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 "filter_volume.h"
  22
  23 #include <framework/mlt_frame.h>
  24
  25 #include <stdio.h>
  26 #include <stdlib.h>
  27 #include <math.h>
  28 #include <ctype.h>
  29 #include <string.h>
  30
  31 #define MAX_CHANNELS 6
  32 #define EPSILON 0.00001
  33
  34 /* The normalise functions come from the normalize utility:
  35    Copyright (C) 1999--2002 Chris Vaill */
  36
  37 #define samp_width 16
  38
  39 #ifndef ROUND
  40 # define ROUND(x) floor((x) + 0.5)
  41 #endif
  42
  43 #define DBFSTOAMP(x) pow(10,(x)/20.0)
  44
  45 /** Return nonzero if the two strings are equal, ignoring case, up to
  46     the first n characters.
  47 */
  48 int strncaseeq(const char *s1, const char *s2, size_t n)
  49 {
  50         for ( ; n > 0; n--)
  51         {
  52                 if (tolower(*s1++) != tolower(*s2++))
  53                         return 0;
  54         }
  55         return 1;
  56 }
  57
  58 /** Limiter function.
  59
  60          / tanh((x + lev) / (1-lev)) * (1-lev) - lev        (for x < -lev)
  61          |
  62     x' = | x                                                (for |x| <= lev)
  63          |
  64          \ tanh((x - lev) / (1-lev)) * (1-lev) + lev        (for x > lev)
  65
  66   With limiter level = 0, this is equivalent to a tanh() function;
  67   with limiter level = 1, this is equivalent to clipping.
  68 */
  69 static inline double limiter( double x, double lmtr_lvl )
  70 {
  71         double xp = x;
  72
  73         if (x < -lmtr_lvl)
  74                 xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl;
  75         else if (x > lmtr_lvl)
  76                 xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl;
  77
  78 //      if ( x != xp )
  79 //              fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp );
  80
  81         return xp;
  82 }
  83
  84
  85 /** Takes a full smoothing window, and returns the value of the center
  86     element, smoothed.
  87
  88     Currently, just does a mean filter, but we could do a median or
  89     gaussian filter here instead.
  90 */
  91 static inline double get_smoothed_data( double *buf, int count )
  92 {
  93         int i, j;
  94         double smoothed = 0;
  95
  96         for ( i = 0, j = 0; i < count; i++ )
  97         {
  98                 if ( buf[ i ] != -1.0 )
  99                 {
 100                         smoothed += buf[ i ];
 101                         j++;
 102                 }
 103         }
 104         smoothed /= j;
 105 //      fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed );
 106
 107         return smoothed;
 108 }
 109
 110 /** Get the max power level (using RMS) and peak level of the audio segment.
 111  */
 112 double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *peak )
 113 {
 114         // Determine numeric limits
 115         int bytes_per_samp = (samp_width - 1) / 8 + 1;
 116         int16_t max = (1 << (bytes_per_samp * 8 - 1)) - 1;
 117         int16_t min = -max - 1;
 118
 119         double *sums = (double *) calloc( channels, sizeof(double) );
 120         int c, i;
 121         int16_t sample;
 122         double pow, maxpow = 0;
 123
 124         /* initialize peaks to effectively -inf and +inf */
 125         int16_t max_sample = min;
 126         int16_t min_sample = max;
 127
 128         for ( i = 0; i < samples; i++ )
 129         {
 130                 for ( c = 0; c < channels; c++ )
 131                 {
 132                         sample = *buffer++;
 133                         sums[ c ] += (double) sample * (double) sample;
 134
 135                         /* track peak */
 136                         if ( sample > max_sample )
 137                                 max_sample = sample;
 138                         else if ( sample < min_sample )
 139                                 min_sample = sample;
 140                 }
 141         }
 142         for ( c = 0; c < channels; c++ )
 143         {
 144                 pow = sums[ c ] / (double) samples;
 145                 if ( pow > maxpow )
 146                         maxpow = pow;
 147         }
 148
 149         free( sums );
 150
 151         /* scale the pow value to be in the range 0.0 -- 1.0 */
 152         maxpow /= ( (double) min * (double) min);
 153
 154         if ( -min_sample > max_sample )
 155                 *peak = min_sample / (double) min;
 156         else
 157                 *peak = max_sample / (double) max;
 158
 159         return sqrt( maxpow );
 160 }
 161
 162 /* ------ End normalize functions --------------------------------------- */
 163
 164 /** Get the audio.
