view libaf/af_resample.h @ 16429:84174804804b

Updates to NUT spec: 1. remove average_bitrate 2. add other_stream_header, for subtitles and metadata 3. add max_pts to index 4. index_ptr - a 64 bit integer to say the total length of all index packets 5. specify how to write "multiple" indexes 6. change forward_ptr behavior, starts right after forward_ptr, ends after checksum 7. remove stream_id <-> stream_class limitation. 8. time_base_nom must also be non zero. 9. rename time_base_nom and time_base_denom, now timebase means the length of a tick, not amounts of ticks 10. remove (old?) sample_rate_mul stuff. 11. specify what exactly the checksum covers. 12. specify that stream classes which have multiple streams must have an info packet.. (in new Semantic requirements section) 13. Rename 'timestamp' to pts. 14. Change date of draft... 15. Add myself to authors...

/*=============================================================================
//	
//  This software has been released under the terms of the GNU General Public
//  license. See http://www.gnu.org/copyleft/gpl.html for details.
//
//  Copyright 2002 Anders Johansson [e-mail]
//
//=============================================================================
*/

/* This file contains the resampling engine, the sample format is
   controlled by the FORMAT parameter, the filter length by the L
   parameter and the resampling type by UP and DN. This file should
   only be included by af_resample.c 
*/ 

#undef L
#undef SHIFT
#undef FORMAT
#undef FIR
#undef ADDQUE

/* The lenght Lxx definition selects the length of each poly phase
   component. Valid definitions are L8 and L16 where the number
   defines the nuber of taps. This definition affects the
   computational complexity, the performance and the memory usage.
*/

/* The FORMAT_x parameter selects the sample format type currently
   float and int16 are supported. Thes two formats are selected by
   defining eiter FORMAT_F or FORMAT_I. The advantage of using float
   is that the amplitude and therefore the SNR isn't affected by the
   filtering, the disadvantage is that it is a lot slower.
*/

#if defined(FORMAT_I)
#define SHIFT >>16
#define FORMAT int16_t
#else 
#define SHIFT
#define FORMAT float
#endif

// Short filter
#if defined(L8) 

#define L   	8	// Filter length
// Unrolled loop to speed up execution 
#define FIR(x,w,y) \
  (y[0])  = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
            + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT



#else  /* L8/L16 */

#define L   	16
// Unrolled loop to speed up execution 
#define FIR(x,w,y) \
  y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
         + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
         + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
         + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT

#endif /* L8/L16 */

// Macro to add data to circular que 
#define ADDQUE(xi,xq,in)\
  xq[xi]=xq[(xi)+L]=*(in);\
  xi=((xi)-1)&(L-1);

#if defined(UP)

  uint32_t		ci    = l->nch; 	// Index for channels
  uint32_t		nch   = l->nch;   	// Number of channels
  uint32_t		inc   = s->up/s->dn; 
  uint32_t		level = s->up%s->dn; 
  uint32_t		up    = s->up;
  uint32_t		dn    = s->dn;
  uint32_t		ns    = c->len/l->bps;
  register FORMAT*	w     = s->w;

  register uint32_t	wi    = 0;
  register uint32_t	xi    = 0; 

  // Index current channel
  while(ci--){
    // Temporary pointers
    register FORMAT*	x     = s->xq[ci];
    register FORMAT*	in    = ((FORMAT*)c->audio)+ci;
    register FORMAT*	out   = ((FORMAT*)l->audio)+ci;
    FORMAT* 		end   = in+ns; // Block loop end
    wi = s->wi; xi = s->xi;

    while(in < end){
      register uint32_t	i = inc;
      if(wi<level) i++;

      ADDQUE(xi,x,in);
      in+=nch;
      while(i--){
	// Run the FIR filter
	FIR((&x[xi]),(&w[wi*L]),out);
	len++; out+=nch;
	// Update wi to point at the correct polyphase component
	wi=(wi+dn)%up;
      }
    }

  }
  // Save values that needs to be kept for next time
  s->wi = wi;
  s->xi = xi;
#endif /* UP */

#if defined(DN) /* DN */
  uint32_t		ci    = l->nch; 	// Index for channels
  uint32_t		nch   = l->nch;   	// Number of channels
  uint32_t		inc   = s->dn/s->up; 
  uint32_t		level = s->dn%s->up; 
  uint32_t		up    = s->up;
  uint32_t		dn    = s->dn;
  uint32_t		ns    = c->len/l->bps;
  FORMAT*		w     = s->w;

  register int32_t	i     = 0;
  register uint32_t	wi    = 0;
  register uint32_t	xi    = 0;
  
  // Index current channel
  while(ci--){
    // Temporary pointers
    register FORMAT*	x     = s->xq[ci];
    register FORMAT*	in    = ((FORMAT*)c->audio)+ci;
    register FORMAT*	out   = ((FORMAT*)l->audio)+ci;
    register FORMAT* 	end   = in+ns;    // Block loop end
    i = s->i; wi = s->wi; xi = s->xi;

    while(in < end){

      ADDQUE(xi,x,in);
      in+=nch;
      if((--i)<=0){
	// Run the FIR filter
	FIR((&x[xi]),(&w[wi*L]),out);
	len++;	out+=nch;

	// Update wi to point at the correct polyphase component
	wi=(wi+dn)%up;  

	// Insert i number of new samples in queue
	i = inc;
	if(wi<level) i++;
      }
    }
  }
  // Save values that needs to be kept for next time
  s->wi = wi;
  s->xi = xi;
  s->i = i;
#endif /* DN */