mpv/libao2/pl_resample.c

/*=============================================================================
//	
//  This file is part of mplayer.
//
//  mplayer is free software; you can redistribute it and/or modify
//  it under the terms of the GNU General Public License as published by
//  the Free Software Foundation; either version 2 of the License, or
//  (at your option) any later version.
//
//  mplayer is distributed in the hope that it will be useful,
//  but WITHOUT ANY WARRANTY; without even the implied warranty of
//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
//  GNU General Public License for more details.
//
//  You should have received a copy of the GNU General Public License
//  along with mplayer; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
//
//  Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/

/* This audio output plugin changes the sample rate. The output
   samplerate from this plugin is specified by using the switch
   `fout=F' where F is the desired output sample frequency 
*/

#define PLUGIN

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>

#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",
        "Anders",
        ""
};

LIBAO_PLUGIN_EXTERN(resample)

#define min(a,b)   (((a) < (b)) ? (a) : (b))
#define max(a,b)   (((a) > (b)) ? (a) : (b))

/* Below definition selects the length of each poly phase component.
   Valid definitions are L4 and L8, where the number denotes the
   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? */

#include "fir.h"
#include "filter.h"

// local data
typedef struct pl_resample_s
{
  int16_t*	data;		// Data buffer
  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(pl_resample.data) 
      free(pl_resample.data);
    pl_resample.len = ao_plugin_data.len;
    pl_resample.data=(int16_t*)malloc(pl_resample.len);
    if(!pl_resample.data)
      return CONTROL_ERROR;
    ao_plugin_data.len = (int)((double)ao_plugin_data.len * 
			     ((double)pl_resample.up)/
			     ((double)pl_resample.dn));
    return CONTROL_OK;
  }
  return -1;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(){
  int fin=ao_plugin_data.rate;
  int fout=ao_plugin_cfg.pl_resample_fout;
  pl_resample.w=pl_resample.ws;
  pl_resample.up=UP;

  // Sheck input format
  if(ao_plugin_data.format != AFMT_S16_LE){
    fprintf(stderr,"[pl_resample] Input audio format not yet suported. \n");
    return 0;
  }
  // Sanity check and calculate down sampling factor
  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;
  }

  // Tell the world what we are up to
  printf("[pl_resample] Up=%i, Down=%i, True fout=%f\n",
	 pl_resample.up,pl_resample.dn,
	 ((float)fin*pl_resample.up)/((float)pl_resample.dn));

  // This plugin changes buffersize and adds some delay
  ao_plugin_data.sz_mult/=((float)pl_resample.up)/((float)pl_resample.dn);
  ao_plugin_data.delay_fix-= ((float)L/2) * (1/fout);
  ao_plugin_data.rate=fout;
  return 1;
}

// close plugin
static void uninit(){
  if(pl_resample.data) 
    free(pl_resample.data);
  pl_resample.data=NULL;
}

// empty buffers
static void reset(){
}

// 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];
    register int16_t*	in    = ((int16_t*)ao_plugin_data.data)+ci;
    register int16_t*	out   = pl_resample.data+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;
}