 165 */
 166
 167 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
 168 {
 169         // Get the properties of the a frame
 170         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 171         double gain = mlt_properties_get_double( properties, "volume.gain" );
 172         double max_gain = mlt_properties_get_double( properties, "volume.max_gain" );
 173         double limiter_level = 0.5; /* -6 dBFS */
 174         int normalise =  mlt_properties_get_int( properties, "volume.normalise" );
 175         double amplitude =  mlt_properties_get_double( properties, "volume.amplitude" );
 176         int i, j;
 177         double sample;
 178         int16_t peak;
 179
 180         // Get the filter from the frame
 181         mlt_filter this = mlt_properties_get_data( properties, "filter_volume", NULL );
 182
 183         // Get the properties from the filter
 184         mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
 185
 186         if ( mlt_properties_get( properties, "volume.limiter" ) != NULL )
 187                 limiter_level = mlt_properties_get_double( properties, "volume.limiter" );
 188
 189         // Get the producer's audio
 190         mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
 191 //      fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
 192
 193         // Determine numeric limits
 194         int bytes_per_samp = (samp_width - 1) / 8 + 1;
 195         int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
 196         int samplemin = -samplemax - 1;
 197
 198         if ( normalise )
 199         {
 200                 int window = mlt_properties_get_int( filter_props, "window" );
 201                 double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
 202
 203                 if ( window > 0 && smooth_buffer != NULL )
 204                 {
 205                         int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
 206
 207                         // Compute the signal power and put into smoothing buffer
 208                         smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
 209 //                      fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
 210                         if ( smooth_buffer[ smooth_index ] > EPSILON )
 211                         {
 212                                 mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
 213
 214                                 // Smooth the data and compute the gain
 215 //                              fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
 216                                 gain *= amplitude / get_smoothed_data( smooth_buffer, window );
 217                         }
 218                 }
 219                 else
 220                 {
 221                         gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak );
 222                 }
 223         }
 224
 225 //      if ( gain > 1.0 && normalise )
 226 //              fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
 227
 228         if ( max_gain > 0 && gain > max_gain )
 229                 gain = max_gain;
 230
 231         // Initialise filter's previous gain value to prevent an inadvertant jump from 0
 232         if ( mlt_properties_get( filter_props, "previous_gain" ) == NULL )
 233                 mlt_properties_set_double( filter_props, "previous_gain", gain );
 234
 235         // Start the gain out at the previous
 236         double previous_gain = mlt_properties_get_double( filter_props, "previous_gain" );
 237
 238         // Determine ramp increment
 239         double gain_step = ( gain - previous_gain ) / *samples;
 240 //      fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
 241
 242         // Save the current gain for the next iteration
 243         mlt_properties_set_double( filter_props, "previous_gain", gain );
 244
 245         // Ramp from the previous gain to the current
 246         gain = previous_gain;
 247
 248         int16_t *p = *buffer;
 249
 250         // Apply the gain
 251         for ( i = 0; i < *samples; i++ )
 252         {
 253                 for ( j = 0; j < *channels; j++ )
 254                 {
 255                         sample = *p * gain;
 256                         *p = ROUND( sample );
 257
 258                         if ( gain > 1.0 )
 259                         {
 260                                 /* use limiter function instead of clipping */
 261                                 if ( normalise )
 262                                         *p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
 263
 264                                 /* perform clipping */
 265                                 else if ( sample > samplemax )
 266                                         *p = samplemax;
 267                                 else if ( sample < samplemin )
 268                                         *p = samplemin;
 269                         }
 270                         p++;
 271                 }
 272                 gain += gain_step;
 273         }
 274
 275         return 0;
 276 }
 277
 278 /** Filter processing.
 279 */
 280
 281 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 282 {
 283         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 284         mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
 285
 286         // Parse the gain property
 287         if ( mlt_properties_get( properties, "gain" ) == NULL )
 288         {
 289                 double gain = 1.0; // no adjustment
 290
 291                 if ( mlt_properties_get( filter_props, "gain" ) != NULL )
 292                 {
 293                         char *p = mlt_properties_get( filter_props, "gain" );
 294
 295                         if ( strncaseeq( p, "normalise", 9 ) )
 296                                 mlt_properties_set( filter_props, "normalise", "" );
 297                         else
 298                         {
 299                                 if ( strcmp( p, "" ) != 0 )
 300                                         gain = fabs( strtod( p, &p) );
 301
 302                                 while ( isspace( *p ) )
 303                                         p++;
 304
 305                                 /* check if "dB" is given after number */
 306                                 if ( strncaseeq( p, "db", 2 ) )
 307                                         gain = DBFSTOAMP( gain );
 308
 309                                 // If there is an end adjust gain to the range
 310                                 if ( mlt_properties_get( filter_props, "end" ) != NULL )
 311                                 {
 312                                         // Determine the time position of this frame in the transition duration
