libaf/af_surround.c

   1 /*
   2    This is an libaf filter to do simple decoding of matrixed surround
   3    sound.  This will provide a (basic) surround-sound effect from
   4    audio encoded for Dolby Surround, Pro Logic etc.
   5
   6  * This program is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, write to the Free Software
  18  * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  19
  20    Original author: Steve Davies <steve@daviesfam.org>
  21 */
  22
  23 /* The principle:  Make rear channels by extracting anti-phase data
  24    from the front channels, delay by 20ms and feed to rear in anti-phase
  25 */
  26
  27
  28 /* SPLITREAR: Define to decode two distinct rear channels - this
  29      doesn't work so well in practice because separation in a passive
  30      matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
  31      dialogue leaks to the rear.  Still - give it a try and send
  32      feedback. Comment this define for old behavior of a single
  33      surround sent to rear in anti-phase */
  34 #define SPLITREAR 1
  35
  36 #include <stdio.h>
  37 #include <stdlib.h>
  38 #include <string.h>
  39 #include <unistd.h>
  40
  41 #include "af.h"
  42 #include "dsp.h"
  43
  44 #define L  32    // Length of fir filter
  45 #define LD 65536 // Length of delay buffer
  46
  47 // 32 Tap fir filter loop unrolled
  48 #define FIR(x,w,y) \
  49   y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3]  \
  50       + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7]  \
  51       + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
  52       + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
  53       + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
  54       + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
  55       + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
  56       + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
  57
  58 // Add to circular queue macro + update index
  59 #ifdef SPLITREAR
  60 #define ADDQUE(qi,rq,lq,r,l)\
  61   lq[qi]=lq[qi+L]=(l);\
  62   rq[qi]=rq[qi+L]=(r);\
  63   qi=(qi-1)&(L-1);
  64 #else
  65 #define ADDQUE(qi,lq,l)\
  66   lq[qi]=lq[qi+L]=(l);\
  67   qi=(qi-1)&(L-1);
  68 #endif
  69
  70 // Macro for updating queue index in delay queues
  71 #define UPDATEQI(qi) qi=(qi+1)&(LD-1)
  72
  73 // instance data
  74 typedef struct af_surround_s
  75 {
  76   float lq[2*L]; // Circular queue for filtering left rear channel
  77   float rq[2*L]; // Circular queue for filtering right rear channel
  78   float w[L];    // FIR filter coefficients for surround sound 7kHz low-pass
  79   float* dr;     // Delay queue right rear channel
  80   float* dl;     // Delay queue left rear channel
  81   float  d;      // Delay time
  82   int i;         // Position in circular buffer
  83   int wi;        // Write index for delay queue
  84   int ri;        // Read index for delay queue
  85 }af_surround_t;
  86
  87 // Initialization and runtime control
  88 static int control(struct af_instance_s* af, int cmd, void* arg)
  89 {
  90   af_surround_t *s = af->setup;
  91   switch(cmd){
  92   case AF_CONTROL_REINIT:{
  93     float fc;
  94     af->data->rate   = ((af_data_t*)arg)->rate;
  95     af->data->nch    = ((af_data_t*)arg)->nch*2;
  96     af->data->format = AF_FORMAT_FLOAT_NE;
  97     af->data->bps    = 4;
  98
  99     if (af->data->nch != 4){
 100       af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
 101       return AF_DETACH;
 102     }
 103     // Surround filer coefficients
 104     fc = 2.0 * 7000.0/(float)af->data->rate;
 105     if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
 106       af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n");
 107       return AF_ERROR;
 108     }
 109
 110     // Free previous delay queues
 111     if(s->dl)
 112       free(s->dl);
 113     if(s->dr)
 114       free(s->dr);
 115     // Allocate new delay queues
 116     s->dl = calloc(LD,af->data->bps);
 117     s->dr = calloc(LD,af->data->bps);
 118     if((NULL == s->dl) || (NULL == s->dr))
 119       af_msg(AF_MSG_FATAL,"[delay] Out of memory\n");
 120
 121     // Initialize delay queue index
 122     if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
 123       return AF_ERROR;
 124 //    printf("%i\n",s->wi);
 125     s->ri = 0;
 126
 127     if((af->data->format != ((af_data_t*)arg)->format) ||
 128        (af->data->bps    != ((af_data_t*)arg)->bps)){
 129       ((af_data_t*)arg)->format = af->data->format;
 130       ((af_data_t*)arg)->bps = af->data->bps;
 131       return AF_FALSE;
 132     }
 133     return AF_OK;
 134   }
 135   case AF_CONTROL_COMMAND_LINE:{
 136     float d = 0;
 137     sscanf((char*)arg,"%f",&d);
 138     if ((d < 0) || (d > 1000)){
 139       af_msg(AF_MSG_ERROR,"[surround] Invalid delay time, valid time values"
 140              " are 0ms to 1000ms current value is %0.3f ms\n",d);
 141       return AF_ERROR;
 142     }
 143     s->d = d;
 144     return AF_OK;
 145   }
 146   }
 147   return AF_UNKNOWN;
 148 }
 149
 150 // Deallocate memory
 151 static void uninit(struct af_instance_s* af)
 152 {
 153   if(af->data->audio)
 154     free(af->data->audio);
 155   if(af->data)
 156     free(af->data);
 157   if(af->setup)
 158     free(af->setup);
 159 }
 160
 161 // The beginnings of an active matrix...
