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 along
  17  * with this program; if not, write to the Free Software Foundation, Inc.,
  18  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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
  40 #include "af.h"
  41 #include "dsp.h"
  42
  43 #define L  32    // Length of fir filter
  44 #define LD 65536 // Length of delay buffer
  45
  46 // 32 Tap fir filter loop unrolled
  47 #define FIR(x,w,y) \
  48   y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3]  \
  49       + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7]  \
  50       + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
  51       + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
  52       + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
  53       + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
  54       + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
  55       + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
  56
  57 // Add to circular queue macro + update index
  58 #ifdef SPLITREAR
  59 #define ADDQUE(qi,rq,lq,r,l)\
  60   lq[qi]=lq[qi+L]=(l);\
  61   rq[qi]=rq[qi+L]=(r);\
  62   qi=(qi-1)&(L-1);
  63 #else
  64 #define ADDQUE(qi,lq,l)\
  65   lq[qi]=lq[qi+L]=(l);\
  66   qi=(qi-1)&(L-1);
  67 #endif
  68
  69 // Macro for updating queue index in delay queues
  70 #define UPDATEQI(qi) qi=(qi+1)&(LD-1)
  71
  72 // instance data
  73 typedef struct af_surround_s
  74 {
  75   float lq[2*L]; // Circular queue for filtering left rear channel
  76   float rq[2*L]; // Circular queue for filtering right rear channel
  77   float w[L];    // FIR filter coefficients for surround sound 7kHz low-pass
  78   float* dr;     // Delay queue right rear channel
  79   float* dl;     // Delay queue left rear channel
  80   float  d;      // Delay time
  81   int i;         // Position in circular buffer
  82   int wi;        // Write index for delay queue
  83   int ri;        // Read index for delay queue
  84 }af_surround_t;
  85
  86 // Initialization and runtime control
  87 static int control(struct af_instance_s* af, int cmd, void* arg)
  88 {
  89   af_surround_t *s = af->setup;
  90   switch(cmd){
  91   case AF_CONTROL_REINIT:{
  92     float fc;
  93     af->data->rate   = ((af_data_t*)arg)->rate;
  94     af->data->nch    = ((af_data_t*)arg)->nch*2;
  95     af->data->format = AF_FORMAT_FLOAT_NE;
  96     af->data->bps    = 4;
  97
  98     if (af->data->nch != 4){
  99       af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
 100       return AF_DETACH;
 101     }
 102     // Surround filer coefficients
 103     fc = 2.0 * 7000.0/(float)af->data->rate;
 104     if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
 105       af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n");
 106       return AF_ERROR;
 107     }
 108
 109     // Free previous delay queues
 110     if(s->dl)
 111       free(s->dl);
 112     if(s->dr)
 113       free(s->dr);
 114     // Allocate new delay queues
 115     s->dl = calloc(LD,af->data->bps);
 116     s->dr = calloc(LD,af->data->bps);
 117     if((NULL == s->dl) || (NULL == s->dr))
 118       af_msg(AF_MSG_FATAL,"[delay] Out of memory\n");
 119
 120     // Initialize delay queue index
 121     if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
 122       return AF_ERROR;
 123 //    printf("%i\n",s->wi);
 124     s->ri = 0;
 125
 126     if((af->data->format != ((af_data_t*)arg)->format) ||
 127        (af->data->bps    != ((af_data_t*)arg)->bps)){
 128       ((af_data_t*)arg)->format = af->data->format;
 129       ((af_data_t*)arg)->bps = af->data->bps;
 130       return AF_FALSE;
 131     }
 132     return AF_OK;
 133   }
 134   case AF_CONTROL_COMMAND_LINE:{
 135     float d = 0;
 136     sscanf((char*)arg,"%f",&d);
 137     if ((d < 0) || (d > 1000)){
 138       af_msg(AF_MSG_ERROR,"[surround] Invalid delay time, valid time values"
 139              " are 0ms to 1000ms current value is %0.3f ms\n",d);
 140       return AF_ERROR;
 141     }
 142     s->d = d;
 143     return AF_OK;
 144   }
 145   }
 146   return AF_UNKNOWN;
 147 }
 148
 149 // Deallocate memory
 150 static void uninit(struct af_instance_s* af)
 151 {
 152   if(af->data)
 153     free(af->data->audio);
 154   free(af->data);
 155   free(af->setup);
 156 }
 157
 158 // The beginnings of an active matrix...
