git.sesse.net Git - mlt/blob - src/modules/sox/filter_sox.c

   1 /*
   2  * filter_sox.c -- apply any number of SOX effects using libst
   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_sox.h"
  22
  23 #include <framework/mlt_frame.h>
  24 #include <framework/mlt_tokeniser.h>
  25
  26 #include <stdio.h>
  27 #include <stdlib.h>
  28 #include <string.h>
  29 #include <math.h>
  30
  31 #include <st.h>
  32
  33 #define BUFFER_LEN 8192
  34 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
  35 #define AMPLITUDE_MIN 0.00001
  36
  37 /** Compute the mean of a set of doubles skipping unset values flagged as -1
  38 */
  39 static inline double mean( double *buf, int count )
  40 {
  41         double mean = 0;
  42         int i;
  43         int j = 0;
  44
  45         for ( i = 0; i < count; i++ )
  46         {
  47                 if ( buf[ i ] != -1.0 )
  48                 {
  49                         mean += buf[ i ];
  50                         j ++;
  51                 }
  52         }
  53         if ( j > 0 )
  54                 mean /= j;
  55
  56         return mean;
  57 }
  58
  59 /** Create an effect state instance for a channels
  60 */
  61 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
  62 {
  63         mlt_tokeniser tokeniser = mlt_tokeniser_init();
  64         eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
  65         char id[ 256 ];
  66         int error = 1;
  67
  68         // Tokenise the effect specification
  69         mlt_tokeniser_parse_new( tokeniser, value, " " );
  70
  71         // Locate the effect
  72         int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
  73
  74         // If valid effect
  75         if ( opt_count != ST_EOF )
  76         {
  77                 // Supply the effect parameters
  78                 if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
  79                 {
  80                         // Set the sox signal parameters
  81                         eff->ininfo.rate = frequency;
  82                         eff->outinfo.rate = frequency;
  83                         eff->ininfo.channels = 1;
  84                         eff->outinfo.channels = 1;
  85
  86                         // Start the effect
  87                         if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
  88                         {
  89                                 // Construct id
  90                                 sprintf( id, "_effect_%d_%d", count, channel );
  91
  92                                 // Save the effect state
  93                                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, mlt_pool_release, NULL );
  94                                 error = 0;
  95                         }
  96                 }
  97         }
  98         // Some error occurred so delete the temp effect state
  99         if ( error == 1 )
 100                 mlt_pool_release( eff );
 101
 102         mlt_tokeniser_close( tokeniser );
 103
 104         return error;
 105 }
 106
 107 /** Get the audio.
 108 */
 109
 110 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
 111 {
 112         // Get the properties of the frame
 113         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
 114
 115         // Get the filter service
 116         mlt_filter filter = mlt_frame_pop_audio( frame );
 117
 118         // Get the filter properties
 119         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
 120
 121         // Get the properties
 122         st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
 123         st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
 124         int channels_avail = *channels;
 125         int i; // channel
 126         int count = mlt_properties_get_int( filter_properties, "effect_count" );
 127
 128         // Get the producer's audio
 129         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
 130
 131         // Duplicate channels as necessary
 132         if ( channels_avail < *channels )
 133         {
 134                 int size = *channels * *samples * sizeof( int16_t );
 135                 int16_t *new_buffer = mlt_pool_alloc( size );
 136                 int j, k = 0;
 137
 138                 // Duplicate the existing channels
 139                 for ( i = 0; i < *samples; i++ )
 140                 {
 141                         for ( j = 0; j < *channels; j++ )
 142                         {
 143                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
 144                                 k = ( k + 1 ) % channels_avail;
 145                         }
 146                 }
 147
 148                 // Update the audio buffer now - destroys the old
 149                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
 150
 151                 *buffer = new_buffer;
 152         }
 153         else if ( channels_avail == 6 && *channels == 2 )
 154         {
 155                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
 156                 int size = *channels * *samples * sizeof( int16_t );
 157                 int16_t *new_buffer = mlt_pool_alloc( size );
 158
 159                 // Drop all but the first *channels
 160                 for ( i = 0; i < *samples; i++ )
 161                 {
 162                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
 163                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
 164                 }
 165
 166                 // Update the audio buffer now - destroys the old
 167                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
 168
 169                 *buffer = new_buffer;
 170         }
 171
 172         // Even though some effects are multi-channel aware, it is not reliable
 173         // We must maintain a separate effect state for each channel
 174         for ( i = 0; i < *channels; i++ )
 175         {
 176                 char id[ 256 ];
 177                 sprintf( id, "_effect_0_%d", i );
 178
 179                 // Get an existing effect state
 180                 eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
 181
 182                 // Validate the existing effect state
 183                 if ( e != NULL && ( e->ininfo.rate != *frequency ||
 184                                                         e->outinfo.rate != *frequency ) )
 185                         e = NULL;
 186
 187                 // (Re)Create the effect state
 188                 if ( e == NULL )
 189                 {
 190                         int j = 0;
 191
 192                         // Reset the count
 193                         count = 0;
 194
 195                         // Loop over all properties
 196                         for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
 197                         {
 198                                 // Get the name of this property
 199                                 char *name = mlt_properties_get_name( filter_properties, j );
 200
 201                                 // If the name does not contain a . and matches effect
 202                                 if ( !strncmp( name, "effect", 6 ) )
 203                                 {
 204                                         // Get the effect specification
 205                                         char *value = mlt_properties_get( filter_properties, name );
 206
 207                                         // Create an instance
 208                                         if ( create_effect( filter, value, count, i, *frequency ) == 0 )
 209                                                 count ++;
 210                                 }
 211                         }
 212
 213                         // Save the number of filters
 214                         mlt_properties_set_int( filter_properties, "effect_count", count );
 215
 216                 }
 217                 if ( *samples > 0 && count > 0 )
 218                 {
 219                         st_sample_t *p = input_buffer;
 220                         st_sample_t *end = p + *samples;
 221                         int16_t *q = *buffer + i;
 222                         st_size_t isamp = *samples;
 223                         st_size_t osamp = *samples;
 224                         double rms = 0;
 225                         int j;
