view libao2/pl_surround.c @ 3639:64ee21b3bd09

Modified the sync code once again, commented out hardware pts sync (I'll likely burn in hell before understanding how to get this bastard to sync well) Added automagic setup of aspect ratio, will remove the "aspect-bug" (I hope). As well as please you rich 16:9 doods ;)

/* 
   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;
}