1 /*****************************************************************************
2 * bandlimited.c : band-limited interpolation resampler
3 *****************************************************************************
4 * Copyright (C) 2002, 2006 the VideoLAN team
7 * Authors: Gildas Bazin <gbazin@netcourrier.com>
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
24 /*****************************************************************************
27 * This implementation of the band-limited interpolationis based on the
29 * http://ccrma-www.stanford.edu/~jos/resample/resample.html
31 * It uses a Kaiser-windowed sinc-function low-pass filter and the width of the
32 * filter is 13 samples.
34 *****************************************************************************/
43 #include "bandlimited.h"
45 /*****************************************************************************
47 *****************************************************************************/
48 static int Create ( vlc_object_t * );
49 static void Close ( vlc_object_t * );
50 static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
53 static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
54 float *f_in, float *f_out, uint32_t ui_remainder,
55 uint32_t ui_output_rate, int16_t Inc,
58 static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
59 float *f_in, float *f_out, uint32_t ui_remainder,
60 uint32_t ui_output_rate, uint32_t ui_input_rate,
61 int16_t Inc, int i_nb_channels );
63 /*****************************************************************************
65 *****************************************************************************/
66 struct aout_filter_sys_t
68 int32_t *p_buf; /* this filter introduces a delay */
75 unsigned int i_remainder; /* remainder of previous sample */
77 audio_date_t end_date;
80 /*****************************************************************************
82 *****************************************************************************/
84 set_category( CAT_AUDIO );
85 set_subcategory( SUBCAT_AUDIO_MISC );
86 set_description( _("Audio filter for band-limited interpolation resampling") );
87 set_capability( "audio filter", 20 );
88 set_callbacks( Create, Close );
91 /*****************************************************************************
92 * Create: allocate linear resampler
93 *****************************************************************************/
94 static int Create( vlc_object_t *p_this )
96 aout_filter_t * p_filter = (aout_filter_t *)p_this;
100 if ( p_filter->input.i_rate == p_filter->output.i_rate
101 || p_filter->input.i_format != p_filter->output.i_format
102 || p_filter->input.i_physical_channels
103 != p_filter->output.i_physical_channels
104 || p_filter->input.i_original_channels
105 != p_filter->output.i_original_channels
106 || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
111 #if !defined( __APPLE__ )
112 if( !config_GetInt( p_this, "hq-resampling" ) )
118 /* Allocate the memory needed to store the module's structure */
119 p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
120 if( p_filter->p_sys == NULL )
122 msg_Err( p_filter, "out of memory" );
126 /* Calculate worst case for the length of the filter wing */
127 d_factor = (double)p_filter->output.i_rate
128 / p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
129 i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
130 * __MAX(1.0, 1.0/d_factor) + 10;
131 p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
132 sizeof(int32_t) * 2 * i_filter_wing;
134 /* Allocate enough memory to buffer previous samples */
135 p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
136 if( p_filter->p_sys->p_buf == NULL )
138 msg_Err( p_filter, "out of memory" );
142 p_filter->p_sys->i_old_wing = 0;
143 p_filter->pf_do_work = DoWork;
145 /* We don't want a new buffer to be created because we're not sure we'll
146 * actually need to resample anything. */
147 p_filter->b_in_place = true;
152 /*****************************************************************************
153 * Close: free our resources
154 *****************************************************************************/
155 static void Close( vlc_object_t * p_this )
157 aout_filter_t * p_filter = (aout_filter_t *)p_this;
158 free( p_filter->p_sys->p_buf );
159 free( p_filter->p_sys );
162 /*****************************************************************************
163 * DoWork: convert a buffer
164 *****************************************************************************/
165 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
166 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
168 float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
170 int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
171 int i_in_nb = p_in_buf->i_nb_samples;
173 double d_factor, d_scale_factor, d_old_scale_factor;
179 /* Check if we really need to run the resampler */
180 if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
182 if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
