comparison libao2/pl_surround.c @ 3485:439b0b1d50b9

include 7kHz lowpass filter for surround channels, as per Dolby recommendation

3484:72e299c83b4f

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

static ao_info_t info =
{
        "Surround decoder plugin",
        "surround",
        "Steve Davies <[e-mail]>",

typedef struct pl_surround_s
{
  int passthrough;      // Just be a "NO-OP"
  int msecs;            // Rear channel delay in milliseconds
  int16_t* databuf;     // Output audio buffer
  int16_t* delaybuf;    // circular buffer to be used for delaying Ls and Rs audio
  int delaybuf_len;     // local buffer length in samples
  int delaybuf_ptr;     // offset in buffer where we are reading/writing
  int rate;             // input data rate
  int format;           // input format
  int input_channels;   // input channels

} pl_surround_t;

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

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

    //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
    // Allocate an output buffer
    if (pl_surround.databuf != NULL) {
      free(pl_surround.databuf);  pl_surround.databuf = NULL;
    }
    pl_surround.databuf = calloc(ao_plugin_data.len, 1);
    // Return back smaller len so we don't get overflowed...  (??seems the right thing to do?)
    ao_plugin_data.len /= 2;
    return CONTROL_OK;
  }
  return -1;
}

  // Input 2 channels, output will be 4 - tell ao_plugin
  ao_plugin_data.channels    = 4;
  ao_plugin_data.sz_mult    /= 2;

  // Figure out buffer space needed for the 15msec delay
  pl_surround.delaybuf_len = 2 * (pl_surround.rate * pl_surround.msecs / 1000);
  // Allocate delay buffer
  pl_surround.delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
  fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d bytes\n",
	  pl_surround.msecs,pl_surround.rate,  pl_surround.delaybuf_len*sizeof(int16_t));
  pl_surround.delaybuf_ptr = 0;

  return 1;
}

// close plugin
static void uninit(){
  //  fprintf(stderr, "pl_surround: uninit called!\n");
  if (pl_surround.passthrough) return;
  if(pl_surround.delaybuf) 
    free(pl_surround.delaybuf);
  if(pl_surround.databuf) 
    free(pl_surround.databuf);
  pl_surround.delaybuf_len=0;
}

// empty buffers
static void reset()
{
  if (pl_surround.passthrough) return;
  //fprintf(stderr, "pl_surround: reset called\n");
  pl_surround.delaybuf_ptr = 0;
  memset(pl_surround.delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
}


// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data

    //       (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 compensate, 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
    out[2] = pl_surround.delaybuf[pl_surround.delaybuf_ptr];
    out[3] = pl_surround.delaybuf[pl_surround.delaybuf_ptr+1];
    // calculate and save surround for 15msecs time
    surround = (in[0]/2 - in[1]/2);
    pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = surround;
    pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = - surround;
    pl_surround.delaybuf_ptr %= pl_surround.delaybuf_len;
    // next samples...
    in = &in[pl_surround.input_channels];  out = &out[4];
  }
  
  // Set output block/len

3485:439b0b1d50b9

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

#include "remez.h"
#include "firfilter.c"

static ao_info_t info =
{
        "Surround decoder plugin",
        "surround",
        "Steve Davies <[e-mail]>",

typedef struct pl_surround_s
{
  int passthrough;      // Just be a "NO-OP"
  int msecs;            // Rear channel delay in milliseconds
  int16_t* databuf;     // Output audio buffer
  int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
  int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
  int delaybuf_len;     // delaybuf buffer length in samples
  int delaybuf_pos;     // offset in buffer where we are reading/writing
  double* filter_coefs_surround; // FIR filter coefficients for surround sound 7kHz lowpass
  int rate;             // input data rate
  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:

    //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
    // Allocate an output buffer
    if (pl_surround.databuf != NULL) {
      free(pl_surround.databuf);  pl_surround.databuf = NULL;
    }
    // Allocate output buffer
    pl_surround.databuf = calloc(ao_plugin_data.len, 1);
    // Return back smaller len so we don't get overflowed...
    ao_plugin_data.len /= 2;
    return CONTROL_OK;
  }
  return -1;
}

  // Input 2 channels, output will be 4 - tell ao_plugin
  ao_plugin_data.channels    = 4;
  ao_plugin_data.sz_mult    /= 2;

  // Figure out buffer space (in int16_ts) needed for the 15msec delay
  pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000);
  // Allocate delay buffers
  pl_surround.Ls_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
  pl_surround.Rs_delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
  fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffers are %d bytes each\n",
	  pl_surround.msecs,pl_surround.rate,  pl_surround.delaybuf_len*sizeof(int16_t));
  pl_surround.delaybuf_pos = 0;
  // Surround filer coefficients
  pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);

  return 1;
}

// close plugin
static void uninit(){
  //  fprintf(stderr, "pl_surround: uninit called!\n");
  if (pl_surround.passthrough) return;
  if(pl_surround.Ls_delaybuf) 
    free(pl_surround.Ls_delaybuf);
  if(pl_surround.Rs_delaybuf) 
    free(pl_surround.Rs_delaybuf);
  if(pl_surround.databuf) 
    free(pl_surround.databuf);
  pl_surround.delaybuf_len=0;
}

// empty buffers
static void reset()
{
  if (pl_surround.passthrough) return;
  //fprintf(stderr, "pl_surround: reset called\n");
  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

    //       (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