 159 static float steering_matrix[][12] = {
 160 //      LL      RL      LR      RR      LS      RS
 161 //      LLs     RLs     LRs     RRs     LC      RC
 162        {.707,   .0,     .0,     .707,   .5,     -.5,
 163         .5878,  -.3928, .3928,  -.5878, .5,     .5},
 164 };
 165
 166 // Experimental moving average dominance
 167 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
 168
 169 // Filter data through filter
 170 static af_data_t* play(struct af_instance_s* af, af_data_t* data){
 171   af_surround_t* s   = (af_surround_t*)af->setup;
 172   float*         m   = steering_matrix[0];
 173   float*         in  = data->audio;     // Input audio data
 174   float*         out = NULL;            // Output audio data
 175   float*         end = in + data->len / sizeof(float); // Loop end
 176   int            i   = s->i;    // Filter queue index
 177   int            ri  = s->ri;   // Read index for delay queue
 178   int            wi  = s->wi;   // Write index for delay queue
 179
 180   if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
 181     return NULL;
 182
 183   out = af->data->audio;
 184
 185   while(in < end){
 186     /* Dominance:
 187        abs(in[0])  abs(in[1]);
 188        abs(in[0]+in[1])  abs(in[0]-in[1]);
 189        10 * log( abs(in[0]) / (abs(in[1])|1) );
 190        10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
 191
 192     /* About volume balancing...
 193        Surround encoding does the following:
 194            Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
 195        So S should be extracted as:
 196            (Lt-Rt)
 197        But we are splitting the S to two output channels, so we
 198        must take 3dB off as we split it:
 199            Ls=Rs=.707*(Lt-Rt)
 200        Trouble is, Lt could be +1, Rt -1, so possibility that S will
 201        overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
 202        6dB (/2). This keeps the overall balance, but guarantees no
 203        overflow. */
 204
 205     // Output front left and right
 206     out[0] = m[0]*in[0] + m[1]*in[1];
 207     out[1] = m[2]*in[0] + m[3]*in[1];
 208
 209     // Low-pass output @ 7kHz
 210     FIR((&s->lq[i]), s->w, s->dl[wi]);
 211
 212     // Delay output by d ms
 213     out[2] = s->dl[ri];
 214
 215 #ifdef SPLITREAR
 216     // Low-pass output @ 7kHz
 217     FIR((&s->rq[i]), s->w, s->dr[wi]);
 218
 219     // Delay output by d ms
 220     out[3] = s->dr[ri];
 221 #else
 222     out[3] = -out[2];
 223 #endif
 224
 225     // Update delay queues indexes
 226     UPDATEQI(ri);
 227     UPDATEQI(wi);
 228
 229     // Calculate and save surround in circular queue
 230 #ifdef SPLITREAR
 231     ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
 232 #else
 233     ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
 234 #endif
 235
 236     // Next sample...
 237     in = &in[data->nch];
 238     out = &out[af->data->nch];
 239   }
 240
 241   // Save indexes
 242   s->i  = i; s->ri = ri; s->wi = wi;
 243
 244   // Set output data
 245   data->audio = af->data->audio;
 246   data->len   *= 2;
 247   data->nch   = af->data->nch;
 248
 249   return data;
 250 }
 251
 252 static int af_open(af_instance_t* af){
 253   af->control=control;
 254   af->uninit=uninit;
 255   af->play=play;
 256   af->mul=2;
 257   af->data=calloc(1,sizeof(af_data_t));
 258   af->setup=calloc(1,sizeof(af_surround_t));
 259   if(af->data == NULL || af->setup == NULL)
 260     return AF_ERROR;
 261   ((af_surround_t*)af->setup)->d = 20;
 262   return AF_OK;
 263 }
 264
 265 af_info_t af_info_surround =
 266 {
 267         "Surround decoder filter",
 268         "surround",
 269         "Steve Davies <steve@daviesfam.org>",
 270         "",
 271         AF_FLAGS_NOT_REENTRANT,
 272         af_open
 273 };