FRAME44k if 44kHz      */ 
  float sig12k8_lf[], /* output: LF decimated signal        */ 
  float sig12k8_hf[], /* output: HF decimated signal        */ 
  int lg,             /* input:  length of LF and HF        */ 
  int fac_fs,         /* input:  at 48kHz, scale fac = fac_fs/FSCALE_DENOM */ 
  int  fac_up,        /* input:  Upsampling factor */ 
  int  fac_down,      /* input:  Downsampling factor */ 
  int L_frame,        /* input:  Working frame len */ 
  float mem[],        /* in/out: mem[L_MEM_DECIM_SPLIT]     */ 
  int *frac_mem       /* in/out: interpol fraction memory   */ 
) 
{ 
  int i, j, ncoef, L_frame_int; 
  float signal[(2*(L_FILT_SPLIT+L_FILT_DECIM))+2*L_FRAME_MAX], *sig; 
  float *filter, gain, *x1, *x2, s; 
 
#ifdef FILTER_44kHz 
  if (lg_input == (2*L_FRAME44k)) 
  { 
    filter = (float *)filter_LP165; 
  } 
#endif 
#ifdef FILTER_48kHz 
  if (lg_input == (2*L_FRAME48k)) 
  { 
    filter = (float *)filter_LP180; 
  } 
#endif 
 
  ncoef = (L_FILT_OVER+1)*fac_down/fac_up; 
  gain = ((float)fac_up) / ((float)fac_down); 
  /* frames length */ 
  L_frame_int = 2*lg;     /* 25k6 rate */ 
  /* load buffer & update memory */ 
  for (i=0; i<(2*(L_FILT_SPLIT+L_FILT_DECIM)); i++)  
  { 
    signal[i] = mem[i]; 
  } 
  for (i=0; i 2 x 12.8kHz) */ 
  sig = signal + L_FILT_SPLIT; 
  for(i=0; i 25.6kHz) */ 
  /* load buffer (LF-HF) & update memory */ 
  for (i=0; i<(2*L_FILT_JOIN); i++)  
  { 
    signal[i] = mem[i+(2*L_FILT_OVER)]; 
  } 
  for (i=0; i> 3; 
  pos = 0; 
  frac = *mem_frac; 
   
  for(i=0; i> 3; 
    c1 = &filter[frac_d]; 
    c2 = &filter[fac_up_d-frac_d]; 
 
    s = 0.0; 
 
    for(j=0; j= fac_up_minus_frac_step)  
    { 
      pos += pos_step_plus_one; 
      frac -= fac_up_minus_frac_step; 
    } 
    else  
    { 
      pos += pos_step; 
      frac += frac_step; 
    } 
  } 
  *mem_frac = frac; 
  return; 
}