 162 static float steering_matrix[][12] = {
 163 //      LL      RL      LR      RR      LS      RS
 164 //      LLs     RLs     LRs     RRs     LC      RC
 165        {.707,   .0,     .0,     .707,   .5,     -.5,
 166         .5878,  -.3928, .3928,  -.5878, .5,     .5},
 167 };
 168
 169 // Experimental moving average dominance
 170 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
 171
 172 // Filter data through filter
 173 static af_data_t* play(struct af_instance_s* af, af_data_t* data){
 174   af_surround_t* s   = (af_surround_t*)af->setup;
 175   float*         m   = steering_matrix[0];
 176   float*         in  = data->audio;     // Input audio data
 177   float*         out = NULL;            // Output audio data
 178   float*         end = in + data->len / sizeof(float); // Loop end
 179   int            i   = s->i;    // Filter queue index
 180   int            ri  = s->ri;   // Read index for delay queue
 181   int            wi  = s->wi;   // Write index for delay queue
 182
 183   if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
 184     return NULL;
 185
 186   out = af->data->audio;
 187
 188   while(in < end){
 189     /* Dominance:
 190        abs(in[0])  abs(in[1]);
 191        abs(in[0]+in[1])  abs(in[0]-in[1]);
 192        10 * log( abs(in[0]) / (abs(in[1])|1) );
 193        10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
 194
 195     /* About volume balancing...
 196        Surround encoding does the following:
 197            Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
 198        So S should be extracted as:
 199            (Lt-Rt)
 200        But we are splitting the S to two output channels, so we
 201        must take 3dB off as we split it:
 202            Ls=Rs=.707*(Lt-Rt)
 203        Trouble is, Lt could be +1, Rt -1, so possibility that S will
 204        overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
 205        6dB (/2). This keeps the overall balance, but guarantees no
 206        overflow. */
 207
 208     // Output front left and right
 209     out[0] = m[0]*in[0] + m[1]*in[1];
 210     out[1] = m[2]*in[0] + m[3]*in[1];
 211
 212     // Low-pass output @ 7kHz
 213     FIR((&s->lq[i]), s->w, s->dl[wi]);
 214
 215     // Delay output by d ms
 216     out[2] = s->dl[ri];
 217
 218 #ifdef SPLITREAR
 219     // Low-pass output @ 7kHz
 220     FIR((&s->rq[i]), s->w, s->dr[wi]);
 221
 222     // Delay output by d ms
 223     out[3] = s->dr[ri];
 224 #else
 225     out[3] = -out[2];
 226 #endif
 227
 228     // Update delay queues indexes
 229     UPDATEQI(ri);
 230     UPDATEQI(wi);
 231
 232     // Calculate and save surround in circular queue
 233 #ifdef SPLITREAR
 234     ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
 235 #else
 236     ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
 237 #endif
 238
 239     // Next sample...
 240     in = &in[data->nch];
 241     out = &out[af->data->nch];
 242   }
 243
 244   // Save indexes
 245   s->i  = i; s->ri = ri; s->wi = wi;
 246
 247   // Set output data
 248   data->audio = af->data->audio;
 249   data->len   = (data->len*af->mul.n)/af->mul.d;
 250   data->nch   = af->data->nch;
 251
 252   return data;
 253 }
 254
 255 static int open(af_instance_t* af){
 256   af->control=control;
 257   af->uninit=uninit;
 258   af->play=play;
 259   af->mul.n=2;
 260   af->mul.d=1;
 261   af->data=calloc(1,sizeof(af_data_t));
 262   af->setup=calloc(1,sizeof(af_surround_t));
 263   if(af->data == NULL || af->setup == NULL)
 264     return AF_ERROR;
 265   ((af_surround_t*)af->setup)->d = 20;
 266   return AF_OK;
 267 }
 268
 269 af_info_t af_info_surround =
 270 {
 271         "Surround decoder filter",
 272         "surround",
 273         "Steve Davies <steve@daviesfam.org>",
 274         "",
 275         AF_FLAGS_NOT_REENTRANT,
 276         open
 277 };