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 "filter_sox.h"
23 #include <framework/mlt_frame.h>
24 #include <framework/mlt_tokeniser.h>
33 # define ST_EOF SOX_EOF
34 # define ST_SUCCESS SOX_SUCCESS
35 # define st_sample_t sox_sample_t
36 # define eff_t sox_effect_t*
37 # define st_size_t sox_size_t
38 # define ST_LIB_VERSION_CODE SOX_LIB_VERSION_CODE
39 # define ST_LIB_VERSION SOX_LIB_VERSION
40 # define ST_SIGNED_WORD_TO_SAMPLE(d,clips) SOX_SIGNED_16BIT_TO_SAMPLE(d,clips)
41 # define ST_SSIZE_MIN SOX_SSIZE_MIN
42 # define ST_SAMPLE_TO_SIGNED_WORD(d,clips) SOX_SAMPLE_TO_SIGNED_16BIT(d,clips)
47 #define BUFFER_LEN 8192
48 #define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
49 #define AMPLITUDE_MIN 0.00001
51 /** Compute the mean of a set of doubles skipping unset values flagged as -1
53 static inline double mean( double *buf
, int count
)
59 for ( i
= 0; i
< count
; i
++ )
61 if ( buf
[ i
] != -1.0 )
73 /** Create an effect state instance for a channels
75 static int create_effect( mlt_filter
this, char *value
, int count
, int channel
, int frequency
)
77 mlt_tokeniser tokeniser
= mlt_tokeniser_init();
79 eff_t eff
= mlt_pool_alloc( sizeof( sox_effect_t
) );
81 eff_t eff
= mlt_pool_alloc( sizeof( struct st_effect
) );
86 // Tokenise the effect specification
87 mlt_tokeniser_parse_new( tokeniser
, value
, " " );
88 if ( tokeniser
->count
< 1 )
93 //fprintf(stderr, "%s: effect %s count %d\n", __FUNCTION__, tokeniser->tokens[0], tokeniser->count );
94 sox_create_effect( eff
, sox_find_effect( tokeniser
->tokens
[0] ) );
95 int opt_count
= tokeniser
->count
- 1;
97 int opt_count
= st_geteffect_opt( eff
, tokeniser
->count
, tokeniser
->tokens
);
101 if ( opt_count
!= ST_EOF
)
103 // Supply the effect parameters
105 if ( ( * eff
->handler
.getopts
)( eff
, opt_count
, &tokeniser
->tokens
[ tokeniser
->count
> 1 ?
1 : 0 ] ) == ST_SUCCESS
)
107 if ( ( * eff
->h
->getopts
)( eff
, opt_count
, &tokeniser
->tokens
[ tokeniser
->count
- opt_count
] ) == ST_SUCCESS
)
110 // Set the sox signal parameters
111 eff
->ininfo
.rate
= frequency
;
112 eff
->outinfo
.rate
= frequency
;
113 eff
->ininfo
.channels
= 1;
114 eff
->outinfo
.channels
= 1;
118 if ( ( * eff
->handler
.start
)( eff
) == ST_SUCCESS
)
120 if ( ( * eff
->h
->start
)( eff
) == ST_SUCCESS
)
124 sprintf( id
, "_effect_%d_%d", count
, channel
);
126 // Save the effect state
127 mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id
, eff
, 0, mlt_pool_release
, NULL
);
132 // Some error occurred so delete the temp effect state
134 mlt_pool_release( eff
);
136 mlt_tokeniser_close( tokeniser
);
144 static int filter_get_audio( mlt_frame frame
, int16_t **buffer
, mlt_audio_format
*format
, int *frequency
, int *channels
, int *samples
)
146 // Get the properties of the frame
147 mlt_properties properties
= MLT_FRAME_PROPERTIES( frame
);
149 // Get the filter service
150 mlt_filter filter
= mlt_frame_pop_audio( frame
);
152 // Get the filter properties
153 mlt_properties filter_properties
= MLT_FILTER_PROPERTIES( filter
);
155 // Get the properties
156 st_sample_t
*input_buffer
= mlt_properties_get_data( filter_properties
, "input_buffer", NULL
);
157 st_sample_t
*output_buffer
= mlt_properties_get_data( filter_properties
, "output_buffer", NULL
);
158 int channels_avail
= *channels
;
160 int count
= mlt_properties_get_int( filter_properties
, "_effect_count" );
162 // Get the producer's audio
163 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, &channels_avail
, samples
);
165 // Duplicate channels as necessary
166 if ( channels_avail
< *channels
