research.m1stereo.tv/gitweb/ - melted/blob - src/modules/normalize/filter_volume.c

   1 /*
   2  * filter_volume.c -- adjust audio volume
   3  * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
   4  * Author: Dan Dennedy <dan@dennedy.org>
   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 Foundation,
  18  * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
  19  */
  20
  21 #include <framework/mlt_filter.h>
  22 #include <framework/mlt_frame.h>
  23
  24 #include <stdio.h>
  25 #include <stdlib.h>
  26 #include <math.h>
  27 #include <ctype.h>
  28 #include <string.h>
  29
  30 #define MAX_CHANNELS 6
  31 #define EPSILON 0.00001
  32
  33 /* The following normalise functions come from the normalize utility:
  34    Copyright (C) 1999--2002 Chris Vaill */
  35
  36 #define samp_width 16
  37
  38 #ifndef ROUND
  39 # define ROUND(x) floor((x) + 0.5)
  40 #endif
  41
  42 #define DBFSTOAMP(x) pow(10,(x)/20.0)
  43
  44 /** Return nonzero if the two strings are equal, ignoring case, up to
  45     the first n characters.
  46 */
  47 int strncaseeq(const char *s1, const char *s2, size_t n)
  48 {
  49         for ( ; n > 0; n--)
  50         {
  51                 if (tolower(*s1++) != tolower(*s2++))
  52                         return 0;
  53         }
  54         return 1;
  55 }
  56
  57 /** Limiter function.
  58
  59          / tanh((x + lev) / (1-lev)) * (1-lev) - lev        (for x < -lev)
  60          |
  61     x' = | x                                                (for |x| <= lev)
  62          |
  63          \ tanh((x - lev) / (1-lev)) * (1-lev) + lev        (for x > lev)
  64
  65   With limiter level = 0, this is equivalent to a tanh() function;
  66   with limiter level = 1, this is equivalent to clipping.
  67 */
  68 static inline double limiter( double x, double lmtr_lvl )
  69 {
  70         double xp = x;
  71
  72         if (x < -lmtr_lvl)
  73                 xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl;
  74         else if (x > lmtr_lvl)
  75                 xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl;
  76
  77 //      if ( x != xp )
  78 //              fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp );
  79
  80         return xp;
  81 }
  82
  83
  84 /** Takes a full smoothing window, and returns the value of the center
  85     element, smoothed.
  86
  87     Currently, just does a mean filter, but we could do a median or
  88     gaussian filter here instead.
  89 */
  90 static inline double get_smoothed_data( double *buf, int count )
  91 {
  92         int i, j;
  93         double smoothed = 0;
  94
  95         for ( i = 0, j = 0; i < count; i++ )
  96         {
  97                 if ( buf[ i ] != -1.0 )
  98                 {
  99                         smoothed += buf[ i ];
 100                         j++;
 101                 }
 102         }
 103         smoothed /= j;
 104 //      fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed );
 105
 106         return smoothed;
 107 }
 108
 109 /** Get the max power level (using RMS) and peak level of the audio segment.
 110  */
 111 double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *peak )
 112 {
 113         // Determine numeric limits
 114         int bytes_per_samp = (samp_width - 1) / 8 + 1;
 115         int16_t max = (1 << (bytes_per_samp * 8 - 1)) - 1;
 116         int16_t min = -max - 1;
 117
 118         double *sums = (double *) calloc( channels, sizeof(double) );
 119         int c, i;
 120         int16_t sample;
 121         double pow, maxpow = 0;
 122
 123         /* initialize peaks to effectively -inf and +inf */
 124         int16_t max_sample = min;
 125         int16_t min_sample = max;
 126
 127         for ( i = 0; i < samples; i++ )
 128         {
 129                 for ( c = 0; c < channels; c++ )
 130                 {
 131                         sample = *buffer++;
 132                         sums[ c ] += (double) sample * (double) sample;
 133
 134                         /* track peak */
 135                         if ( sample > max_sample )
 136                                 max_sample = sample;
 137                         else if ( sample < min_sample )
 138                                 min_sample = sample;
 139                 }
 140         }
 141         for ( c = 0; c < channels; c++ )
 142         {
 143                 pow = sums[ c ] / (double) samples;
 144                 if ( pow > maxpow )
 145                         maxpow = pow;
 146         }
 147
 148         free( sums );
 149
 150         /* scale the pow value to be in the range 0.0 -- 1.0 */
 151         maxpow /= ( (double) min * (double) min);
 152
 153         if ( -min_sample > max_sample )
 154                 *peak = min_sample / (double) min;
 155         else
 156                 *peak = max_sample / (double) max;
 157
 158         return sqrt( maxpow );
 159 }
 160
 161 /* ------ End normalize functions --------------------------------------- */
 162
 163 /** Get the audio.
