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 // Get the producer's audio
129 mlt_frame_get_audio( frame
, buffer
, format
, frequency
, &channels_avail
, samples
);
131 // Duplicate channels as necessary
132 if ( channels_avail
< *channels
)
134 int size
= *channels
* *samples
* sizeof( int16_t );
135 int16_t *new_buffer
= mlt_pool_alloc( size
);
138 // Duplicate the existing channels
139 for ( i
= 0; i
< *samples
; i
++ )
141 for ( j
= 0; j
< *channels
; j
++ )
143 new_buffer
[ ( i
* *channels
) + j
] = (*buffer
)[ ( i
* channels_avail
) + k
];
144 k
= ( k
+ 1 ) % channels_avail
;
148 // Update the audio buffer now - destroys the old
149 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
151 *buffer
= new_buffer
;
153 else if ( channels_avail
== 6 && *channels
== 2 )
155 // Nasty hack for ac3 5.1 audio - may be a cause of failure?
156 int size
= *channels
* *samples
* sizeof( int16_t );
157 int16_t *new_buffer
= mlt_pool_alloc( size
);
159 // Drop all but the first *channels
160 for ( i
= 0; i
< *samples
; i
++ )
162 new_buffer
[ ( i
* *channels
) + 0 ] = (*buffer
)[ ( i
* channels_avail
) + 2 ];
163 new_buffer
[ ( i
* *channels
) + 1 ] = (*buffer
)[ ( i
* channels_avail
) + 3 ];
166 // Update the audio buffer now - destroys the old
167 mlt_properties_set_data( properties
, "audio", new_buffer
, size
, ( mlt_destructor
)mlt_pool_release
, NULL
);
169 *buffer
= new_buffer
;
172 // Even though some effects are multi-channel aware, it is not reliable
173 // We must maintain a separate effect state for each channel
174 for ( i
= 0; i
< *channels
; i
++ )
177 sprintf( id
, "_effect_0_%d", i
);
179 // Get an existing effect state
180 eff_t e
= mlt_properties_get_data( filter_properties
, id
, NULL
);
182 // Validate the existing effect state
183 if ( e
!= NULL
&& ( e
->ininfo
.rate
!= *frequency
||
184 e
->outinfo
.rate
!= *frequency
) )
187 // (Re)Create the effect state
195 // Loop over all properties
196 for ( j
= 0; j
< mlt_properties_count( filter_properties
); j
++ )
198 // Get the name of this property
199 char *name
= mlt_properties_get_name( filter_properties
, j
);
201 // If the name does not contain a . and matches effect
202 if ( !strncmp( name
, "effect", 6 ) )
204 // Get the effect specification
205 char *value
= mlt_properties_get( filter_properties
, name
);
207 // Create an instance
208 if ( create_effect( filter
, value
, count
, i
, *frequency
) == 0 )
213 // Save the number of filters
214 mlt_properties_set_int( filter_properties
, "effect_count", count
);
217 if ( *samples
> 0 && count
> 0 )
219 st_sample_t
*p
= input_buffer
;
220 st_sample_t
*end
= p
+ *samples
;
221 int16_t *q
= *buffer
+ i
;
222 st_size_t isamp
= *samples
;
223 st_size_t osamp
= *samples
;
226 char *normalise
= mlt_properties_get( filter_properties
, "normalise" );
227 double normalised_gain
= 1.0;
229 // Convert to sox encoding
232 *p
= ST_SIGNED_WORD_TO_SAMPLE( *q
);
234 // Compute rms amplitude while we are accessing each sample
235 rms
+= ( double )*p
* ( double )*p
;
241 // Compute final rms amplitude
242 rms
= sqrt( rms
/ *samples
/ ST_SSIZE_MIN
/ ST_SSIZE_MIN
);
246 int window
= mlt_properties_get_int( filter_properties
, "window" );
247 double *smooth_buffer
= mlt_properties_get_data( filter_properties
, "smooth_buffer", NULL
);
248 double max_gain
= mlt_properties_get_double( filter_properties
, "max_gain" );
250 // Default the maximum gain factor to 20dBFS
