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[mlt] / 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 library is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * This library 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 GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with this library; if not, write to the Free Software
18  * Foundation, 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 #ifdef SOX14
32 #       include <sox.h>
33 #       define ST_EOF SOX_EOF
34 #       define ST_SUCCESS SOX_SUCCESS
35 #       define st_sample_t sox_sample_t
36 #       define eff_t sox_effect_t*
37 #       define st_size_t sox_size_t
38 #       define ST_LIB_VERSION_CODE SOX_LIB_VERSION_CODE
39 #       define ST_LIB_VERSION SOX_LIB_VERSION
40 #       define ST_SIGNED_WORD_TO_SAMPLE(d,clips) SOX_SIGNED_16BIT_TO_SAMPLE(d,clips)
41 #       define ST_SSIZE_MIN SOX_SSIZE_MIN
42 #       define ST_SAMPLE_TO_SIGNED_WORD(d,clips) SOX_SAMPLE_TO_SIGNED_16BIT(d,clips)
43 #else
44 #       include <st.h>
45 #endif
46
47 #define BUFFER_LEN 8192
48 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
49 #define AMPLITUDE_MIN 0.00001
50
51 /** Compute the mean of a set of doubles skipping unset values flagged as -1
52 */
53 static inline double mean( double *buf, int count )
54 {
55         double mean = 0;
56         int i;
57         int j = 0;
58         
59         for ( i = 0; i < count; i++ )
60         {
61                 if ( buf[ i ] != -1.0 )
62                 {
63                         mean += buf[ i ];
64                         j ++;
65                 }
66         }
67         if ( j > 0 )
68                 mean /= j;
69         
70         return mean;
71 }
72
73 /** Create an effect state instance for a channels
74 */
75 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
76 {
77         mlt_tokeniser tokeniser = mlt_tokeniser_init();
78 #ifdef SOX14
79         eff_t eff = mlt_pool_alloc( sizeof( sox_effect_t ) );
80 #else
81         eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
82 #endif
83         char id[ 256 ];
84         int error = 1;
85
86         // Tokenise the effect specification
87         mlt_tokeniser_parse_new( tokeniser, value, " " );
88         if ( tokeniser->count < 1 )
89                 return error;
90
91         // Locate the effect
92 #ifdef SOX14
93         //fprintf(stderr, "%s: effect %s count %d\n", __FUNCTION__, tokeniser->tokens[0], tokeniser->count );
94         sox_create_effect( eff, sox_find_effect( tokeniser->tokens[0] ) );
95         int opt_count = tokeniser->count - 1;
96 #else
97         int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
98 #endif
99         
100         // If valid effect
101         if ( opt_count != ST_EOF )
102         {
103                 // Supply the effect parameters
104 #ifdef SOX14
105                 if ( ( * eff->handler.getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count > 1 ? 1 : 0  ] ) == ST_SUCCESS )
106 #else
107                 if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
108 #endif
109                 {
110                         // Set the sox signal parameters
111                         eff->ininfo.rate = frequency;
112                         eff->outinfo.rate = frequency;
113                         eff->ininfo.channels = 1;
114                         eff->outinfo.channels = 1;
115                         
116                         // Start the effect
117 #ifdef SOX14
118                         if ( ( * eff->handler.start )( eff ) == ST_SUCCESS )
119 #else
120                         if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
121 #endif
122                         {
123                                 // Construct id
124                                 sprintf( id, "_effect_%d_%d", count, channel );
125
126                                 // Save the effect state
127                                 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, mlt_pool_release, NULL );
128                                 error = 0;
129                         }
130                 }
131         }
132         // Some error occurred so delete the temp effect state
133         if ( error == 1 )
134                 mlt_pool_release( eff );
135         
136         mlt_tokeniser_close( tokeniser );
137         
138         return error;
139 }
140
141 /** Get the audio.