 164 */
 165
 166 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
 167 {
 168         // Get the properties of the a frame
 169         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 170         double gain = mlt_properties_get_double( properties, "volume.gain" );
 171         double max_gain = mlt_properties_get_double( properties, "volume.max_gain" );
 172         double limiter_level = 0.5; /* -6 dBFS */
 173         int normalise =  mlt_properties_get_int( properties, "volume.normalise" );
 174         double amplitude =  mlt_properties_get_double( properties, "volume.amplitude" );
 175         int i, j;
 176         double sample;
 177         int16_t peak;
 178
 179         // Get the filter from the frame
 180         mlt_filter this = mlt_properties_get_data( properties, "filter_volume", NULL );
 181
 182         // Get the properties from the filter
 183         mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
 184
 185         if ( mlt_properties_get( properties, "volume.limiter" ) != NULL )
 186                 limiter_level = mlt_properties_get_double( properties, "volume.limiter" );
 187
 188         // Get the producer's audio
 189         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
 190 //      fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
 191
 192         // Determine numeric limits
 193         int bytes_per_samp = (samp_width - 1) / 8 + 1;
 194         int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
 195         int samplemin = -samplemax - 1;
 196
 197         if ( normalise )
 198         {
 199                 int window = mlt_properties_get_int( filter_props, "window" );
 200                 double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
 201
 202                 if ( window > 0 && smooth_buffer != NULL )
 203                 {
 204                         int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
 205
 206                         // Compute the signal power and put into smoothing buffer
 207                         smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
 208 //                      fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
 209                         if ( smooth_buffer[ smooth_index ] > EPSILON )
 210                         {
 211                                 mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
 212
 213                                 // Smooth the data and compute the gain
 214 //                              fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
 215                                 gain *= amplitude / get_smoothed_data( smooth_buffer, window );
 216                         }
 217                 }
 218                 else
 219                 {
 220                         gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak );
 221                 }
 222         }
 223
 224 //      if ( gain > 1.0 && normalise )
 225 //              fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
 226
 227         if ( max_gain > 0 && gain > max_gain )
 228                 gain = max_gain;
 229
 230         // Initialise filter's previous gain value to prevent an inadvertant jump from 0
 231         if ( mlt_properties_get( filter_props, "previous_gain" ) == NULL )
 232                 mlt_properties_set_double( filter_props, "previous_gain", gain );
 233
 234         // Start the gain out at the previous
 235         double previous_gain = mlt_properties_get_double( filter_props, "previous_gain" );
 236
 237         // Determine ramp increment
 238         double gain_step = ( gain - previous_gain ) / *samples;
 239 //      fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
 240
 241         // Save the current gain for the next iteration
 242         mlt_properties_set_double( filter_props, "previous_gain", gain );
 243
 244         // Ramp from the previous gain to the current
 245         gain = previous_gain;
 246
 247         int16_t *p = *buffer;
 248
 249         // Apply the gain
 250         for ( i = 0; i < *samples; i++ )
 251         {
 252                 for ( j = 0; j < *channels; j++ )
 253                 {
 254                         sample = *p * gain;
 255                         *p = ROUND( sample );
 256
 257                         if ( gain > 1.0 )
 258                         {
 259                                 /* use limiter function instead of clipping */
 260                                 if ( normalise )
 261                                         *p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
 262
 263                                 /* perform clipping */
 264                                 else if ( sample > samplemax )
 265                                         *p = samplemax;
 266                                 else if ( sample < samplemin )
 267                                         *p = samplemin;
 268                         }
 269                         p++;
 270                 }
 271                 gain += gain_step;
 272         }
 273
 274         return 0;
 275 }
 276
 277 /** Filter processing.
 278 */
 279
 280 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 281 {
 282         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 283         mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
 284
 285         // Parse the gain property
 286         if ( mlt_properties_get( properties, "gain" ) == NULL )
 287         {
 288                 double gain = 1.0; // no adjustment
 289
 290                 if ( mlt_properties_get( filter_props, "gain" ) != NULL )
 291                 {
 292                         char *p = mlt_properties_get( filter_props, "gain" );
 293
 294                         if ( strncaseeq( p, "normalise", 9 ) )
 295                                 mlt_properties_set( filter_props, "normalise", "" );
 296                         else
 297                         {
 298                                 if ( strcmp( p, "" ) != 0 )
 299                                         gain = fabs( strtod( p, &p) );
 300
 301                                 while ( isspace( *p ) )
 302                                         p++;
 303
 304                                 /* check if "dB" is given after number */
 305                                 if ( strncaseeq( p, "db", 2 ) )
 306                                         gain = DBFSTOAMP( gain );
 307
 308                                 // If there is an end adjust gain to the range
 309                                 if ( mlt_properties_get( filter_props, "end" ) != NULL )
 310                                 {
 311                                         // Determine the time position of this frame in the transition duration
 312                                         mlt_position in = mlt_filter_get_in( this );
