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 "filter_volume.h"
23 #include <framework/mlt_frame.h>
31 #define MAX_CHANNELS 6
32 #define EPSILON 0.00001
34 /* The normalise functions come from the normalize utility:
35 Copyright (C) 1999--2002 Chris Vaill */
40 # define ROUND(x) floor((x) + 0.5)
43 #define DBFSTOAMP(x) pow(10,(x)/20.0)
45 /** Return nonzero if the two strings are equal, ignoring case, up to
46 the first n characters.
48 int strncaseeq(const char *s1
, const char *s2
, size_t n
)
52 if (tolower(*s1
++) != tolower(*s2
++))
60 / tanh((x + lev) / (1-lev)) * (1-lev) - lev (for x < -lev)
62 x' = | x (for |x| <= lev)
64 \ tanh((x - lev) / (1-lev)) * (1-lev) + lev (for x > lev)
66 With limiter level = 0, this is equivalent to a tanh() function;
67 with limiter level = 1, this is equivalent to clipping.
69 static inline double limiter( double x
, double lmtr_lvl
)
74 xp
= tanh((x
+ lmtr_lvl
) / (1-lmtr_lvl
)) * (1-lmtr_lvl
) - lmtr_lvl
;
75 else if (x
> lmtr_lvl
)
76 xp
= tanh((x
- lmtr_lvl
) / (1-lmtr_lvl
)) * (1-lmtr_lvl
) + lmtr_lvl
;
79 // fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp );
85 /** Takes a full smoothing window, and returns the value of the center
88 Currently, just does a mean filter, but we could do a median or
89 gaussian filter here instead.
91 static inline double get_smoothed_data( double *buf
, int count
)
96 for ( i
= 0, j
= 0; i
< count
; i
++ )
98 if ( buf
[ i
] != -1.0 )
100 smoothed
+= buf
[ i
];
105 // fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed );
110 /** Get the max power level (using RMS) and peak level of the audio segment.
112 double signal_max_power( int16_t *buffer
, int channels
, int samples
, int16_t *peak
)
114 // Determine numeric limits
115 int bytes_per_samp
= (samp_width
- 1) / 8 + 1;
116 int16_t max
= (1 << (bytes_per_samp
* 8 - 1)) - 1;
117 int16_t min
= -max
- 1;
119 double *sums
= (double *) calloc( channels
, sizeof(double) );
122 double pow
, maxpow
= 0;
124 /* initialize peaks to effectively -inf and +inf */
125 int16_t max_sample
= min
;
126 int16_t min_sample
= max
;
128 for ( i
= 0; i
< samples
; i
++ )
130 for ( c
= 0; c
< channels
; c
++ )
133 sums
[ c
] += (double) sample
* (double) sample
;
136 if ( sample
> max_sample
)
138 else if ( sample
< min_sample
)
142 for ( c
= 0; c
< channels
; c
++ )
144 pow
= sums
[ c
] / (double) samples
;
151 /* scale the pow value to be in the range 0.0 -- 1.0 */
152 maxpow
/= ( (double) min
* (double) min
);
154 if ( -min_sample
> max_sample
)
155 *peak
= min_sample
/ (double) min
;
157 *peak
= max_sample
/ (double) max
;
159 return sqrt( maxpow
);
162 /* ------ End normalize functions --------------------------------------- */
167 static int filter_get_audio( mlt_frame frame
, int16_t **buffer
, mlt_audio_format
*format
, int *frequency
, int *channels
, int *samples
)
169 // Get the properties of the a frame
170 mlt_properties properties
= MLT_FRAME_PROPERTIES( frame
);
171 double gain
= mlt_properties_get_double( properties
, "volume.gain" );
172 double max_gain
= mlt_properties_get_double( properties
, "volume.max_gain" );
173 double limiter_level
= 0.5; /* -6 dBFS */
174 int normalise
= mlt_properties_get_int( properties
, "volume.normalise" );
175 double amplitude
= mlt_properties_get_double( properties
, "volume.amplitude" );
180 // Get the filter from the frame
181 mlt_filter
this = mlt_properties_get_data( properties
, "filter_volume", NULL
);
183 // Get the properties from the filter
184 mlt_properties filter_props
= MLT_FILTER_PROPERTIES( this );
186 if ( mlt_properties_get( properties
, "volume.limiter" ) != NULL
)
187 limiter_level
= mlt_properties_get_double( properties
