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 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.
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.
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.
21 #include "filter_sox.h"
23 #include <framework/mlt_frame.h>
24 #include <framework/mlt_tokeniser.h>
33 #define BUFFER_LEN 8192
34 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
35 #define AMPLITUDE_MIN 0.00001
37 /** Compute the mean of a set of doubles skipping unset values flagged as -1
39 static inline double mean( double *buf, int count )
45 for ( i = 0; i < count; i++ )
47 if ( buf[ i ] != -1.0 )
59 /** Create an effect state instance for a channels
61 static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
63 mlt_tokeniser tokeniser = mlt_tokeniser_init();
64 eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
68 // Tokenise the effect specification
69 mlt_tokeniser_parse_new( tokeniser, value, " " );
72 int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
75 if ( opt_count != ST_EOF )
77 // Supply the effect parameters
78 if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
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;
87 if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
90 sprintf( id, "_effect_%d_%d", count, channel );
92 // Save the effect state
93 mlt_properties_set_data( mlt_filter_properties( this ), id, eff, 0, mlt_pool_release, NULL );
98 // Some error occurred so delete the temp effect state
100 mlt_pool_release( eff );
102 mlt_tokeniser_close( tokeniser );
110 static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
112 // Get the properties of the frame
113 mlt_properties properties = mlt_frame_properties( frame );
115 // Get the filter service
116 mlt_filter filter = mlt_frame_pop_audio( frame );
118 // Get the filter properties
119 mlt_properties filter_properties = mlt_filter_properties( filter );
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;
126 int count = mlt_properties_get_int( filter_properties, "effect_count" );
128 // Restore the original get_audio
129 frame->get_audio = mlt_frame_pop_audio( frame );
131 // Get the producer's audio
132 mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
134 // Duplicate channels as necessary
135 if ( channels_avail < *channels )
137 int size = *channels * *samples * sizeof( int16_t );
138 int16_t *new_buffer = mlt_pool_alloc( size );
141 // Duplicate the existing channels
142 for ( i = 0; i < *samples; i++ )
144 for ( j = 0; j < *channels; j++ )
146 new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
147 k = ( k + 1 ) % channels_avail;
151 // Update the audio buffer now - destroys the old
152 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
154 *buffer = new_buffer;
156 else if ( channels_avail == 6 && *channels == 2 )
158 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
159 int size = *channels * *samples * sizeof( int16_t );
160 int16_t *new_buffer = mlt_pool_alloc( size );
162 // Drop all but the first *channels
163 for ( i = 0; i < *samples; i++ )
165 new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
166 new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
169 // Update the audio buffer now - destroys the old
170 mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
172 *buffer = new_buffer;
175 // Even though some effects are multi-channel aware, it is not reliable
176 // We must maintain a separate effect state for each channel
177 for ( i = 0; i < *channels; i++ )
180 sprintf( id, "_effect_0_%d", i );
182 // Get an existing effect state
183 eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
185 // Validate the existing effect state
186 if ( e != NULL && ( e->ininfo.rate != *frequency ||
187 e->outinfo.rate != *frequency ) )
190 // (Re)Create the effect state
198 // Loop over all properties
199 for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
201 // Get the name of this property
202 char *name = mlt_properties_get_name( filter_properties, j );
204 // If the name does not contain a . and matches effect
205 if ( !strncmp( name, "effect", 6 ) )
207 // Get the effect specification
208 char *value = mlt_properties_get( filter_properties, name );
210 // Create an instance
211 if ( create_effect( filter, value, count, i, *frequency ) == 0 )
216 // Save the number of filters
217 mlt_properties_set_int( filter_properties, "effect_count", count );
220 if ( *samples > 0 && count > 0 )
222 st_sample_t *p = input_buffer;
223 st_sample_t *end = p + *samples;
224 int16_t *q = *buffer + i;
225 st_size_t isamp = *samples;
226 st_size_t osamp = *samples;
229 char *normalise = mlt_properties_get( filter_properties, "normalise" );
230 double normalised_gain = 1.0;
232 // Convert to sox encoding
235 *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
237 // Compute rms amplitude while we are accessing each sample
238 rms += ( double )*p * ( double )*p;
244 // Compute final rms amplitude
245 rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
249 int window = mlt_properties_get_int( filter_properties, "window" );
250 double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
251 double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
253 // Default the maximum gain factor to 20dBFS
257 // The smoothing buffer prevents radical shifts in the gain level
258 if ( window > 0 && smooth_buffer != NULL )
260 int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
261 smooth_buffer[ smooth_index ] = rms;
263 // Ignore very small values that adversely affect the mean
264 if ( rms > AMPLITUDE_MIN )
265 mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
267 // Smoothing is really just a mean over the past N values
268 normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
272 // Determine gain to apply as current amplitude
273 normalised_gain = AMPLITUDE_NORM / rms;
276 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
278 // Govern the maximum gain
279 if ( normalised_gain > max_gain )
280 normalised_gain = max_gain;
284 for ( j = 0; j < count; j++ )
286 sprintf( id, "_effect_%d_%d", j, i );
287 e = mlt_properties_get_data( filter_properties, id, NULL );
289 // We better have this guy
292 float saved_gain = 1.0;
294 // XXX: hack to apply the normalised gain level to the vol effect
295 if ( normalise && strcmp( e->name, "vol" ) == 0 )
297 float *f = ( float * )( e->priv );
299 *f = saved_gain * normalised_gain;
303 if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
305 // Swap input and output buffer pointers for subsequent effects
307 input_buffer = output_buffer;
311 // XXX: hack to restore the original vol gain to prevent accumulation
312 if ( normalise && strcmp( e->name, "vol" ) == 0 )
314 float *f = ( float * )( e->priv );
320 // Convert back to signed 16bit
326 *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
335 /** Filter processing.
338 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
340 if ( frame->get_audio != NULL )
342 // Add the filter to the frame
343 mlt_frame_push_audio( frame, frame->get_audio );
344 mlt_frame_push_audio( frame, this );
345 frame->get_audio = filter_get_audio;
347 // Parse the window property and allocate smoothing buffer if needed
348 mlt_properties properties = mlt_filter_properties( this );
349 int window = mlt_properties_get_int( properties, "window" );
350 if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
352 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
353 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
355 for ( i = 0; i < window; i++ )
356 smooth_buffer[ i ] = -1.0;
357 mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
364 /** Constructor for the filter.
367 mlt_filter filter_sox_init( char *arg )
369 mlt_filter this = mlt_filter_new( );
372 void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
373 void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
374 mlt_properties properties = mlt_filter_properties( this );
376 this->process = filter_process;
379 mlt_properties_set( properties, "effect", arg );
380 mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
381 mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
382 mlt_properties_set_int( properties, "window", 75 );
387 // What to do when a libst internal failure occurs
390 // Is there a build problem with my sox-devel package?
392 void gsm_create(void){}
395 void gsm_decode(void){}
398 void gsm_encode(void){}
401 void gsm_destroy(void){}
404 void gsm_option(void){}