183 p_filter->p_sys->i_old_wing &&
185 p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
186 p_filter->input.i_bytes_per_frame )
188 /* output the whole thing with the samples from last time */
189 memmove( ((float *)(p_in_buf->p_buffer)) +
190 i_nb_channels * p_filter->p_sys->i_old_wing,
191 p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
192 memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
193 i_nb_channels * p_filter->p_sys->i_old_wing,
194 p_filter->p_sys->i_old_wing *
195 p_filter->input.i_bytes_per_frame );
197 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
198 p_filter->p_sys->i_old_wing;
200 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
201 p_out_buf->end_date =
202 aout_DateIncrement( &p_filter->p_sys->end_date,
203 p_out_buf->i_nb_samples );
205 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
206 p_filter->input.i_bytes_per_frame;
208 p_filter->b_continuity = false;
209 p_filter->p_sys->i_old_wing = 0;
213 if( !p_filter->b_continuity )
215 /* Continuity in sound samples has been broken, we'd better reset
217 p_filter->b_continuity = true;
218 p_filter->p_sys->i_remainder = 0;
219 aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
220 aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
221 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
222 p_filter->p_sys->d_old_factor = 1;
223 p_filter->p_sys->i_old_wing = 0;
227 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
228 p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
229 p_filter->p_sys->i_old_wing, i_in_nb );
232 /* Prepare the source buffer */
233 i_in_nb += (p_filter->p_sys->i_old_wing * 2);
235 p_in = p_in_orig = (float *)alloca( i_in_nb *
236 p_filter->input.i_bytes_per_frame );
238 p_in = p_in_orig = (float *)malloc( i_in_nb *
239 p_filter->input.i_bytes_per_frame );
246 /* Copy all our samples in p_in */
247 if( p_filter->p_sys->i_old_wing )
249 vlc_memcpy( p_in, p_filter->p_sys->p_buf,
250 p_filter->p_sys->i_old_wing * 2 *
251 p_filter->input.i_bytes_per_frame );
253 vlc_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 * i_nb_channels,
255 p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );
257 /* Make sure the output buffer is reset */
258 memset( p_out, 0, p_out_buf->i_size );
260 /* Calculate the new length of the filter wing */
261 d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
262 i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
264 /* Account for increased filter gain when using factors less than 1 */
265 d_old_scale_factor = SMALL_FILTER_SCALE *
266 p_filter->p_sys->d_old_factor + 0.5;
267 d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
269 /* Apply the old rate until we have enough samples for the new one */
270 i_in = p_filter->p_sys->i_old_wing;
271 p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
272 for( ; i_in < i_filter_wing &&
273 (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
275 if( p_filter->p_sys->d_old_factor == 1 )
277 /* Just copy the samples */
279 p_filter->input.i_bytes_per_frame );
280 p_in += i_nb_channels;
281 p_out += i_nb_channels;
286 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
289 if( p_filter->p_sys->d_old_factor >= 1 )
291 /* FilterFloatUP() is faster if we can use it */
293 /* Perform left-wing inner product */
294 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
295 SMALL_FILTER_NWING, p_in, p_out,
296 p_filter->p_sys->i_remainder,
297 p_filter->output.i_rate,
299 /* Perform right-wing inner product */
300 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
301 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
302 p_filter->output.i_rate -
303 p_filter->p_sys->i_remainder,
304 p_filter->output.i_rate,
308 /* Normalize for unity filter gain */
309 for( i = 0; i < i_nb_channels; i++ )
311 *(p_out+i) *= d_old_scale_factor;
316 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
317 <= (unsigned int)i_out+1 )
319 p_out += i_nb_channels;
321 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
327 /* Perform left-wing inner product */
328 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
329 SMALL_FILTER_NWING, p_in, p_out,
330 p_filter->p_sys->i_remainder,
331 p_filter->output.i_rate, p_filter->input.i_rate,
333 /* Perform right-wing inner product */
334 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
335 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
336 p_filter->output.i_rate -
337 p_filter->p_sys->i_remainder,
338 p_filter->output.i_rate, p_filter->input.i_rate,
342 p_out += i_nb_channels;
345 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
348 p_in += i_nb_channels;
349 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
352 /* Apply the new rate for the rest of the samples */
353 if( i_in < i_in_nb - i_filter_wing )
355 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
356 p_filter->p_sys->d_old_factor = d_factor;
357 p_filter->p_sys->i_old_wing = i_filter_wing;
359 for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