)
168 int size
= *channels
* *samples
* sizeof( int16_t );
169 int16_t *new_buffer
= mlt_pool_alloc( size
);
172 // Duplicate the existing channels
173 for ( i
= 0; i
< *samples
; i
++ )
175 for ( j
= 0; j
< *channels
; j
++ )
177 new_buffer
[ ( i
* *channels
) + j
] = (*buffer
)[ ( i
* channels_avail
) + k
];
178 k
= ( k
+ 1 ) % channels_avail
;
182 // Update the audio buffer now - destroys the old
183 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
185 *buffer
= new_buffer
;
187 else if ( channels_avail
== 6 && *channels
== 2 )
189 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
190 int size
= *channels
* *samples
* sizeof( int16_t );
191 int16_t *new_buffer
= mlt_pool_alloc( size
);
193 // Drop all but the first *channels
194 for ( i
= 0; i
< *samples
; i
++ )
196 new_buffer
[ ( i
* *channels
) + 0 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
197 new_buffer
[ ( i
* *channels
) + 1 ] = (*buffer
)[ ( i
* channels_avail
) + 3 ];
200 // Update the audio buffer now - destroys the old
201 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
203 *buffer
= new_buffer
;
206 // Even though some effects are multi-channel aware, it is not reliable
207 // We must maintain a separate effect state for each channel
208 for ( i
= 0; i
< *channels
; i
++ )
211 sprintf( id
, "_effect_0_%d", i
);
213 // Get an existing effect state
214 eff_t e
= mlt_properties_get_data( filter_properties
, id
, NULL
);
216 // Validate the existing effect state
217 if ( e
!= NULL
&& ( e
->ininfo
.rate
!= *frequency
||
218 e
->outinfo
.rate
!= *frequency
) )
221 // (Re)Create the effect state
229 // Loop over all properties
230 for ( j
= 0; j
< mlt_properties_count( filter_properties
); j
++ )
232 // Get the name of this property
233 char *name
= mlt_properties_get_name( filter_properties
, j
);
235 // If the name does not contain a . and matches effect
236 if ( !strncmp( name
, "effect", 6 ) )
238 // Get the effect specification
239 char *value
= mlt_properties_get( filter_properties
, name
);
241 // Create an instance
242 if ( create_effect( filter
, value
, count
, i
, *frequency
) == 0 )
247 // Save the number of filters
248 mlt_properties_set_int( filter_properties
, "_effect_count", count
);
251 if ( *samples
> 0 && count
> 0 )
253 st_sample_t
*p
= input_buffer
;
254 st_sample_t
*end
= p
+ *samples
;
255 int16_t *q
= *buffer
+ i
;
256 st_size_t isamp
= *samples
;
257 st_size_t osamp
= *samples
;
260 char *normalise
= mlt_properties_get( filter_properties
, "normalise" );
261 double normalised_gain
= 1.0;
262 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
263 st_sample_t dummy_clipped_count
= 0;
266 // Convert to sox encoding
269 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
270 *p
= ST_SIGNED_WORD_TO_SAMPLE( *q
, dummy_clipped_count
);
272 *p
= ST_SIGNED_WORD_TO_SAMPLE( *q
);
274 // Compute rms amplitude while we are accessing each sample
275 rms
+= ( double )*p
* ( double )*p
;
281 // Compute final rms amplitude
282 rms
= sqrt( rms
/ *samples
/ ST_SSIZE_MIN
/ ST_SSIZE_MIN
);
286 int window
= mlt_properties_get_int( filter_properties
, "window" );
287 double *smooth_buffer
= mlt_properties_get_data( filter_properties
, "smooth_buffer", NULL
);
288 double max_gain
= mlt_properties_get_double( filter_properties
, "max_gain" );
290 // Default the maximum gain factor to 20dBFS
294 // The smoothing buffer prevents radical shifts in the gain level
295 if ( window
> 0 && smooth_buffer
!= NULL
)
297 int smooth_index
= mlt_properties_get_int( filter_properties
, "_smooth_index" );
298 smooth_buffer
[ smooth_index
] = rms