 313                                         mlt_position in = mlt_filter_get_in( this );
 314                                         mlt_position out = mlt_filter_get_out( this );
 315                                         mlt_position time = mlt_frame_get_position( frame );
 316                                         double position = ( double )( time - in ) / ( double )( out - in + 1 );
 317
 318                                         double end = -1;
 319                                         char *p = mlt_properties_get( filter_props, "end" );
 320                                         if ( strcmp( p, "" ) != 0 )
 321                                                 end = fabs( strtod( p, &p) );
 322
 323                                         while ( isspace( *p ) )
 324                                                 p++;
 325
 326                                         /* check if "dB" is given after number */
 327                                         if ( strncaseeq( p, "db", 2 ) )
 328                                                 end = DBFSTOAMP( gain );
 329
 330                                         if ( end != -1 )
 331                                                 gain += ( end - gain ) * position;
 332                                 }
 333                         }
 334                 }
 335                 mlt_properties_set_double( properties, "volume.gain", gain );
 336         }
 337
 338         // Parse the maximum gain property
 339         if ( mlt_properties_get( filter_props, "max_gain" ) != NULL )
 340         {
 341                 char *p = mlt_properties_get( filter_props, "max_gain" );
 342                 double gain = fabs( strtod( p, &p) ); // 0 = no max
 343
 344                 while ( isspace( *p ) )
 345                         p++;
 346
 347                 /* check if "dB" is given after number */
 348                 if ( strncaseeq( p, "db", 2 ) )
 349                         gain = DBFSTOAMP( gain );
 350
 351                 mlt_properties_set_double( properties, "volume.max_gain", gain );
 352         }
 353
 354         // Parse the limiter property
 355         if ( mlt_properties_get( filter_props, "limiter" ) != NULL )
 356         {
 357                 char *p = mlt_properties_get( filter_props, "limiter" );
 358                 double level = 0.5; /* -6dBFS */
 359                 if ( strcmp( p, "" ) != 0 )
 360                         level = strtod( p, &p);
 361
 362                 while ( isspace( *p ) )
 363                         p++;
 364
 365                 /* check if "dB" is given after number */
 366                 if ( strncaseeq( p, "db", 2 ) )
 367                 {
 368                         if ( level > 0 )
 369                                 level = -level;
 370                         level = DBFSTOAMP( level );
 371                 }
 372                 else
 373                 {
 374                         if ( level < 0 )
 375                                 level = -level;
 376                 }
 377                 mlt_properties_set_double( properties, "volume.limiter", level );
 378         }
 379
 380         // Parse the normalise property
 381         if ( mlt_properties_get( filter_props, "normalise" ) != NULL )
 382         {
 383                 char *p = mlt_properties_get( filter_props, "normalise" );
 384                 double amplitude = 0.2511886431509580; /* -12dBFS */
 385                 if ( strcmp( p, "" ) != 0 )
 386                         amplitude = strtod( p, &p);
 387
 388                 while ( isspace( *p ) )
 389                         p++;
 390
 391                 /* check if "dB" is given after number */
 392                 if ( strncaseeq( p, "db", 2 ) )
 393                 {
 394                         if ( amplitude > 0 )
 395                                 amplitude = -amplitude;
 396                         amplitude = DBFSTOAMP( amplitude );
 397                 }
 398                 else
 399                 {
 400                         if ( amplitude < 0 )
 401                                 amplitude = -amplitude;
 402                         if ( amplitude > 1.0 )
 403                                 amplitude = 1.0;
 404                 }
 405
 406                 // If there is an end adjust gain to the range
 407                 if ( mlt_properties_get( filter_props, "end" ) != NULL )
 408                 {
 409                         // Determine the time position of this frame in the transition duration
 410                         mlt_position in = mlt_filter_get_in( this );
 411                         mlt_position out = mlt_filter_get_out( this );
 412                         mlt_position time = mlt_frame_get_position( frame );
 413                         double position = ( double )( time - in ) / ( double )( out - in + 1 );
 414                         amplitude *= position;
 415                 }
 416                 mlt_properties_set_int( properties, "volume.normalise", 1 );
 417                 mlt_properties_set_double( properties, "volume.amplitude", amplitude );
 418         }
 419
 420         // Parse the window property and allocate smoothing buffer if needed
 421         int window = mlt_properties_get_int( filter_props, "window" );
 422         if ( mlt_properties_get( filter_props, "smooth_buffer" ) == NULL && window > 1 )
 423         {
 424                 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
 425                 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
 426                 int i;
 427                 for ( i = 0; i < window; i++ )
 428                         smooth_buffer[ i ] = -1.0;
 429                 mlt_properties_set_data( filter_props, "smooth_buffer", smooth_buffer, 0, free, NULL );
 430         }
 431
 432         // Put a filter reference onto the frame
 433         mlt_properties_set_data( properties, "filter_volume", this, 0, NULL, NULL );
 434
 435         // Override the get_audio method
 436         mlt_frame_push_audio( frame, filter_get_audio );
 437
 438         return frame;
 439 }
 440
 441 /** Constructor for the filter.
 442 */
 443
 444 mlt_filter filter_volume_init( char *arg )
 445 {
 446         mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 );
 447         if ( this != NULL && mlt_filter_init( this, NULL ) == 0 )
 448         {
 449                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
 450                 this->process = filter_process;
 451                 if ( arg != NULL )
 452                         mlt_properties_set( properties, "gain", arg );
 453
 454                 mlt_properties_set_int( properties, "window", 75 );
 455                 mlt_properties_set( properties, "max_gain", "20dB" );
 456         }
 457         return this;
 458 }