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>
32 # define ST_EOF SOX_EOF
33 # define ST_SUCCESS SOX_SUCCESS
34 # define st_sample_t sox_sample_t
35 # define eff_t sox_effect_t*
36 # define st_size_t sox_size_t
37 # define ST_LIB_VERSION_CODE SOX_LIB_VERSION_CODE
38 # define ST_LIB_VERSION SOX_LIB_VERSION
39 # define ST_SIGNED_WORD_TO_SAMPLE(d,clips) SOX_SIGNED_16BIT_TO_SAMPLE(d,clips)
40 # define ST_SSIZE_MIN SOX_SSIZE_MIN
41 # define ST_SAMPLE_TO_SIGNED_WORD(d,clips) SOX_SAMPLE_TO_SIGNED_16BIT(d,clips)
46 #define BUFFER_LEN 8192
47 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
48 #define AMPLITUDE_MIN 0.00001
50 /** Compute the mean of a set of doubles skipping unset values flagged as -1
52 static inline double mean( double *buf, int count )
58 for ( i = 0; i < count; i++ )
60 if ( buf[ i ] != -1.0 )
72 /** Create an effect state instance for a channels
74 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
76 mlt_tokeniser tokeniser = mlt_tokeniser_init();
78 eff_t eff = mlt_pool_alloc( sizeof( sox_effect_t ) );
80 eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
85 // Tokenise the effect specification
86 mlt_tokeniser_parse_new( tokeniser, value, " " );
87 if ( tokeniser->count < 1 )
92 //fprintf(stderr, "%s: effect %s count %d\n", __FUNCTION__, tokeniser->tokens[0], tokeniser->count );
93 sox_create_effect( eff, sox_find_effect( tokeniser->tokens[0] ) );
94 int opt_count = tokeniser->count - 1;
96 int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
100 if ( opt_count != ST_EOF )
102 // Supply the effect parameters
104 if ( ( * eff->handler.getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count > 1 ? 1 : 0 ] ) == ST_SUCCESS )
106 if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
109 // Set the sox signal parameters
110 eff->ininfo.rate = frequency;
111 eff->outinfo.rate = frequency;
112 eff->ininfo.channels = 1;
113 eff->outinfo.channels = 1;
117 if ( ( * eff->handler.start )( eff ) == ST_SUCCESS )
119 if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
123 sprintf( id, "_effect_%d_%d", count, channel );
125 // Save the effect state
126 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, mlt_pool_release, NULL );
131 // Some error occurred so delete the temp effect state
133 mlt_pool_release( eff );
135 mlt_tokeniser_close( tokeniser );
143 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
145 // Get the properties of the frame
146 mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
148 // Get the filter service
149 mlt_filter filter = mlt_frame_pop_audio( frame );
151 // Get the filter properties
152 mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );
154 // Get the properties
155 st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
156 st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
157 int channels_avail = *channels;
159 int count = mlt_properties_get_int( filter_properties, "_effect_count" );
161 // Get the producer's audio
162 mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
164 // Duplicate channels as necessary
165 if ( channels_avail < *channels )
167 int size = *channels * *samples * sizeof( int16_t );
168 int16_t *new_buffer = mlt_pool_alloc( size );
171 // Duplicate the existing channels
172 for ( i = 0; i < *samples; i++ )
174 for ( j = 0; j < *channels; j++ )
176 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
177 k = ( k + 1 ) % channels_avail;
181 // Update the audio buffer now - destroys the old
182 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
184 *buffer = new_buffer;
186 else if ( channels_avail == 6 && *channels == 2 )
188 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
189 int size = *channels * *samples * sizeof( int16_t );
190 int16_t *new_buffer = mlt_pool_alloc( size );
192 // Drop all but the first *channels
193 for ( i = 0; i < *samples; i++ )
195 new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
196 new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
199 // Update the audio buffer now - destroys the old
200 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
202 *buffer = new_buffer;
205 // Even though some effects are multi-channel aware, it is not reliable
206 // We must maintain a separate effect state for each channel
207 for ( i = 0; i < *channels; i++ )
210 sprintf( id, "_effect_0_%d", i );
212 // Get an existing effect state
213 eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
215 // Validate the existing effect state
216 if ( e != NULL && ( e->ininfo.rate != *frequency ||
217 e->outinfo.rate != *frequency ) )
220 // (Re)Create the effect state
228 // Loop over all properties
229 for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
231 // Get the name of this property
232 char *name = mlt_properties_get_name( filter_properties, j );
234 // If the name does not contain a . and matches effect
235 if ( !strncmp( name, "effect", 6 ) )