361 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
366 /* FilterFloatUP() is faster if we can use it */
368 /* Perform left-wing inner product */
369 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
370 SMALL_FILTER_NWING, p_in, p_out,
371 p_filter->p_sys->i_remainder,
372 p_filter->output.i_rate,
375 /* Perform right-wing inner product */
376 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
377 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
378 p_filter->output.i_rate -
379 p_filter->p_sys->i_remainder,
380 p_filter->output.i_rate,
384 /* Normalize for unity filter gain */
385 for( i = 0; i < i_nb_channels; i++ )
387 *(p_out+i) *= d_old_scale_factor;
391 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
392 <= (unsigned int)i_out+1 )
394 p_out += i_nb_channels;
396 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
402 /* Perform left-wing inner product */
403 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
404 SMALL_FILTER_NWING, p_in, p_out,
405 p_filter->p_sys->i_remainder,
406 p_filter->output.i_rate, p_filter->input.i_rate,
408 /* Perform right-wing inner product */
409 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
410 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
411 p_filter->output.i_rate -
412 p_filter->p_sys->i_remainder,
413 p_filter->output.i_rate, p_filter->input.i_rate,
417 p_out += i_nb_channels;
420 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
423 p_in += i_nb_channels;
424 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
427 /* Buffer i_filter_wing * 2 samples for next time */
428 if( p_filter->p_sys->i_old_wing )
430 memcpy( p_filter->p_sys->p_buf,
431 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
432 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
433 p_filter->input.i_bytes_per_frame );
437 msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
438 i_out * p_filter->input.i_bytes_per_frame );
441 /* Free the temp buffer */
446 /* Finalize aout buffer */
447 p_out_buf->i_nb_samples = i_out;
448 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
449 p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
450 p_out_buf->i_nb_samples );
452 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
453 i_nb_channels * sizeof(int32_t);
457 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
458 float *p_out, uint32_t ui_remainder,
459 uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
461 float *Hp, *Hdp, *End;
463 uint32_t ui_linear_remainder;
466 Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
467 Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
471 ui_linear_remainder = (ui_remainder<<Nhc) -
472 (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
474 if (Inc == 1) /* If doing right wing... */
475 { /* ...drop extra coeff, so when Ph is */
476 End--; /* 0.5, we don't do too many mult's */
477 if (ui_remainder == 0) /* If the phase is zero... */
478 { /* ...then we've already skipped the */
479 Hp += Npc; /* first sample, so we must also */
480 Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
485 t = *Hp; /* Get filter coeff */
486 /* t is now interp'd filter coeff */
487 t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
488 for( i = 0; i < i_nb_channels; i++ )
491 temp *= *(p_in+i); /* Mult coeff by input sample */
492 *(p_out+i) += temp; /* The filter output */
494 Hdp += Npc; /* Filter coeff differences step */
495 Hp += Npc; /* Filter coeff step */
496 p_in += (Inc * i_nb_channels); /* Input signal step */
500 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
501 float *p_out, uint32_t ui_remainder,
502 uint32_t ui_output_rate, uint32_t ui_input_rate,
503 int16_t Inc, int i_nb_channels )
505 float *Hp, *Hdp, *End;
507 uint32_t ui_linear_remainder;
508 int i, ui_counter = 0;
510 Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
511 Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
515 if (Inc == 1) /* If doing right wing... */
516 { /* ...drop extra coeff, so when Ph is */
517 End--; /* 0.5, we don't do too many mult's */
518 if (ui_remainder == 0) /* If the phase is zero... */
519 { /* ...then we've already skipped the */
520 Hp = Imp + /* first sample, so we must also */
521 (ui_output_rate << Nhc) / ui_input_rate;
522 Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
523 (ui_output_rate << Nhc) / ui_input_rate;
529 t = *Hp; /* Get filter coeff */
530 /* t is now interp'd filter coeff */
531 ui_linear_remainder =
532 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
533 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
534 ui_input_rate * ui_input_rate;
535 t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
536 for( i = 0; i < i_nb_channels; i++ )
539 temp *= *(p_in+i); /* Mult coeff by input sample */
540 *(p_out+i) += temp; /* The filter output */
545 /* Filter coeff step */
546 Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
548 /* Filter coeff differences step */
549 Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
552 p_in += (Inc * i_nb_channels); /* Input signal step */