 313                                         mlt_position out = mlt_filter_get_out( this );
 314                                         mlt_position time = mlt_frame_get_position( frame );
 315                                         double position = ( double )( time - in ) / ( double )( out - in + 1 );
 316
 317                                         double end = -1;
 318                                         char *p = mlt_properties_get( filter_props, "end" );
 319                                         if ( strcmp( p, "" ) != 0 )
 320                                                 end = fabs( strtod( p, &p) );
 321
 322                                         while ( isspace( *p ) )
 323                                                 p++;
 324
 325                                         /* check if "dB" is given after number */
 326                                         if ( strncaseeq( p, "db", 2 ) )
 327                                                 end = DBFSTOAMP( gain );
 328
 329                                         if ( end != -1 )
 330                                                 gain += ( end - gain ) * position;
 331                                 }
 332                         }
 333                 }
 334                 mlt_properties_set_double( properties, "volume.gain", gain );
 335         }
 336
 337         // Parse the maximum gain property
 338         if ( mlt_properties_get( filter_props, "max_gain" ) != NULL )
 339         {
 340                 char *p = mlt_properties_get( filter_props, "max_gain" );
 341                 double gain = fabs( strtod( p, &p) ); // 0 = no max
 342
 343                 while ( isspace( *p ) )
 344                         p++;
 345
 346                 /* check if "dB" is given after number */
 347                 if ( strncaseeq( p, "db", 2 ) )
 348                         gain = DBFSTOAMP( gain );
 349
 350                 mlt_properties_set_double( properties, "volume.max_gain", gain );
 351         }
 352
 353         // Parse the limiter property
 354         if ( mlt_properties_get( filter_props, "limiter" ) != NULL )
 355         {
 356                 char *p = mlt_properties_get( filter_props, "limiter" );
 357                 double level = 0.5; /* -6dBFS */
 358                 if ( strcmp( p, "" ) != 0 )
 359                         level = strtod( p, &p);
 360
 361                 while ( isspace( *p ) )
 362                         p++;
 363
 364                 /* check if "dB" is given after number */
 365                 if ( strncaseeq( p, "db", 2 ) )
 366                 {
 367                         if ( level > 0 )
 368                                 level = -level;
 369                         level = DBFSTOAMP( level );
 370                 }
 371                 else
 372                 {
 373                         if ( level < 0 )
 374                                 level = -level;
 375                 }
 376                 mlt_properties_set_double( properties, "volume.limiter", level );
 377         }
 378
 379         // Parse the normalise property
 380         if ( mlt_properties_get( filter_props, "normalise" ) != NULL )
 381         {
 382                 char *p = mlt_properties_get( filter_props, "normalise" );
 383                 double amplitude = 0.2511886431509580; /* -12dBFS */
 384                 if ( strcmp( p, "" ) != 0 )
 385                         amplitude = strtod( p, &p);
 386
 387                 while ( isspace( *p ) )
 388                         p++;
 389
 390                 /* check if "dB" is given after number */
 391                 if ( strncaseeq( p, "db", 2 ) )
 392                 {
 393                         if ( amplitude > 0 )
 394                                 amplitude = -amplitude;
 395                         amplitude = DBFSTOAMP( amplitude );
 396                 }
 397                 else
 398                 {
 399                         if ( amplitude < 0 )
 400                                 amplitude = -amplitude;
 401                         if ( amplitude > 1.0 )
 402                                 amplitude = 1.0;
 403                 }
 404
 405                 // If there is an end adjust gain to the range
 406                 if ( mlt_properties_get( filter_props, "end" ) != NULL )
 407                 {
 408                         // Determine the time position of this frame in the transition duration
 409                         mlt_position in = mlt_filter_get_in( this );
 410                         mlt_position out = mlt_filter_get_out( this );
 411                         mlt_position time = mlt_frame_get_position( frame );
 412                         double position = ( double )( time - in ) / ( double )( out - in + 1 );
 413                         amplitude *= position;
 414                 }
 415                 mlt_properties_set_int( properties, "volume.normalise", 1 );
 416                 mlt_properties_set_double( properties, "volume.amplitude", amplitude );
 417         }
 418
 419         // Parse the window property and allocate smoothing buffer if needed
 420         int window = mlt_properties_get_int( filter_props, "window" );
 421         if ( mlt_properties_get( filter_props, "smooth_buffer" ) == NULL && window > 1 )
 422         {
 423                 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
 424                 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
 425                 int i;
 426                 for ( i = 0; i < window; i++ )
 427                         smooth_buffer[ i ] = -1.0;
 428                 mlt_properties_set_data( filter_props, "smooth_buffer", smooth_buffer, 0, free, NULL );
 429         }
 430
 431         // Put a filter reference onto the frame
 432         mlt_properties_set_data( properties, "filter_volume", this, 0, NULL, NULL );
 433
 434         // Override the get_audio method
 435         mlt_frame_push_audio( frame, filter_get_audio );
 436
 437         return frame;
 438 }
 439
 440 /** Constructor for the filter.
 441 */
 442
 443 mlt_filter filter_volume_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
 444 {
 445         mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 );
 446         if ( this != NULL && mlt_filter_init( this, NULL ) == 0 )
 447         {
 448                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
 449                 this->process = filter_process;
 450                 if ( arg != NULL )
 451                         mlt_properties_set( properties, "gain", arg );
 452
 453                 mlt_properties_set_int( properties, "window", 75 );
 454                 mlt_properties_set( properties, "max_gain", "20dB" );
 455         }
 456         return this;
 457 }