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>
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.
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.
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
21 #include <framework/mlt_filter.h>
22 #include <framework/mlt_frame.h>
23 #include <framework/mlt_tokeniser.h>
30 // TODO: does not support multiple effects with SoX v14.1.0+
34 # define ST_EOF SOX_EOF
35 # define ST_SUCCESS SOX_SUCCESS
36 # define st_sample_t sox_sample_t
37 # define eff_t sox_effect_t*
38 # define ST_LIB_VERSION_CODE SOX_LIB_VERSION_CODE
39 # define ST_LIB_VERSION SOX_LIB_VERSION
40 # if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,2,0))
41 # define st_size_t size_t
43 # define st_size_t sox_size_t
45 # define ST_SIGNED_WORD_TO_SAMPLE(d,clips) SOX_SIGNED_16BIT_TO_SAMPLE(d,clips)
46 # if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
47 # define ST_SSIZE_MIN SOX_SAMPLE_MIN
49 # define ST_SSIZE_MIN SOX_SSIZE_MIN
51 # define ST_SAMPLE_TO_SIGNED_WORD(d,clips) SOX_SAMPLE_TO_SIGNED_16BIT(d,clips)
56 #define BUFFER_LEN 8192
57 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
58 #define AMPLITUDE_MIN 0.00001
60 /** Compute the mean of a set of doubles skipping unset values flagged as -1
62 static inline double mean( double *buf, int count )
68 for ( i = 0; i < count; i++ )
70 if ( buf[ i ] != -1.0 )
82 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
83 static void delete_effect( eff_t effp )
86 free( (void*)effp->in_encoding );
91 /** Create an effect state instance for a channels
93 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
95 mlt_tokeniser tokeniser = mlt_tokeniser_init();
99 // Tokenise the effect specification
100 mlt_tokeniser_parse_new( tokeniser, value, " " );
101 if ( tokeniser->count < 1 )
105 mlt_destructor effect_destructor = mlt_pool_release;
107 //fprintf(stderr, "%s: effect %s count %d\n", __FUNCTION__, tokeniser->tokens[0], tokeniser->count );
108 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
109 sox_effect_handler_t const *eff_handle = sox_find_effect( tokeniser->tokens[0] );
110 if (eff_handle == NULL ) return error;
111 eff_t eff = sox_create_effect( eff_handle );
112 effect_destructor = ( mlt_destructor ) delete_effect;
113 sox_encodinginfo_t *enc = calloc( 1, sizeof( sox_encodinginfo_t ) );
114 enc->encoding = SOX_ENCODING_SIGN2;
115 enc->bits_per_sample = 16;
116 eff->in_encoding = eff->out_encoding = enc;
118 eff_t eff = mlt_pool_alloc( sizeof( sox_effect_t ) );
119 sox_create_effect( eff, sox_find_effect( tokeniser->tokens[0] ) );
121 int opt_count = tokeniser->count - 1;
123 eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
124 int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
128 if ( opt_count != ST_EOF )
130 // Supply the effect parameters
132 if ( ( * eff->handler.getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count > 1 ? 1 : 0 ] ) == ST_SUCCESS )
134 if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
137 // Set the sox signal parameters
138 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
139 eff->in_signal.rate = frequency;
140 eff->out_signal.rate = frequency;
141 eff->in_signal.channels = 1;
142 eff->out_signal.channels = 1;
143 eff->in_signal.precision = 16;
144 eff->out_signal.precision = 16;
145 eff->in_signal.length = 0;
146 eff->out_signal.length = 0;
148 eff->ininfo.rate = frequency;
149 eff->outinfo.rate = frequency;
150 eff->ininfo.channels = 1;
151 eff->outinfo.channels = 1;
156 if ( ( * eff->handler.start )( eff ) == ST_SUCCESS )
158 if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
162 sprintf( id, "_effect_%d_%d", count, channel );
164 // Save the effect state
165 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, effect_destructor, NULL );
170 // Some error occurred so delete the temp effect state
172 effect_destructor( eff );
174 mlt_tokeniser_close( tokeniser );
182 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
184 // Get the properties of the frame
185 mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
187 // Get the filter service
188 mlt_filter filter = mlt_frame_pop_audio( frame );
190 // Get the filter properties
191 mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
193 // Get the properties
194 st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
195 st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
196 int channels_avail = *channels;
198 int count = mlt_properties_get_int( filter_properties, "_effect_count" );
200 // Get the producer's audio
201 mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
203 // Duplicate channels as necessary
204 if ( channels_avail < *channels )
206 int size = *channels * *samples * sizeof( int16_t );
207 int16_t *new_buffer = mlt_pool_alloc( size );
210 // Duplicate the existing channels
211 for ( i = 0; i < *samples; i++ )
213 for ( j = 0; j < *channels; j++ )
215 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
216 k = ( k + 1 ) % channels_avail;
220 // Update the audio buffer now - destroys the old
221 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
223 *buffer = new_buffer;
225 else if ( channels_avail == 6 && *channels == 2 )
227 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
228 int size = *channels * *samples * sizeof( int16_t );
229 int16_t *new_buffer = mlt_pool_alloc( size );
231 // Drop all but the first *channels
232 for ( i = 0; i < *samples; i++ )
234 new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
235 new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
238 // Update the audio buffer now - destroys the old
239 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
241 *buffer = new_buffer;
244 // Even though some effects are multi-channel aware, it is not reliable