254 // The smoothing buffer prevents radical shifts in the gain level
255 if ( window
> 0 && smooth_buffer
!= NULL
)
257 int smooth_index
= mlt_properties_get_int( filter_properties
, "_smooth_index" );
258 smooth_buffer
[ smooth_index
] = rms
;
260 // Ignore very small values that adversely affect the mean
261 if ( rms
> AMPLITUDE_MIN
)
262 mlt_properties_set_int( filter_properties
, "_smooth_index", ( smooth_index
+ 1 ) % window
);
264 // Smoothing is really just a mean over the past N values
265 normalised_gain
= AMPLITUDE_NORM
/ mean( smooth_buffer
, window
);
269 // Determine gain to apply as current amplitude
270 normalised_gain
= AMPLITUDE_NORM
/ rms
;
273 //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
275 // Govern the maximum gain
276 if ( normalised_gain
> max_gain
)
277 normalised_gain
= max_gain
;
281 for ( j
= 0; j
< count
; j
++ )
283 sprintf( id
, "_effect_%d_%d", j
, i
);
284 e
= mlt_properties_get_data( filter_properties
, id
, NULL
);
286 // We better have this guy
289 float saved_gain
= 1.0;
291 // XXX: hack to apply the normalised gain level to the vol effect
292 if ( normalise
&& strcmp( e
->name
, "vol" ) == 0 )
294 float *f
= ( float * )( e
->priv
);
296 *f
= saved_gain
* normalised_gain
;
300 if ( ( * e
->h
->flow
)( e
, input_buffer
, output_buffer
, &isamp
, &osamp
) == ST_SUCCESS
)
302 // Swap input and output buffer pointers for subsequent effects
304 input_buffer
= output_buffer
;
308 // XXX: hack to restore the original vol gain to prevent accumulation
309 if ( normalise
&& strcmp( e
->name
, "vol" ) == 0 )
311 float *f
= ( float * )( e
->priv
);
317 // Convert back to signed 16bit
323 *q
= ST_SAMPLE_TO_SIGNED_WORD( *p
++ );
332 /** Filter processing.
335 static mlt_frame
filter_process( mlt_filter
this, mlt_frame frame
)
337 if ( mlt_frame_is_test_audio( frame
) == 0 )
339 // Add the filter to the frame
340 mlt_frame_push_audio( frame
, this );
341 mlt_frame_push_audio( frame
, filter_get_audio
);
343 // Parse the window property and allocate smoothing buffer if needed
344 mlt_properties properties
= MLT_FILTER_PROPERTIES( this );
345 int window
= mlt_properties_get_int( properties
, "window" );
346 if ( mlt_properties_get( properties
, "smooth_buffer" ) == NULL
&& window
> 1 )
348 // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
349 double *smooth_buffer
= (double*) calloc( window
, sizeof( double ) );
351 for ( i
= 0; i
< window
; i
++ )
352 smooth_buffer
[ i
] = -1.0;
353 mlt_properties_set_data( properties
, "smooth_buffer", smooth_buffer
, 0, free
, NULL
);
360 /** Constructor for the filter.
363 mlt_filter
filter_sox_init( char *arg
)
365 mlt_filter
this = mlt_filter_new( );
368 void *input_buffer
= mlt_pool_alloc( BUFFER_LEN
);
369 void *output_buffer
= mlt_pool_alloc( BUFFER_LEN
);
370 mlt_properties properties
= MLT_FILTER_PROPERTIES( this );
372 this->process
= filter_process
;
375 mlt_properties_set( properties
, "effect", arg
);
376 mlt_properties_set_data( properties
, "input_buffer", input_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
377 mlt_properties_set_data( properties
, "output_buffer", output_buffer
, BUFFER_LEN
, mlt_pool_release
, NULL
);
378 mlt_properties_set_int( properties
, "window", 75 );
383 // What to do when a libst internal failure occurs
386 // Is there a build problem with my sox-devel package?
388 void gsm_create(void){}
391 void gsm_decode(void){}
394 void gsm_encode(void){}
397 void gsm_destroy(void){}
400 void gsm_option(void){}