237 // Get the effect specification
238 char *value = mlt_properties_get( filter_properties, name );
240 // Create an instance
241 if ( create_effect( filter, value, count, i, *frequency ) == 0 )
246 // Save the number of filters
247 mlt_properties_set_int( filter_properties, "_effect_count", count );
250 if ( *samples > 0 && count > 0 )
252 st_sample_t *p = input_buffer;
253 st_sample_t *end = p + *samples;
254 int16_t *q = *buffer + i;
255 st_size_t isamp = *samples;
256 st_size_t osamp = *samples;
259 char *normalise = mlt_properties_get( filter_properties, "normalise" );
260 double normalised_gain = 1.0;
261 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
262 st_sample_t dummy_clipped_count = 0;
265 // Convert to sox encoding
268 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
269 *p = ST_SIGNED_WORD_TO_SAMPLE( *q, dummy_clipped_count );
271 *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
273 // Compute rms amplitude while we are accessing each sample
274 rms += ( double )*p * ( double )*p;
280 // Compute final rms amplitude
281 rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
285 int window = mlt_properties_get_int( filter_properties, "window" );
286 double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
287 double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
289 // Default the maximum gain factor to 20dBFS
293 // The smoothing buffer prevents radical shifts in the gain level
294 if ( window > 0 && smooth_buffer != NULL )
296 int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
297 smooth_buffer[ smooth_index ] = rms;
299 // Ignore very small values that adversely affect the mean
300 if ( rms > AMPLITUDE_MIN )
301 mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
303 // Smoothing is really just a mean over the past N values
304 normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
308 // Determine gain to apply as current amplitude
309 normalised_gain = AMPLITUDE_NORM / rms;
312 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
314 // Govern the maximum gain
315 if ( normalised_gain > max_gain )
316 normalised_gain = max_gain;
320 for ( j = 0; j < count; j++ )
322 sprintf( id, "_effect_%d_%d", j, i );
323 e = mlt_properties_get_data( filter_properties, id, NULL );
325 // We better have this guy
328 float saved_gain = 1.0;
330 // XXX: hack to apply the normalised gain level to the vol effect
332 if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
334 if ( normalise && strcmp( e->name, "vol" ) == 0 )
337 float *f = ( float * )( e->priv );
339 *f = saved_gain * normalised_gain;
344 if ( ( * e->handler.flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
346 if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
349 // Swap input and output buffer pointers for subsequent effects
351 input_buffer = output_buffer;
355 // XXX: hack to restore the original vol gain to prevent accumulation
357 if ( normalise && strcmp( e->handler.name, "vol" ) == 0 )
359 if ( normalise && strcmp( e->name, "vol" ) == 0 )
362 float *f = ( float * )( e->priv );
368 // Convert back to signed 16bit
374 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
375 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++, dummy_clipped_count );
377 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
387 /** Filter processing.
390 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
392 if ( mlt_frame_is_test_audio( frame ) == 0 )
394 // Add the filter to the frame
395 mlt_frame_push_audio( frame, this );
396 mlt_frame_push_audio( frame, filter_get_audio );
398 // Parse the window property and allocate smoothing buffer if needed
399 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
400 int window = mlt_properties_get_int( properties, "window" );
401 if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
403 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
404 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
406 for ( i = 0; i < window; i++ )
407 smooth_buffer[ i ] = -1.0;
408 mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
415 /** Constructor for the filter.
418 mlt_filter filter_sox_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
420 mlt_filter this = mlt_filter_new( );
423 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
424 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
425 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
427 this->process = filter_process;
430 mlt_properties_set( properties, "effect", arg );
431 mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
432 mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
433 mlt_properties_set_int( properties, "window", 75 );
438 // What to do when a libst internal failure occurs
441 // Is there a build problem with my sox-devel package?
443 void gsm_create(void){}
446 void gsm_decode(void){}
449 void gsm_encode(void){}
452 void gsm_destroy(void){}
455 void gsm_option(void){}