245 // We must maintain a separate effect state for each channel
246 for ( i = 0; i < *channels; i++ )
249 sprintf( id, "_effect_0_%d", i );
251 // Get an existing effect state
252 eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
254 // Validate the existing effect state
255 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(14,1,0))
256 if ( e != NULL && ( e->in_signal.rate != *frequency ||
257 e->out_signal.rate != *frequency ) )
259 if ( e != NULL && ( e->ininfo.rate != *frequency ||
260 e->outinfo.rate != *frequency ) )
264 // (Re)Create the effect state
272 // Loop over all properties
273 for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
275 // Get the name of this property
276 char *name = mlt_properties_get_name( filter_properties, j );
278 // If the name does not contain a . and matches effect
279 if ( !strncmp( name, "effect", 6 ) )
281 // Get the effect specification
282 char *value = mlt_properties_get( filter_properties, name );
284 // Create an instance
285 if ( create_effect( filter, value, count, i, *frequency ) == 0 )
290 // Save the number of filters
291 mlt_properties_set_int( filter_properties, "_effect_count", count );
294 if ( *samples > 0 && count > 0 )
296 st_sample_t *p = input_buffer;
297 st_sample_t *end = p + *samples;
298 int16_t *q = *buffer + i;
299 st_size_t isamp = *samples;
300 st_size_t osamp = *samples;
303 char *normalise = mlt_properties_get( filter_properties, "normalise" );
304 double normalised_gain = 1.0;
305 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
306 st_sample_t dummy_clipped_count = 0;
309 // Convert to sox encoding
312 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
313 *p = ST_SIGNED_WORD_TO_SAMPLE( *q, dummy_clipped_count );
315 *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
317 // Compute rms amplitude while we are accessing each sample
318 rms += ( double )*p * ( double )*p;
324 // Compute final rms amplitude
325 rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
329 int window = mlt_properties_get_int( filter_properties, "window" );
330 double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
331 double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
333 // Default the maximum gain factor to 20dBFS
337 // The smoothing buffer prevents radical shifts in the gain level
338 if ( window > 0 && smooth_buffer != NULL )
340 int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
341 smooth_buffer[ smooth_index ] = rms;
343 // Ignore very small values that adversely affect the mean
344 if ( rms > AMPLITUDE_MIN )
345 mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
347 // Smoothing is really just a mean over the past N values
348 normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
352 // Determine gain to apply as current amplitude
353 normalised_gain = AMPLITUDE_NORM / rms;
356 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
358 // Govern the maximum gain
359 if ( normalised_gain > max_gain )
360 normalised_gain = max_gain;
364 for ( j = 0; j < count; j++ )
366 sprintf( id, "_effect_%d_%d", j, i );
367 e = mlt_properties_get_data( filter_properties, id, NULL );
369 // We better have this guy
372 float saved_gain = 1.0;
374 // XXX: hack to apply the normalised gain level to the vol effect
376 if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
378 if ( normalise && strcmp( e->name, "vol" ) == 0 )
381 float *f = ( float * )( e->priv );
383 *f = saved_gain * normalised_gain;
388 if ( ( * e->handler.flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
390 if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
393 // Swap input and output buffer pointers for subsequent effects
395 input_buffer = output_buffer;
399 // XXX: hack to restore the original vol gain to prevent accumulation
401 if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
403 if ( normalise && strcmp( e->name, "vol" ) == 0 )
406 float *f = ( float * )( e->priv );
412 // Convert back to signed 16bit
418 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
419 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++, dummy_clipped_count );
421 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
431 /** Filter processing.
434 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
436 if ( mlt_frame_is_test_audio( frame ) == 0 )
438 // Add the filter to the frame
439 mlt_frame_push_audio( frame, this );
440 mlt_frame_push_audio( frame, filter_get_audio );
442 // Parse the window property and allocate smoothing buffer if needed
443 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
444 int window = mlt_properties_get_int( properties, "window" );
445 if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
447 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
448 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
450 for ( i = 0; i < window; i++ )
451 smooth_buffer[ i ] = -1.0;
452 mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
459 /** Constructor for the filter.
462 mlt_filter filter_sox_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
464 mlt_filter this = mlt_filter_new( );
467 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
468 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
469 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
471 this->process = filter_process;
474 mlt_properties_set( properties, "effect", arg );
475 mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
476 mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
477 mlt_properties_set_int( properties, "window", 75 );
482 // What to do when a libst internal failure occurs
485 // Is there a build problem with my sox-devel package?
487 void gsm_create(void){}
490 void gsm_decode(void){}
493 void gsm_encode(void){}
496 void gsm_destroy(void){}
499 void gsm_option(void){}