, "volume.limiter" );
189 // Get the producer's audio
190 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, channels
, samples
);
191 // fprintf( stderr, "filter_volume: frequency %d\n", *frequency );
193 // Determine numeric limits
194 int bytes_per_samp
= (samp_width
- 1) / 8 + 1;
195 int samplemax
= (1 << (bytes_per_samp
* 8 - 1)) - 1;
196 int samplemin
= -samplemax
- 1;
200 int window
= mlt_properties_get_int( filter_props
, "window" );
201 double *smooth_buffer
= mlt_properties_get_data( filter_props
, "smooth_buffer", NULL
);
203 if ( window
> 0 && smooth_buffer
!= NULL
)
205 int smooth_index
= mlt_properties_get_int( filter_props
, "_smooth_index" );
207 // Compute the signal power and put into smoothing buffer
208 smooth_buffer
[ smooth_index
] = signal_max_power( *buffer
, *channels
, *samples
, &peak
);
209 // fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] );
210 if ( smooth_buffer
[ smooth_index
] > EPSILON
)
212 mlt_properties_set_int( filter_props
, "_smooth_index", ( smooth_index
+ 1 ) % window
);
214 // Smooth the data and compute the gain
215 // fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window );
216 gain
*= amplitude
/ get_smoothed_data( smooth_buffer
, window
);
221 gain
*= amplitude
/ signal_max_power( *buffer
, *channels
, *samples
, &peak
);
225 // if ( gain > 1.0 && normalise )
226 // fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain );
228 if ( max_gain
> 0 && gain
> max_gain
)
231 // Initialise filter's previous gain value to prevent an inadvertant jump from 0
232 if ( mlt_properties_get( filter_props
, "previous_gain" ) == NULL
)
233 mlt_properties_set_double( filter_props
, "previous_gain", gain
);
235 // Start the gain out at the previous
236 double previous_gain
= mlt_properties_get_double( filter_props
, "previous_gain" );
238 // Determine ramp increment
239 double gain_step
= ( gain
- previous_gain
) / *samples
;
240 // fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step );
242 // Save the current gain for the next iteration
243 mlt_properties_set_double( filter_props
, "previous_gain", gain
);
245 // Ramp from the previous gain to the current
246 gain
= previous_gain
;
248 int16_t *p
= *buffer
;
251 for ( i
= 0; i
< *samples
; i
++ )
253 for ( j
= 0; j
< *channels
; j
++ )
256 *p
= ROUND( sample
);
260 /* use limiter function instead of clipping */
262 *p
= ROUND( samplemax
* limiter( sample
/ (double) samplemax
, limiter_level
) );
264 /* perform clipping */
265 else if ( sample
> samplemax
)
267 else if ( sample
< samplemin
)
278 /** Filter processing.
281 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
283 mlt_properties properties
= MLT_FRAME_PROPERTIES( frame
);
284 mlt_properties filter_props
= MLT_FILTER_PROPERTIES( this );
286 // Parse the gain property
287 if ( mlt_properties_get( properties
, "gain" ) == NULL
)
289 double gain
= 1.0; // no adjustment
291 if ( mlt_properties_get( filter_props
, "gain" ) != NULL
)
293 char *p
= mlt_properties_get( filter_props
, "gain" );
295 if ( strncaseeq( p
, "normalise", 9 ) )
296 mlt_properties_set( filter_props
, "normalise", "" );
299 if ( strcmp( p
, "" ) != 0 )
300 gain
= fabs( strtod( p
, &p
) );
302 while ( isspace( *p
) )
305 /* check if "dB" is given after number */
306 if ( strncaseeq( p
, "db", 2 ) )
307 gain
= DBFSTOAMP( gain
);
309 // If there is an end adjust gain to the range
310 if ( mlt_properties_get( filter_props
, "end" ) != NULL
)
312 // Determine the time position of this frame in the transition duration
313 mlt_position in
= mlt_filter_get_in( this );
314 mlt_position out
= mlt_filter_get_out( this );
315 mlt_position time
= mlt_frame_get_position( frame
);
316 double position
