comparison libao2/pl_surround.c @ 3681:26126e5c3532

surround channels can now be decoded in stereoish

3680:2cc5737ec923

   Original author: Steve Davies <[e-mail]>
*/

/* The principle:  Make rear channels by extracting anti-phase data
   from the front channels, delay by 15msec and feed to rear in anti-phase
   www.dolby.com has the background
*/


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

  int format;           // input format
  int input_channels;   // input channels

} pl_surround_t;

static pl_surround_t pl_surround={0,15,NULL,NULL,NULL,0,0,NULL,0,0,0};

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

  pl_surround.delaybuf_pos = 0;
  memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
  memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
}


// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
  int16_t *in, *out;
  int i, samples;
  int surround;

  if (pl_surround.passthrough) return 1;

  // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);

  samples  = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
  out = pl_surround.databuf;  in = (int16_t *)ao_plugin_data.data;

  // Testing - place a 1kHz tone in the front channels in anti-phase
  //sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
  //sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);

  for (i=0; i<samples; i++) {

    // About volume balancing...
    //   Surround encoding does the following:
    //       Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
    //   So S should be extracted as:
    //       (Lt-Rt)
    //   But we are splitting the S to two output channels, so we
    //   must take 3dB off as we split it:
    //       Ls=Rs=.707*(Lt-Rt)
    //   Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
    //   clip.  So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2).

    // output front left and right
    out[0] = in[0]*.707;
    out[1] = in[1]*.707;
    // output Ls and Rs - from 15msec ago, lowpass filtered @ 7kHz
    out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
		       pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
    out[3] = - out[2];
    //    out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos,
    //		       pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
    // calculate and save surround for 15msecs time
    surround = (in[0]/2 - in[1]/2);
    pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] = surround;
    pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] = - surround;
    pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
    // next samples...
    in = &in[pl_surround.input_channels];  out = &out[4];
  }
  
  // Set output block/len
  ao_plugin_data.data=pl_surround.databuf;
  ao_plugin_data.len=samples*sizeof(int16_t)*4;
  return 1;

3681:26126e5c3532

   Original author: Steve Davies <[e-mail]>
*/

/* The principle:  Make rear channels by extracting anti-phase data
   from the front channels, delay by 20msec and feed to rear in anti-phase
*/


// SPLITREAR: Define to decode two distinct rear channels -
// 	this doesn't work so well in practice because
//      separation in a passive matrix is not high.
//      C (dialogue) to Ls and Rs 14dB or so -
//      so dialogue leaks to the rear.
//      Still - give it a try and send feedback.
//      comment this define for old behaviour of a single
//      surround sent to rear in anti-phase
#define SPLITREAR


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

  int format;           // input format
  int input_channels;   // input channels

} pl_surround_t;

static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};

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

  pl_surround.delaybuf_pos = 0;
  memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
  memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
}

// The beginnings of an active matrix...
static double steering_matrix[][12] = {
//	LL	RL	LR	RR	LS	RS	LLs	RLs	LRs	RRs	LC	RC	
       {.707,	.0,	.0,	.707,	.5,	-.5,	.5878,	-.3928,	.3928,	-.5878,	.5,	.5},
};

// Experimental moving average dominances
static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;

// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
  int16_t *in, *out;
  int i, samples;
  double *matrix = steering_matrix[0]; // later we'll index based on detected dominance

  if (pl_surround.passthrough) return 1;

  // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);

  samples  = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
  out = pl_surround.databuf;  in = (int16_t *)ao_plugin_data.data;

  // Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
  //sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
  //sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);

  for (i=0; i<samples; i++) {

    // Dominance:
    //abs(in[0])  abs(in[1]);
    //abs(in[0]+in[1])  abs(in[0]-in[1]);
    //10 * log( abs(in[0]) / (abs(in[1])|1) );
    //10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) );

    // About volume balancing...
    //   Surround encoding does the following:
    //       Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
    //   So S should be extracted as:
    //       (Lt-Rt)
    //   But we are splitting the S to two output channels, so we
    //   must take 3dB off as we split it:
    //       Ls=Rs=.707*(Lt-Rt)
    //   Trouble is, Lt could be +32767, Rt -32768, so possibility that S will
    //   overflow.  So to avoid that, we cut L/R by 3dB (*.707), and S by 6dB (/2).
    //   this keeps the overall balance, but guarantees no overflow.

    // output front left and right
    out[0] = matrix[0]*in[0] + matrix[1]*in[1];
    out[1] = matrix[2]*in[0] + matrix[3]*in[1];
    // output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
    out[2] = firfilter(pl_surround.Ls_delaybuf, pl_surround.delaybuf_pos,
		       pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
#ifdef SPLITREAR
    out[3] = firfilter(pl_surround.Rs_delaybuf, pl_surround.delaybuf_pos,
		       pl_surround.delaybuf_len, 32, pl_surround.filter_coefs_surround);
#else
    out[3] = -out[2];
#endif
    // calculate and save surround for 20msecs time
#ifdef SPLITREAR
    pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
      matrix[6]*in[0] + matrix[7]*in[1];
    pl_surround.Rs_delaybuf[pl_surround.delaybuf_pos++] =
      matrix[8]*in[0] + matrix[9]*in[1];
#else
    pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
      matrix[4]*in[0] + matrix[5]*in[1];
#endif
    pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;

    // next samples...
    in = &in[pl_surround.input_channels];  out = &out[4];
  }

  // Show some state
  //printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
  
  // Set output block/len
  ao_plugin_data.data=pl_surround.databuf;
  ao_plugin_data.len=samples*sizeof(int16_t)*4;
  return 1;