142 */
143
144 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
145 {
146         // Get the properties of the frame
147         mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
148
149         // Get the filter service
150         mlt_filter filter = mlt_frame_pop_audio( frame );
151
152         // Get the filter properties
153         mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
154
155         // Get the properties
156         st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
157         st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
158         int channels_avail = *channels;
159         int i; // channel
160         int count = mlt_properties_get_int( filter_properties, "_effect_count" );
161
162         // Get the producer's audio
163         mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
164
165         // Duplicate channels as necessary
166         if ( channels_avail < *channels )
167         {
168                 int size = *channels * *samples * sizeof( int16_t );
169                 int16_t *new_buffer = mlt_pool_alloc( size );
170                 int j, k = 0;
171                 
172                 // Duplicate the existing channels
173                 for ( i = 0; i < *samples; i++ )
174                 {
175                         for ( j = 0; j < *channels; j++ )
176                         {
177                                 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
178                                 k = ( k + 1 ) % channels_avail;
179                         }
180                 }
181                 
182                 // Update the audio buffer now - destroys the old
183                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
184                 
185                 *buffer = new_buffer;
186         }
187         else if ( channels_avail == 6 && *channels == 2 )
188         {
189                 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
190                 int size = *channels * *samples * sizeof( int16_t );
191                 int16_t *new_buffer = mlt_pool_alloc( size );
192                 
193                 // Drop all but the first *channels
194                 for ( i = 0; i < *samples; i++ )
195                 {
196                         new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
197                         new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
198                 }
199
200                 // Update the audio buffer now - destroys the old
201                 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
202                 
203                 *buffer = new_buffer;
204         }
205
206         // Even though some effects are multi-channel aware, it is not reliable
207         // We must maintain a separate effect state for each channel
208         for ( i = 0; i < *channels; i++ )
209         {
210                 char id[ 256 ];
211                 sprintf( id, "_effect_0_%d", i );
212                 
213                 // Get an existing effect state
214                 eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
215                 
216                 // Validate the existing effect state
217                 if ( e != NULL && ( e->ininfo.rate != *frequency || 
218                                                         e->outinfo.rate != *frequency ) )
219                         e = NULL;
220                 
221                 // (Re)Create the effect state
222                 if ( e == NULL )
223                 {
224                         int j = 0;
225                         
226                         // Reset the count
227                         count = 0;
228         
229                         // Loop over all properties
230                         for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
231                         {
232                                 // Get the name of this property
233                                 char *name = mlt_properties_get_name( filter_properties, j );
234         
235                                 // If the name does not contain a . and matches effect
236                                 if ( !strncmp( name, "effect", 6 ) )
237                                 {
238                                         // Get the effect specification
239                                         char *value = mlt_properties_get( filter_properties, name );
240         
241                                         // Create an instance
242                                         if ( create_effect( filter, value, count, i, *frequency ) == 0 )
243                                                 count ++;
244                                 }
245                         }
246                         
247                         // Save the number of filters
248                         mlt_properties_set_int( filter_properties, "_effect_count", count );
249                         
250                 }
251                 if ( *samples > 0 && count > 0 )
252                 {
253                         st_sample_t *p = input_buffer;
254                         st_sample_t *end = p + *samples;
255                         int16_t *q = *buffer + i;
256                         st_size_t isamp = *samples;
257                         st_size_t osamp = *samples;
258                         double rms = 0;
259                         int j;
260                         char *normalise = mlt_properties_get( filter_properties, "normalise" );
261                         double normalised_gain = 1.0;
262 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
263                         st_sample_t dummy_clipped_count = 0;
264 #endif
265                         
266                         // Convert to sox encoding
267                         while( p != end )
268                         {
269 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
270                                 *p = ST_SIGNED_WORD_TO_SAMPLE( *q, dummy_clipped_count );
271 #else
272                                 *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
273 #endif
274                                 // Compute rms amplitude while we are accessing each sample
275                                 rms += ( double )*p * ( double )*p;
276                                 
277                                 p ++;
278                                 q += *channels;
279                         }
280                         
281                         // Compute final rms amplitude
282                         rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
283                         
284                         if ( normalise )
285                         {
286                                 int window = mlt_properties_get_int( filter_properties, "window" );
287                                 double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
288                                 double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
289                                 
290                                 // Default the maximum gain factor to 20dBFS
291                                 if ( max_gain == 0 )
292                                         max_gain = 10.0;
293                                 
294                                 // The smoothing buffer prevents radical shifts in the gain level
295                                 if ( window > 0 && smooth_buffer != NULL )
296                                 {
