annotate libaf/af_surround.c @ 12337:6f1b4c989914

soft skipping for mencoder. rather than skipping decoding/filtering frames that will be skipped, mencoded tells vf_softskip (if present) that it should drop the next frame. this allows filters that need to see every input frame (inverse telecine, denoise3d, ...) to see skipped frames before they get dropped. in principle, a smarter softskip filter could be written that would buffer frames and choose to drop the one with least change, rather than strictly dropping the next one.

/* 
   This is an libaf filter 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 20ms 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 behavior of a single
     surround sent to rear in anti-phase */
#define SPLITREAR 1

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

#include "af.h"
#include "dsp.h"

#define L  32    // Length of fir filter
#define LD 65536 // Length of delay buffer

// 32 Tap fir filter loop unrolled
#define FIR(x,w,y) \
  y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3]  \
      + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7]  \
      + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
      + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
      + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
      + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
      + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
      + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])

// Add to circular queue macro + update index
#ifdef SPLITREAR
#define ADDQUE(qi,rq,lq,r,l)\
  lq[qi]=lq[qi+L]=(l);\
  rq[qi]=rq[qi+L]=(r);\
  qi=(qi-1)&(L-1);
#else
#define ADDQUE(qi,lq,l)\
  lq[qi]=lq[qi+L]=(l);\
  qi=(qi-1)&(L-1);
#endif

// Macro for updating queue index in delay queues
#define UPDATEQI(qi) qi=(qi+1)&(LD-1)

// instance data
typedef struct af_surround_s
{
  float lq[2*L]; // Circular queue for filtering left rear channel
  float rq[2*L]; // Circular queue for filtering right rear channel
  float w[L]; 	 // FIR filter coefficients for surround sound 7kHz low-pass
  float* dr;	 // Delay queue right rear channel
  float* dl;	 // Delay queue left rear channel
  float  d;	 // Delay time
  int i;       	 // Position in circular buffer
  int wi;	 // Write index for delay queue
  int ri;	 // Read index for delay queue
}af_surround_t;

// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
  af_surround_t *s = af->setup;
  switch(cmd){
  case AF_CONTROL_REINIT:{
    float fc;
    af->data->rate   = ((af_data_t*)arg)->rate;
    af->data->nch    = ((af_data_t*)arg)->nch*2;
    af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
    af->data->bps    = 4;

    if (af->data->nch != 4){
      af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
      return AF_DETACH;
    }
    // Surround filer coefficients
    fc = 2.0 * 7000.0/(float)af->data->rate;
    if (-1 == design_fir(L, s->w, &fc, LP|HAMMING, 0)){
      af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n");
      return AF_ERROR;
    }

    // Free previous delay queues
    if(s->dl)
      free(s->dl);
    if(s->dr)
      free(s->dr);
    // Allocate new delay queues
    s->dl = calloc(LD,af->data->bps);
    s->dr = calloc(LD,af->data->bps);
    if((NULL == s->dl) || (NULL == s->dr))
      af_msg(AF_MSG_FATAL,"[delay] Out of memory\n");
    
    // Initialize delay queue index
    if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
      return AF_ERROR;
    printf("%i\n",s->wi);
    s->ri = 0;

    if((af->data->format != ((af_data_t*)arg)->format) || 
       (af->data->bps    != ((af_data_t*)arg)->bps)){
      ((af_data_t*)arg)->format = af->data->format;
      ((af_data_t*)arg)->bps = af->data->bps;
      return AF_FALSE;
    }
    return AF_OK;
  }
  case AF_CONTROL_COMMAND_LINE:{
    float d = 0;
    sscanf((char*)arg,"%f",&d);
    if ((d < 0) || (d > 1000)){
      af_msg(AF_MSG_ERROR,"[surround] Invalid delay time, valid time values"
	     " are 0ms to 1000ms current value is %0.3f ms\n",d);
      return AF_ERROR;
    }
    s->d = d;
    return AF_OK;
  }
  }
  return AF_UNKNOWN;
}

// Deallocate memory
static void uninit(struct af_instance_s* af)
{
  if(af->data->audio)
    free(af->data->audio);
  if(af->data)
    free(af->data);
  if(af->setup)
    free(af->setup);
}

// The beginnings of an active matrix...
static float 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 dominance
//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;

// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data){
  af_surround_t* s   = (af_surround_t*)af->setup;
  float*	 m   = steering_matrix[0]; 
  float*     	 in  = data->audio; 	// Input audio data
  float*     	 out = NULL;		// Output audio data
  float*	 end = in + data->len / sizeof(float); // Loop end
  int 		 i   = s->i;	// Filter queue index
  int 		 ri  = s->ri;	// Read index for delay queue
  int 		 wi  = s->wi;	// Write index for delay queue

  if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
    return NULL;

  out = af->data->audio;

  while(in < end){
    /* 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 +1, Rt -1, 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] = m[0]*in[0] + m[1]*in[1];
    out[1] = m[2]*in[0] + m[3]*in[1];

    // Low-pass output @ 7kHz
    FIR((&s->lq[i]), s->w, s->dl[wi]);

    // Delay output by d ms
    out[2] = s->dl[ri];

#ifdef SPLITREAR
    // Low-pass output @ 7kHz
    FIR((&s->rq[i]), s->w, s->dr[wi]);

    // Delay output by d ms
    out[3] = s->dr[ri];
#else
    out[3] = -out[2];
#endif

    // Update delay queues indexes
    UPDATEQI(ri);
    UPDATEQI(wi);

    // Calculate and save surround in circular queue
#ifdef SPLITREAR
    ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
#else
    ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
#endif

    // Next sample...
    in = &in[data->nch];  
    out = &out[af->data->nch];
  }
  
  // Save indexes
  s->i  = i; s->ri = ri; s->wi = wi;

  // Set output data
  data->audio = af->data->audio;
  data->len   = (data->len*af->mul.n)/af->mul.d;
  data->nch   = af->data->nch;

  return data;
}

static int open(af_instance_t* af){
  af->control=control;
  af->uninit=uninit;
  af->play=play;
  af->mul.n=2;
  af->mul.d=1;
  af->data=calloc(1,sizeof(af_data_t));
  af->setup=calloc(1,sizeof(af_surround_t));
  if(af->data == NULL || af->setup == NULL)
    return AF_ERROR;
  ((af_surround_t*)af->setup)->d = 20;
  return AF_OK;
}

af_info_t af_info_surround =
{
        "Surround decoder filter",
        "surround",
        "Steve Davies <[e-mail]>",
        "",
        AF_FLAGS_NOT_REENTRANT,
        open
};