= ( double )( time
- in
) / ( double )( out
- in
+ 1 );
319 char *p
= mlt_properties_get( filter_props
, "end" );
320 if ( strcmp( p
, "" ) != 0 )
321 end
= fabs( strtod( p
, &p
) );
323 while ( isspace( *p
) )
326 /* check if "dB" is given after number */
327 if ( strncaseeq( p
, "db", 2 ) )
328 end
= DBFSTOAMP( gain
);
331 gain
+= ( end
- gain
) * position
;
335 mlt_properties_set_double( properties
, "volume.gain", gain
);
338 // Parse the maximum gain property
339 if ( mlt_properties_get( filter_props
, "max_gain" ) != NULL
)
341 char *p
= mlt_properties_get( filter_props
, "max_gain" );
342 double gain
= fabs( strtod( p
, &p
) ); // 0 = no max
344 while ( isspace( *p
) )
347 /* check if "dB" is given after number */
348 if ( strncaseeq( p
, "db", 2 ) )
349 gain
= DBFSTOAMP( gain
);
351 mlt_properties_set_double( properties
, "volume.max_gain", gain
);
354 // Parse the limiter property
355 if ( mlt_properties_get( filter_props
, "limiter" ) != NULL
)
357 char *p
= mlt_properties_get( filter_props
, "limiter" );
358 double level
= 0.5; /* -6dBFS */
359 if ( strcmp( p
, "" ) != 0 )
360 level
= strtod( p
, &p
);
362 while ( isspace( *p
) )
365 /* check if "dB" is given after number */
366 if ( strncaseeq( p
, "db", 2 ) )
370 level
= DBFSTOAMP( level
);
377 mlt_properties_set_double( properties
, "volume.limiter", level
);
380 // Parse the normalise property
381 if ( mlt_properties_get( filter_props
, "normalise" ) != NULL
)
383 char *p
= mlt_properties_get( filter_props
, "normalise" );
384 double amplitude
= 0.2511886431509580; /* -12dBFS */
385 if ( strcmp( p
, "" ) != 0 )
386 amplitude
= strtod( p
, &p
);
388 while ( isspace( *p
) )
391 /* check if "dB" is given after number */
392 if ( strncaseeq( p
, "db", 2 ) )
395 amplitude
= -amplitude
;
396 amplitude
= DBFSTOAMP( amplitude
);
401 amplitude
= -amplitude
;
402 if ( amplitude
> 1.0 )
406 // If there is an end adjust gain to the range
407 if ( mlt_properties_get( filter_props
, "end" ) != NULL
)
409 // Determine the time position of this frame in the transition duration
410 mlt_position in
= mlt_filter_get_in( this );
411 mlt_position out
= mlt_filter_get_out( this );
412 mlt_position time
= mlt_frame_get_position( frame
);
413 double position
= ( double )( time
- in
) / ( double )( out
- in
+ 1 );
414 amplitude
*= position
;
416 mlt_properties_set_int( properties
, "volume.normalise", 1 );
417 mlt_properties_set_double( properties
, "volume.amplitude", amplitude
);
420 // Parse the window property and allocate smoothing buffer if needed
421 int window
= mlt_properties_get_int( filter_props
, "window" );
422 if ( mlt_properties_get( filter_props
, "smooth_buffer" ) == NULL
&& window
> 1 )
424 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
425 double *smooth_buffer
= (double*) calloc( window
, sizeof( double ) );
427 for ( i
= 0; i
< window
; i
++ )
428 smooth_buffer
[ i
] = -1.0;
429 mlt_properties_set_data( filter_props
, "smooth_buffer", smooth_buffer
, 0, free
, NULL
);
432 // Put a filter reference onto the frame
433 mlt_properties_set_data( properties
, "filter_volume", this, 0, NULL
, NULL
);
435 // Override the get_audio method
436 mlt_frame_push_audio( frame
, filter_get_audio
);
441 /** Constructor for the filter.
444 mlt_filter
filter_volume_init( char *arg
)
446 mlt_filter
this = calloc( sizeof( struct mlt_filter_s
), 1 );
447 if ( this != NULL
&& mlt_filter_init( this, NULL
) == 0 )
449 mlt_properties properties
= MLT_FILTER_PROPERTIES( this );
450 this->process
= filter_process
;
452 mlt_properties_set( properties
, "gain", arg
);
454 mlt_properties_set_int( properties
, "window", 75 );
455 mlt_properties_set( properties
, "max_gain", "20dB" );