comparison libao2/pl_resample.c @ 4171:585f0c77d8f5

Sync problem when using fractional resampling fixed + speed increased.

4170:32ad3c2833cf

#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
//#include "../config.h"

static ao_info_t info =
{
        "Sample frequency conversion audio plugin",
        "resample",

   length of the filter. This definition affects the computational
   complexity (see play()), the performance (see filter.h) and the
   memory usage. For now the filterlenght is choosen to 4 and without
   assembly optimization if no SSE is present.
*/
#ifdef HAVE_SSE
#define L8    	1	// Filter bank type
#define W 	W8	// Filter bank parameters
#define L   	8	// Filter length
#else	
#define L4	1
#define W 	W4
#define L   	4
#endif

#define CH  6	// Max number of channels
#define UP  128  /* Up sampling factor. Increasing this value will
                    improve frequency accuracy. Think about the L1
                    cashing of filter parameters - how big can it be? */

  int16_t*  	w;		// Current filter weights
  uint16_t  	dn;     	// Down sampling factor
  uint16_t	up;		// Up sampling factor 
  int 		channels;	// Number of channels
  int 		len;		// Lenght of buffer
  int 		bypass;		// Bypass this plugin
  int16_t	ws[UP*L];	// List of all available filters	
  int16_t 	xs[CH][L*2]; 	// Circular buffers
} pl_resample_t;

static pl_resample_t 	pl_resample	= {NULL,NULL,1,1,1,0,0,W};

// to set/get/query special features/parameters
static int control(int cmd,int arg){
  switch(cmd){
  case AOCONTROL_PLUGIN_SET_LEN:

  if((float)max(fin,fout)/(float)min(fin,fout) > 10){
    fprintf(stderr,"[pl_resample] The difference between fin and fout is too large.\n");
    return 0;
  }
  pl_resample.dn=(int)(0.5+((float)(fin*pl_resample.up))/((float)fout));
  if(pl_resample.dn == pl_resample.up){
    fprintf(stderr,"[pl_resample] Fin is too close to fout no conversion is needed.\n");
    pl_resample.bypass=1;
    return 1;
  }
  pl_resample.channels=ao_plugin_data.channels;
  if(ao_plugin_data.channels>CH){
     fprintf(stderr,"[pl_resample] Too many channels, max is 6.\n");
    return 0;
  }

// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
// FIXME: this routine needs to be optimized (it is probably possible to do a lot here)
static int play(){
  static uint16_t	pwi = 0; // Index for w
  static uint16_t	pxi = 0; // Index for circular queue
  static uint16_t	pi =  1; // Number of new samples to put in x queue

  uint16_t		ci    = pl_resample.channels; 	// Index for channels
  uint16_t		len   = 0; 			// Number of output samples
  uint16_t		nch   = pl_resample.channels;   // Number of channels
  uint16_t		inc   = pl_resample.dn/pl_resample.up; 
  uint16_t		level = pl_resample.dn%pl_resample.up; 
  uint16_t		up    = pl_resample.up;
  uint16_t		dn    = pl_resample.dn;

  register uint16_t	i,wi,xi; // Temporary indexes

  if(pl_resample.bypass)
    return 1;
  
  // Index current channel
  while(ci--){
    // Temporary pointers
    register int16_t*	x     = pl_resample.xs[ci];

    // Block loop end
    register int16_t* 	end   = in+ao_plugin_data.len/2;
    i = pi; wi = pwi; xi = pxi;

    LOAD_QUE(x);
    if(0!=i) goto L1; 
    while(in < end){
      // Update wi to point at the correct polyphase component
      wi=(wi+dn)%up;  

      /* Update circular buffer x. This loop will be updated 0 or 1 time
	 for upsamling and inc or inc + 1 times for downsampling */
      if(wi<level) goto L3;
      if(0==i) goto L2;
  L1:   i--;
  L3:  	UPDATE_QUE(in);
        in+=nch;
	if(in >= end) goto L2;
      if(i) goto L1;
  L2: if(i) goto L5;
      i=inc;

      /* Get the correct polyphase component and the correct startpoint
	 in the circular bufer and run the FIR filter */
      FIR((&x[xi]),(&pl_resample.w[wi*L]),out);
      len++;
      out+=nch;
    }
L5:
    SAVE_QUE(x);
  }

  // Save values that needs to be kept for next time
  pwi = wi;
  pxi = xi;
  pi = i;
  // Set new data
  ao_plugin_data.len=len*2;
  ao_plugin_data.data=pl_resample.data;
  return 1;
}

4171:585f0c77d8f5

#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
// #include "../config.h"

static ao_info_t info =
{
        "Sample frequency conversion audio plugin",
        "resample",

   length of the filter. This definition affects the computational
   complexity (see play()), the performance (see filter.h) and the
   memory usage. For now the filterlenght is choosen to 4 and without
   assembly optimization if no SSE is present.
*/