;
300 // Ignore very small values that adversely affect the mean
301 if ( rms
> AMPLITUDE_MIN
)
302 mlt_properties_set_int( filter_properties
, "_smooth_index", ( smooth_index
+ 1 ) % window
);
304 // Smoothing is really just a mean over the past N values
305 normalised_gain
= AMPLITUDE_NORM
/ mean( smooth_buffer
, window
);
309 // Determine gain to apply as current amplitude
310 normalised_gain
= AMPLITUDE_NORM
/ rms
;
313 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
315 // Govern the maximum gain
316 if ( normalised_gain
> max_gain
)
317 normalised_gain
= max_gain
;
321 for ( j
= 0; j
< count
; j
++ )
323 sprintf( id
, "_effect_%d_%d", j
, i
);
324 e
= mlt_properties_get_data( filter_properties
, id
, NULL
);
326 // We better have this guy
329 float saved_gain
= 1.0;
331 // XXX: hack to apply the normalised gain level to the vol effect
333 if ( normalise
&& strcmp( e
->handler
.name
, "vol" ) == 0 )
335 if ( normalise
&& strcmp( e
->name
, "vol" ) == 0 )
338 float *f
= ( float * )( e
->priv
);
340 *f
= saved_gain
* normalised_gain
;
345 if ( ( * e
->handler
.flow
)( e
, input_buffer
, output_buffer
, &isamp
, &osamp
) == ST_SUCCESS
)
347 if ( ( * e
->h
->flow
)( e
, input_buffer
, output_buffer
, &isamp
, &osamp
) == ST_SUCCESS
)
350 // Swap input and output buffer pointers for subsequent effects
352 input_buffer
= output_buffer
;
356 // XXX: hack to restore the original vol gain to prevent accumulation
358 if ( normalise
&& strcmp( e
->handler
.name
, "vol" ) == 0 )
360 if ( normalise
&& strcmp( e
->name
, "vol" ) == 0 )
363 float *f
= ( float * )( e
->priv
);
369 // Convert back to signed 16bit
375 #if (ST_LIB_VERSION_CODE >= ST_LIB_VERSION(13,0,0))
376 *q
= ST_SAMPLE_TO_SIGNED_WORD( *p
++, dummy_clipped_count
);
378 *q
= ST_SAMPLE_TO_SIGNED_WORD( *p
++ );
388 /** Filter processing.
391 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
393 if ( mlt_frame_is_test_audio( frame
) == 0 )
395 // Add the filter to the frame
396 mlt_frame_push_audio( frame
, this );
397 mlt_frame_push_audio( frame
, filter_get_audio
);
399 // Parse the window property and allocate smoothing buffer if needed
400 mlt_properties properties
= MLT_FILTER_PROPERTIES( this );
401 int window
= mlt_properties_get_int( properties
, "window" );
402 if ( mlt_properties_get( properties
, "smooth_buffer" ) == NULL
&& window
> 1 )
404 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
405 double *smooth_buffer
= (double*) calloc( window
, sizeof( double ) );
407 for ( i
= 0; i
< window
; i
++ )
408 smooth_buffer
[ i
] = -1.0;
409 mlt_properties_set_data( properties
, "smooth_buffer", smooth_buffer
, 0, free
, NULL
);
416 /** Constructor for the filter.
419 mlt_filter
filter_sox_init( char *arg
)
421 mlt_filter
this = mlt_filter_new( );
424 void *input_buffer
= mlt_pool_alloc( BUFFER_LEN
);
425 void *output_buffer
= mlt_pool_alloc( BUFFER_LEN
);
426 mlt_properties properties
= MLT_FILTER_PROPERTIES( this );
428 this->process
= filter_process
;
431 mlt_properties_set( properties
, "effect", arg
);
432 mlt_properties_set_data( properties
, "input_buffer", input_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
433 mlt_properties_set_data( properties
, "output_buffer", output_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
434 mlt_properties_set_int( properties
, "window", 75 );
439 // What to do when a libst internal failure occurs
442 // Is there a build problem with my sox-devel package?
444 void gsm_create(void){}
447 void gsm_decode(void){}
450 void gsm_encode(void){}
453 void gsm_destroy(void){}
456 void gsm_option(void){}