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 *****************************************************************************/
40 #include <vlc_common.h>
41 #include <vlc_plugin.h>
44 #include "bandlimited.h"
46 /*****************************************************************************
48 *****************************************************************************/
49 static int Create ( vlc_object_t * );
50 static void Close ( vlc_object_t * );
51 static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
54 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
55 float *f_in, float *f_out, uint32_t ui_remainder,
56 uint32_t ui_output_rate, int16_t Inc,
59 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
60 float *f_in, float *f_out, uint32_t ui_remainder,
61 uint32_t ui_output_rate, uint32_t ui_input_rate,
62 int16_t Inc, int i_nb_channels );
64 /*****************************************************************************
66 *****************************************************************************/
67 struct aout_filter_sys_t
69 int32_t *p_buf; /* this filter introduces a delay */
76 unsigned int i_remainder; /* remainder of previous sample */
78 audio_date_t end_date;
81 /*****************************************************************************
83 *****************************************************************************/
85 set_category( CAT_AUDIO );
86 set_subcategory( SUBCAT_AUDIO_MISC );
87 set_description( N_("Audio filter for band-limited interpolation resampling") );
88 set_capability( "audio filter", 20 );
89 set_callbacks( Create, Close );
92 /*****************************************************************************
93 * Create: allocate linear resampler
94 *****************************************************************************/
95 static int Create( vlc_object_t *p_this )
97 aout_filter_t * p_filter = (aout_filter_t *)p_this;
101 if ( p_filter->input.i_rate == p_filter->output.i_rate
102 || p_filter->input.i_format != p_filter->output.i_format
103 || p_filter->input.i_physical_channels
104 != p_filter->output.i_physical_channels
105 || p_filter->input.i_original_channels
106 != p_filter->output.i_original_channels
107 || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
112 #if !defined( __APPLE__ )
113 if( !config_GetInt( p_this, "hq-resampling" ) )
119 /* Allocate the memory needed to store the module's structure */
120 p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
121 if( p_filter->p_sys == NULL )
124 /* Calculate worst case for the length of the filter wing */
125 d_factor = (double)p_filter->output.i_rate
126 / p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
127 i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
128 * __MAX(1.0, 1.0/d_factor) + 10;
129 p_filter->p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
130 sizeof(int32_t) * 2 * i_filter_wing;
132 /* Allocate enough memory to buffer previous samples */
133 p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
134 if( p_filter->p_sys->p_buf == NULL )
137 p_filter->p_sys->i_old_wing = 0;
138 p_filter->pf_do_work = DoWork;
140 /* We don't want a new buffer to be created because we're not sure we'll
141 * actually need to resample anything. */
142 p_filter->b_in_place = true;
147 /*****************************************************************************
148 * Close: free our resources
149 *****************************************************************************/
150 static void Close( vlc_object_t * p_this )
152 aout_filter_t * p_filter = (aout_filter_t *)p_this;
153 free( p_filter->p_sys->p_buf );
154 free( p_filter->p_sys );
157 /*****************************************************************************
158 * DoWork: convert a buffer
159 *****************************************************************************/
160 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
161 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
163 float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
165 int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
166 int i_in_nb = p_in_buf->i_nb_samples;
168 double d_factor, d_scale_factor, d_old_scale_factor;
174 /* Check if we really need to run the resampler */
175 if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
177 if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
178 p_filter->p_sys->i_old_wing &&
180 p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
181 p_filter->input.i_bytes_per_frame )
183 /* output the whole thing with the samples from last time */
184 memmove( ((float *)(p_in_buf->p_buffer)) +
185 i_nb_channels * p_filter->p_sys->i_old_wing,
186 p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
187 memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
188 i_nb_channels * p_filter->p_sys->i_old_wing,
189 p_filter->p_sys->i_old_wing *
190 p_filter->input.i_bytes_per_frame );
192 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
193 p_filter->p_sys->i_old_wing;
195 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
196 p_out_buf->end_date =
197 aout_DateIncrement( &p_filter->p_sys->end_date,
198 p_out_buf->i_nb_samples );
200 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
201 p_filter->input.i_bytes_per_frame;
203 p_filter->b_continuity = false;
204 p_filter->p_sys->i_old_wing = 0;
208 if( !p_filter->b_continuity )
210 /* Continuity in sound samples has been broken, we'd better reset
212 p_filter->b_continuity = true;
213 p_filter->p_sys->i_remainder = 0;
214 aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
215 aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
216 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
217 p_filter->p_sys->d_old_factor = 1;
218 p_filter->p_sys->i_old_wing = 0;
222 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
223 p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
224 p_filter->p_sys->i_old_wing, i_in_nb );
227 /* Prepare the source buffer */
228 i_in_nb += (p_filter->p_sys->i_old_wing * 2);
230 p_in = p_in_orig = (float *)alloca( i_in_nb *
231 p_filter->input.i_bytes_per_frame );
233 p_in = p_in_orig = (float *)malloc( i_in_nb *
234 p_filter->input.i_bytes_per_frame );
241 /* Copy all our samples in p_in */
242 if( p_filter->p_sys->i_old_wing )
244 vlc_memcpy( p_in, p_filter->p_sys->p_buf,
245 p_filter->p_sys->i_old_wing * 2 *