// #ifdef HAVE_SSE
#define L8    	1	// Filter bank type
#define W 	W8	// Filter bank parameters
#define L   	8	// Filter length
// #else	
// #define L4	1
// #define W 	W4
// #define L   	4
// #endif

#define CH  6	// Max number of channels
#define UP  128  /* Up sampling factor. Increasing this value will
                    improve frequency accuracy. Think about the L1
                    cashing of filter parameters - how big can it be? */

  int16_t*  	w;		// Current filter weights
  uint16_t  	dn;     	// Down sampling factor
  uint16_t	up;		// Up sampling factor 
  int 		channels;	// Number of channels
  int 		len;		// Lenght of buffer
  int16_t	ws[UP*L];	// List of all available filters	
  int16_t 	xs[CH][L*2]; 	// Circular buffers
} pl_resample_t;

static pl_resample_t 	pl_resample	= {NULL,NULL,1,1,1,0,W};

// to set/get/query special features/parameters
static int control(int cmd,int arg){
  switch(cmd){
  case AOCONTROL_PLUGIN_SET_LEN:

  if((float)max(fin,fout)/(float)min(fin,fout) > 10){
    fprintf(stderr,"[pl_resample] The difference between fin and fout is too large.\n");
    return 0;
  }
  pl_resample.dn=(int)(0.5+((float)(fin*pl_resample.up))/((float)fout));

  pl_resample.channels=ao_plugin_data.channels;
  if(ao_plugin_data.channels>CH){
     fprintf(stderr,"[pl_resample] Too many channels, max is 6.\n");
    return 0;
  }

// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
// FIXME: this routine needs to be optimized (it is probably possible to do a lot here)
static int play(){
  if(pl_resample.up==pl_resample.dn){
    register int16_t*	in    = ((int16_t*)ao_plugin_data.data);
    register int16_t* 	end   = in+ao_plugin_data.len/2;
    while(in < end) *in=(*in++)>>1;
    return 1;
  }
  if(pl_resample.up>pl_resample.dn)
    return upsample();
  if(pl_resample.up<pl_resample.dn)
    return downsample();
}

int upsample(){
  static uint16_t	pwi = 0; // Index for w
  static uint16_t	pxi = 0; // Index for circular queue

  uint16_t		ci    = pl_resample.channels; 	// Index for channels
  uint16_t		nch   = pl_resample.channels;   // Number of channels
  uint16_t		len   = 0; 			// Number of input samples
  uint16_t		inc   = pl_resample.up/pl_resample.dn; 
  uint16_t		level = pl_resample.up%pl_resample.dn; 
  uint16_t		up    = pl_resample.up;
  uint16_t		dn    = pl_resample.dn;

  register int16_t*	w     = pl_resample.w;
  register uint16_t	wi,xi; // Temporary indexes

  // Index current channel
  while(ci--){
    // Temporary pointers
    register int16_t*	x     = pl_resample.xs[ci];
    register int16_t*	in    = ((int16_t*)ao_plugin_data.data)+ci;
    register int16_t*	out   = pl_resample.data+ci;
    int16_t* 		end   = in+ao_plugin_data.len/2; // Block loop end

    wi = pwi; xi = pxi;

    LOAD_QUE(x);
    while(in < end){
      register uint16_t	i = inc;
      if(wi<level) i++;

      UPDATE_QUE(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_QUE(x);
  }

  // Save values that needs to be kept for next time
  pwi = wi;
  pxi = xi;

  // Set new data
  ao_plugin_data.len=len*2;
  ao_plugin_data.data=pl_resample.data;
  return 1;
}

int downsample(){
  static uint16_t	pwi = 0; // Index for w
  static uint16_t	pxi = 0; // Index for circular queue
  static uint16_t	pi =  1; // Number of new samples to put in x queue

  uint16_t		ci    = pl_resample.channels; 	// Index for channels
  uint16_t		len   = 0; 			// Number of input samples
  uint16_t		nch   = pl_resample.channels;   // Number of channels
  uint16_t		inc   = pl_resample.dn/pl_resample.up; 
  uint16_t		level = pl_resample.dn%pl_resample.up; 
  uint16_t		up    = pl_resample.up;
  uint16_t		dn    = pl_resample.dn;

  register uint16_t	i,wi,xi; // Temporary indexes

  
  // Index current channel
  while(ci--){
    // Temporary pointers
    register int16_t*	x     = pl_resample.xs[ci];

    // Block loop end
    register int16_t* 	end   = in+ao_plugin_data.len/2;
    i = pi; wi = pwi; xi = pxi;

    LOAD_QUE(x);
    while(in < end){

      UPDATE_QUE(in);
      in+=nch;
      
      if(!--i){
	// Run the FIR filter
	FIR((&x[xi]),(&pl_resample.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_QUE(x);
  }
  // Save values that needs to be kept for next time
  pwi = wi;
  pxi = xi;
  pi = i;
  // Set new data
  ao_plugin_data.len=len*2;
  ao_plugin_data.data=pl_resample.data;
  return 1;
}