2 * filter_volume.c -- adjust audio volume
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 <framework/mlt_filter.h>
22 #include <framework/mlt_frame.h>
30 #define MAX_CHANNELS 6
31 #define EPSILON 0.00001
33 /* The following normalise functions come from the normalize utility:
34 Copyright (C) 1999--2002 Chris Vaill */
39 # define ROUND(x) floor((x) + 0.5)
42 #define DBFSTOAMP(x) pow(10,(x)/20.0)
44 /** Return nonzero if the two strings are equal, ignoring case, up to
45 the first n characters.
47 int strncaseeq(const char *s1, const char *s2, size_t n)
51 if (tolower(*s1++) != tolower(*s2++))
59 / tanh((x + lev) / (1-lev)) * (1-lev) - lev (for x < -lev)
61 x' = | x (for |x| <= lev)
63 \ tanh((x - lev) / (1-lev)) * (1-lev) + lev (for x > lev)
65 With limiter level = 0, this is equivalent to a tanh() function;
66 with limiter level = 1, this is equivalent to clipping.
68 static inline double limiter( double x, double lmtr_lvl )
73 xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl;
74 else if (x > lmtr_lvl)
75 xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl;
78 // fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp );
84 /** Takes a full smoothing window, and returns the value of the center
87 Currently, just does a mean filter, but we could do a median or
88 gaussian filter here instead.
90 static inline double get_smoothed_data( double *buf, int count )
95 for ( i = 0, j = 0; i < count; i++ )
97 if ( buf[ i ] != -1.0 )
103 if (j) smoothed /= j;
104 // fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed );
109 /** Get the max power level (using RMS) and peak level of the audio segment.
111 double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *peak )
113 // Determine numeric limits
114 int bytes_per_samp = (samp_width - 1) / 8 + 1;
115 int16_t max = (1 << (bytes_per_samp * 8 - 1)) - 1;
116 int16_t min = -max - 1;
118 double *sums = (double *) calloc( channels, sizeof(double) );
121 double pow, maxpow = 0;
123 /* initialize peaks to effectively -inf and +inf */
124 int16_t max_sample = min;
125 int16_t min_sample = max;
127 for ( i = 0; i < samples; i++ )
129 for ( c = 0; c < channels; c++ )
132 sums[ c ] += (double) sample * (double) sample;
135 if ( sample > max_sample )
137 else if ( sample < min_sample )
141 for ( c = 0; c < channels; c++ )
143 pow = sums[ c ] / (double) samples;
150 /* scale the pow value to be in the range 0.0 -- 1.0 */
151 maxpow /= ( (double) min * (double) min);
153 if ( -min_sample > max_sample )
154 *peak = min_sample / (double) min;
156 *peak = max_sample / (double) max;
158 return sqrt( maxpow );
161 /* ------ End normalize functions --------------------------------------- */
166 static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
168 // Get the filter from the frame
169 mlt_filter this = mlt_frame_pop_audio( frame );
171 // Get the properties from the filter
172 mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
174 // Get the frame's filter instance properties
175 mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( this ) );
177 // Get the parameters
178 double gain = mlt_properties_get_double( instance_props, "gain" );
179 double max_gain = mlt_properties_get_double( instance_props, "max_gain" );
180 double limiter_level = 0.5; /* -6 dBFS */
181 int normalise = mlt_properties_get_int( instance_props, "normalise" );
182 double amplitude = mlt_properties_get_double( instance_props, "amplitude" );
187 if ( mlt_properties_get( instance_props, "limiter" ) != NULL )
188 limiter_level = mlt_properties_get_double( instance_props, "limiter" );
190 // Get the producer's audio
191 *format = mlt_audio_s16;
192 mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
193 // fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
195 // Determine numeric limits
196 int bytes_per_samp = (samp_width - 1) / 8 + 1;
197 int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
198 int samplemin = -samplemax - 1;
200 mlt_service_lock( MLT_FILTER_SERVICE( this ) );
204 int window = mlt_properties_get_int( filter_props, "window" );
205 double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
207 if ( window > 0 && smooth_buffer != NULL )
209 int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" );
211 // Compute the signal power and put into smoothing buffer
212 smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak );
213 // fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
214 if ( smooth_buffer[ smooth_index ] > EPSILON )
216 mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window );
218 // Smooth the data and compute the gain
219 // fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
220 gain *= amplitude / get_smoothed_data( smooth_buffer, window );
225 gain *= amplitude / signal_max_power( (int16_t*) *buffer, *channels, *samples, &peak );
229 // if ( gain > 1.0 && normalise )
230 // fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
232 if ( max_gain > 0 && gain > max_gain )
235 // Initialise filter's previous gain value to prevent an inadvertant jump from 0
236 mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
237 mlt_position current_position = mlt_frame_get_position( frame );
238 if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
239 || current_position != last_position + 1 )
240 mlt_properties_set_double( filter_props, "_previous_gain", gain );
242 // Start the gain out at the previous
243 double previous_gain = mlt_properties_get_double( filter_props, "_previous_gain" );
245 // Determine ramp increment
246 double gain_step = ( gain - previous_gain ) / *samples;
247 // fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
249 // Save the current gain for the next iteration
250 mlt_properties_set_double( filter_props, "_previous_gain", gain );
251 mlt_properties_set_position( filter_props, "_last_position", current_position );
253 mlt_service_unlock( MLT_FILTER_SERVICE( this ) );
255 // Ramp from the previous gain to the current
256 gain = previous_gain;
258 int16_t *p = (int16_t*) *buffer;
261 for ( i = 0; i < *samples; i++ )
263 for ( j = 0; j < *channels; j++ )
266 *p = ROUND( sample );
270 /* use limiter function instead of clipping */
272 *p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
274 /* perform clipping */
275 else if ( sample > samplemax )
277 else if ( sample < samplemin )
288 /** Filter processing.