246 p_filter->input.i_bytes_per_frame );
248 vlc_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 * i_nb_channels,
250 p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );
252 /* Make sure the output buffer is reset */
253 memset( p_out, 0, p_out_buf->i_size );
255 /* Calculate the new length of the filter wing */
256 d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
257 i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
259 /* Account for increased filter gain when using factors less than 1 */
260 d_old_scale_factor = SMALL_FILTER_SCALE *
261 p_filter->p_sys->d_old_factor + 0.5;
262 d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
264 /* Apply the old rate until we have enough samples for the new one */
265 i_in = p_filter->p_sys->i_old_wing;
266 p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
267 for( ; i_in < i_filter_wing &&
268 (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
270 if( p_filter->p_sys->d_old_factor == 1 )
272 /* Just copy the samples */
274 p_filter->input.i_bytes_per_frame );
275 p_in += i_nb_channels;
276 p_out += i_nb_channels;
281 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
284 if( p_filter->p_sys->d_old_factor >= 1 )
286 /* FilterFloatUP() is faster if we can use it */
288 /* Perform left-wing inner product */
289 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
290 SMALL_FILTER_NWING, p_in, p_out,
291 p_filter->p_sys->i_remainder,
292 p_filter->output.i_rate,
294 /* Perform right-wing inner product */
295 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
296 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
297 p_filter->output.i_rate -
298 p_filter->p_sys->i_remainder,
299 p_filter->output.i_rate,
303 /* Normalize for unity filter gain */
304 for( i = 0; i < i_nb_channels; i++ )
306 *(p_out+i) *= d_old_scale_factor;
311 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
312 <= (unsigned int)i_out+1 )
314 p_out += i_nb_channels;
316 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
322 /* Perform left-wing inner product */
323 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
324 SMALL_FILTER_NWING, p_in, p_out,
325 p_filter->p_sys->i_remainder,
326 p_filter->output.i_rate, p_filter->input.i_rate,
328 /* Perform right-wing inner product */
329 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
330 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
331 p_filter->output.i_rate -
332 p_filter->p_sys->i_remainder,
333 p_filter->output.i_rate, p_filter->input.i_rate,
337 p_out += i_nb_channels;
340 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
343 p_in += i_nb_channels;
344 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
347 /* Apply the new rate for the rest of the samples */
348 if( i_in < i_in_nb - i_filter_wing )
350 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
351 p_filter->p_sys->d_old_factor = d_factor;
352 p_filter->p_sys->i_old_wing = i_filter_wing;
354 for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
356 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
361 /* FilterFloatUP() is faster if we can use it */
363 /* Perform left-wing inner product */
364 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
365 SMALL_FILTER_NWING, p_in, p_out,
366 p_filter->p_sys->i_remainder,
367 p_filter->output.i_rate,
370 /* Perform right-wing inner product */
371 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
372 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
373 p_filter->output.i_rate -
374 p_filter->p_sys->i_remainder,
375 p_filter->output.i_rate,
379 /* Normalize for unity filter gain */
380 for( i = 0; i < i_nb_channels; i++ )
382 *(p_out+i) *= d_old_scale_factor;
386 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
387 <= (unsigned int)i_out+1 )
389 p_out += i_nb_channels;
391 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
397 /* Perform left-wing inner product */
398 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
399 SMALL_FILTER_NWING, p_in, p_out,
400 p_filter->p_sys->i_remainder,
401 p_filter->output.i_rate, p_filter->input.i_rate,
403 /* Perform right-wing inner product */
404 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
405 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
406 p_filter->output.i_rate -
407 p_filter->p_sys->i_remainder,
408 p_filter->output.i_rate, p_filter->input.i_rate,
412 p_out += i_nb_channels;
415 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
418 p_in += i_nb_channels;
419 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
422 /* Buffer i_filter_wing * 2 samples for next time */
423 if( p_filter->p_sys->i_old_wing )
425 memcpy( p_filter->p_sys->p_buf,
426 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
427 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
428 p_filter->input.i_bytes_per_frame );
432 msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
433 i_out * p_filter->input.i_bytes_per_frame );
436 /* Free the temp buffer */
441 /* Finalize aout buffer */
442 p_out_buf->i_nb_samples = i_out;
443 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
444 p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
445 p_out_buf->i_nb_samples );
447 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
448 i_nb_channels * sizeof(int32_t);
452 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
453 float *p_out, uint32_t ui_remainder,
454 uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
456 const float *Hp, *Hdp, *End;
458 uint32_t ui_linear_remainder;
461 Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
462 Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
466 ui_linear_remainder = (ui_remainder<<Nhc) -
467 (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
469 if (Inc == 1) /* If doing right wing... */
470 { /* ...drop extra coeff, so when Ph is */
471 End--; /* 0.5, we don't do too many mult's */
472 if (ui_remainder == 0) /* If the phase is zero... */
473 { /* ...then we've already skipped the */
474 Hp += Npc; /* first sample, so we must also */
475 Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
480 t = *Hp; /* Get filter coeff */
481 /* t is now interp'd filter coeff */
482 t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
483 for( i = 0; i < i_nb_channels; i++ )
486 temp *= *(p_in+i); /* Mult coeff by input sample */
487 *(p_out+i) += temp; /* The filter output */
489 Hdp += Npc; /* