view libao2/pl_surround.c @ 3591:f7fa59e4da4f

similar to 1.18, full language support for gui

/* 
   This is an ao2 plugin to do simple decoding of matrixed surround
   sound.  This will provide a (basic) surround-sound effect from
   audio encoded for Dolby Surround, Pro Logic etc.

 * This program 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.
 *
 * This program 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 this program; if not, write to the Free Software
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

   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>

#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]>",
        ""
};

LIBAO_PLUGIN_EXTERN(surround)

// local data
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:
    if (pl_surround.passthrough) return CONTROL_OK;
    //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
    //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;
}

// open & setup audio device
// return: 1=success 0=fail
static int init(){

  fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
  if (ao_plugin_data.channels != 2) {
    fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
    pl_surround.passthrough = 1;
    return 1;
  }
  if (ao_plugin_data.format != AFMT_S16_LE) {
    fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n");
    pl_surround.passthrough = 1;
    return 1;
  }

  pl_surround.passthrough = 0;

  /* Store info on input format to expect */
  pl_surround.rate=ao_plugin_data.rate;
  pl_surround.format=ao_plugin_data.format;
  pl_surround.input_channels=ao_plugin_data.channels;

  // 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
  // Extra 31 samples allow for lowpass filter delay (taps-1)
  pl_surround.delaybuf_len = (pl_surround.rate * pl_surround.msecs / 1000) + 31;
  // 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);
  //dump_filter_coefficients(pl_surround.filter_coefs_surround);
  //testfilter(pl_surround.filter_coefs_surround, 32, 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
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;
}