291 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
293 mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
294 mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( this ) );
296 double gain = 1.0; // no adjustment
298 // Parse the gain property
299 if ( mlt_properties_get( filter_props, "gain" ) != NULL )
301 char *p = mlt_properties_get( filter_props, "gain" );
303 if ( strncaseeq( p, "normalise", 9 ) )
304 mlt_properties_set( filter_props, "normalise", "" );
307 if ( strcmp( p, "" ) != 0 )
308 gain = strtod( p, &p );
310 while ( isspace( *p ) )
313 /* check if "dB" is given after number */
314 if ( strncaseeq( p, "db", 2 ) )
315 gain = DBFSTOAMP( gain );
319 // If there is an end adjust gain to the range
320 if ( mlt_properties_get( filter_props, "end" ) != NULL )
323 char *p = mlt_properties_get( filter_props, "end" );
324 if ( strcmp( p, "" ) != 0 )
325 end = strtod( p, &p );
327 while ( isspace( *p ) )
330 /* check if "dB" is given after number */
331 if ( strncaseeq( p, "db", 2 ) )
332 end = DBFSTOAMP( gain );
337 gain += ( end - gain ) * mlt_filter_get_progress( this, frame );
341 mlt_properties_set_double( instance_props, "gain", gain );
343 // Parse the maximum gain property
344 if ( mlt_properties_get( filter_props, "max_gain" ) != NULL )
346 char *p = mlt_properties_get( filter_props, "max_gain" );
347 double gain = strtod( p, &p ); // 0 = no max
349 while ( isspace( *p ) )
352 /* check if "dB" is given after number */
353 if ( strncaseeq( p, "db", 2 ) )
354 gain = DBFSTOAMP( gain );
358 mlt_properties_set_double( instance_props, "max_gain", gain );
361 // Parse the limiter property
362 if ( mlt_properties_get( filter_props, "limiter" ) != NULL )
364 char *p = mlt_properties_get( filter_props, "limiter" );
365 double level = 0.5; /* -6dBFS */
366 if ( strcmp( p, "" ) != 0 )
367 level = strtod( p, &p);
369 while ( isspace( *p ) )
372 /* check if "dB" is given after number */
373 if ( strncaseeq( p, "db", 2 ) )
377 level = DBFSTOAMP( level );
384 mlt_properties_set_double( instance_props, "limiter", level );
387 // Parse the normalise property
388 if ( mlt_properties_get( filter_props, "normalise" ) != NULL )
390 char *p = mlt_properties_get( filter_props, "normalise" );
391 double amplitude = 0.2511886431509580; /* -12dBFS */
392 if ( strcmp( p, "" ) != 0 )
393 amplitude = strtod( p, &p);
395 while ( isspace( *p ) )
398 /* check if "dB" is given after number */
399 if ( strncaseeq( p, "db", 2 ) )
402 amplitude = -amplitude;
403 amplitude = DBFSTOAMP( amplitude );
408 amplitude = -amplitude;
409 if ( amplitude > 1.0 )
413 // If there is an end adjust gain to the range
414 if ( mlt_properties_get( filter_props, "end" ) != NULL )
416 amplitude *= mlt_filter_get_progress( this, frame );
418 mlt_properties_set_int( instance_props, "normalise", 1 );
419 mlt_properties_set_double( instance_props, "amplitude", amplitude );
422 // Parse the window property and allocate smoothing buffer if needed
423 int window = mlt_properties_get_int( filter_props, "window" );
424 if ( mlt_properties_get( filter_props, "smooth_buffer" ) == NULL && window > 1 )
426 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
427 double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
429 for ( i = 0; i < window; i++ )
430 smooth_buffer[ i ] = -1.0;
431 mlt_properties_set_data( filter_props, "smooth_buffer", smooth_buffer, 0, free, NULL );
434 // Push the filter onto the stack
435 mlt_frame_push_audio( frame, this );
437 // Override the get_audio method
438 mlt_frame_push_audio( frame, filter_get_audio );
443 /** Constructor for the filter.
446 mlt_filter filter_volume_init( mlt_profile profile, mlt_service_type type, const char *id, char *arg )
448 mlt_filter this = calloc( 1, sizeof( struct mlt_filter_s ) );
449 if ( this != NULL && mlt_filter_init( this, NULL ) == 0 )
451 mlt_properties properties = MLT_FILTER_PROPERTIES( this );
452 this->process = filter_process;
454 mlt_properties_set( properties, "gain", arg );
456 mlt_properties_set_int( properties, "window", 75 );
457 mlt_properties_set( properties, "max_gain", "20dB" );