297                                         int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
298                                         smooth_buffer[ smooth_index ] = rms;
299                                         
300                                         // Ignore very small values that adversely affect the mean
301                                         if ( rms > AMPLITUDE_MIN )
302                                                 mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
303                                         
304                                         // Smoothing is really just a mean over the past N values
305                                         normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
306                                 }
307                                 else if ( rms > 0 )
308                                 {
309                                         // Determine gain to apply as current amplitude
310                                         normalised_gain = AMPLITUDE_NORM / rms;
311                                 }
312                                         
313                                 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
314                                 
315                                 // Govern the maximum gain
316                                 if ( normalised_gain > max_gain )
317                                         normalised_gain = max_gain;
318                         }
319                         
320                         // For each effect
321                         for ( j = 0; j < count; j++ )
322                         {
323                                 sprintf( id, "_effect_%d_%d", j, i );
324                                 e = mlt_properties_get_data( filter_properties, id, NULL );
325                                 
326                                 // We better have this guy
327                                 if ( e != NULL )
328                                 {
329                                         float saved_gain = 1.0;
330                                         
331                                         // XXX: hack to apply the normalised gain level to the vol effect
332 #ifdef SOX14
333                                         if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
334 #else
335                                         if ( normalise && strcmp( e->name, "vol" ) == 0 )
336 #endif
337                                         {
338                                                 float *f = ( float * )( e->priv );
339                                                 saved_gain = *f;
340                                                 *f = saved_gain * normalised_gain;
341                                         }
342                                         
343                                         // Apply the effect
344 #ifdef SOX14
345                                         if ( ( * e->handler.flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
346 #else
347                                         if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
348 #endif
349                                         {
350                                                 // Swap input and output buffer pointers for subsequent effects
351                                                 p = input_buffer;
352                                                 input_buffer = output_buffer;
353                                                 output_buffer = p;
354                                         }
355                                         
356                                         // XXX: hack to restore the original vol gain to prevent accumulation
357 #ifdef SOX14
358                                         if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
359 #else
360                                         if ( normalise && strcmp( e->name, "vol" ) == 0 )
361 #endif
362                                         {
363                                                 float *f = ( float * )( e->priv );
364                                                 *f = saved_gain;
365                                         }
366                                 }
367                         }
368                         
369                         // Convert back to signed 16bit
370                         p = input_buffer;
371                         q = *buffer + i;
372                         end = p + *samples;
373                         while ( p != end )
374                         {
375 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
376                                 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++, dummy_clipped_count );
377 #else
378                                 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
379 #endif
380                                 q += *channels;
381                         }
382                 }
383         }
384
385         return 0;
386 }
387
388 /** Filter processing.
389 */
390
391 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
392 {
393         if ( mlt_frame_is_test_audio( frame ) == 0 )
394         {
395                 // Add the filter to the frame
396                 mlt_frame_push_audio( frame, this );
397                 mlt_frame_push_audio( frame, filter_get_audio );
398                 
399                 // Parse the window property and allocate smoothing buffer if needed
400                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
401                 int window = mlt_properties_get_int( properties, "window" );
402                 if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
403                 {
404                         // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
405                         double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
406                         int i;
407                         for ( i = 0; i < window; i++ )
408                                 smooth_buffer[ i ] = -1.0;
409                         mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
410                 }
411         }
412
413         return frame;
414 }
415
416 /** Constructor for the filter.
417 */
418
419 mlt_filter filter_sox_init( char *arg )
420 {
421         mlt_filter this = mlt_filter_new( );
422         if ( this != NULL )
423         {
424                 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
425                 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
426                 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
427                 
428                 this->process = filter_process;
429                 
430                 if ( arg != NULL )
431                         mlt_properties_set( properties, "effect", arg );
432                 mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
433                 mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
434                 mlt_properties_set_int( properties, "window", 75 );
435         }
436         return this;
437 }
438
439 // What to do when a libst internal failure occurs
440 void cleanup(void){}
441
442 // Is there a build problem with my sox-devel package?
443 #ifndef gsm_create
444 void gsm_create(void){}
445 #endif
446 #ifndef gsm_decode
447 void gsm_decode(void){}
448 #endif
449 #ifndef gdm_encode
450 void gsm_encode(void){}
451 #endif
452 #ifndef gsm_destroy
453 void gsm_destroy(void){}
454 #endif
455 #ifndef gsm_option
456 void gsm_option(void){}
457 #endif