 226                         char *normalise = mlt_properties_get( filter_properties, "normalise" );
 227                         double normalised_gain = 1.0;
 228
 229                         // Convert to sox encoding
 230                         while( p != end )
 231                         {
 232                                 *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
 233
 234                                 // Compute rms amplitude while we are accessing each sample
 235                                 rms += ( double )*p * ( double )*p;
 236
 237                                 p ++;
 238                                 q += *channels;
 239                         }
 240
 241                         // Compute final rms amplitude
 242                         rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
 243
 244                         if ( normalise )
 245                         {
 246                                 int window = mlt_properties_get_int( filter_properties, "window" );
 247                                 double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
 248                                 double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
 249
 250                                 // Default the maximum gain factor to 20dBFS
 251                                 if ( max_gain == 0 )
 252                                         max_gain = 10.0;
 253
 254                                 // The smoothing buffer prevents radical shifts in the gain level
 255                                 if ( window > 0 && smooth_buffer != NULL )
 256                                 {
 257                                         int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
 258                                         smooth_buffer[ smooth_index ] = rms;
 259
 260                                         // Ignore very small values that adversely affect the mean
 261                                         if ( rms > AMPLITUDE_MIN )
 262                                                 mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
 263
 264                                         // Smoothing is really just a mean over the past N values
 265                                         normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
 266                                 }
 267                                 else if ( rms > 0 )
 268                                 {
 269                                         // Determine gain to apply as current amplitude
 270                                         normalised_gain = AMPLITUDE_NORM / rms;
 271                                 }
 272
 273                                 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
 274
 275                                 // Govern the maximum gain
 276                                 if ( normalised_gain > max_gain )
 277                                         normalised_gain = max_gain;
 278                         }
 279
 280                         // For each effect
 281                         for ( j = 0; j < count; j++ )
 282                         {
 283                                 sprintf( id, "_effect_%d_%d", j, i );
 284                                 e = mlt_properties_get_data( filter_properties, id, NULL );
 285
 286                                 // We better have this guy
 287                                 if ( e != NULL )
 288                                 {
 289                                         float saved_gain = 1.0;
 290
 291                                         // XXX: hack to apply the normalised gain level to the vol effect
 292                                         if ( normalise && strcmp( e->name, "vol" ) == 0 )
 293                                         {
 294                                                 float *f = ( float * )( e->priv );
 295                                                 saved_gain = *f;
 296                                                 *f = saved_gain * normalised_gain;
 297                                         }
 298
 299                                         // Apply the effect
 300                                         if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
 301                                         {
 302                                                 // Swap input and output buffer pointers for subsequent effects
 303                                                 p = input_buffer;
 304                                                 input_buffer = output_buffer;
 305                                                 output_buffer = p;
 306                                         }
 307
 308                                         // XXX: hack to restore the original vol gain to prevent accumulation
 309                                         if ( normalise && strcmp( e->name, "vol" ) == 0 )
 310                                         {
 311                                                 float *f = ( float * )( e->priv );
 312                                                 *f = saved_gain;
 313                                         }
 314                                 }
 315                         }
 316
 317                         // Convert back to signed 16bit
 318                         p = input_buffer;
 319                         q = *buffer + i;
 320                         end = p + *samples;
 321                         while ( p != end )
 322                         {
 323                                 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
 324                                 q += *channels;
 325                         }
 326                 }
 327         }
 328
 329         return 0;
 330 }
 331
 332 /** Filter processing.
 333 */
 334
 335 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 336 {
 337         if ( mlt_frame_is_test_audio( frame ) != 0 )
 338         {
 339                 // Add the filter to the frame
 340                 mlt_frame_push_audio( frame, this );
 341                 mlt_frame_push_audio( frame, filter_get_audio );
 342
 343                 // Parse the window property and allocate smoothing buffer if needed
 344                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
 345                 int window = mlt_properties_get_int( properties, "window" );
 346                 if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
 347                 {
 348                         // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
 349                         double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
 350                         int i;
 351                         for ( i = 0; i < window; i++ )
 352                                 smooth_buffer[ i ] = -1.0;
 353                         mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
 354                 }
 355         }
 356
 357         return frame;
 358 }
 359
 360 /** Constructor for the filter.
 361 */
 362
 363 mlt_filter filter_sox_init( char *arg )
 364 {
 365         mlt_filter this = mlt_filter_new( );
 366         if ( this != NULL )
 367         {
 368                 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
 369                 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
 370                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
 371
 372                 this->process = filter_process;
 373
 374                 if ( arg != NULL )
 375                         mlt_properties_set( properties, "effect", arg );
 376                 mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 377                 mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
 378                 mlt_properties_set_int( properties, "window", 75 );
 379         }
 380         return this;
 381 }
 382
 383 // What to do when a libst internal failure occurs
 384 void cleanup(void){}
 385
 386 // Is there a build problem with my sox-devel package?
 387 #ifndef gsm_create
 388 void gsm_create(void){}
 389 #endif
 390 #ifndef gsm_decode
 391 void gsm_decode(void){}
 392 #endif
 393 #ifndef gdm_encode
 394 void gsm_encode(void){}
 395 #endif
 396 #ifndef gsm_destroy
 397 void gsm_destroy(void){}
 398 #endif
 399 #ifndef gsm_option
 400 void gsm_option(void){}
 401 #endif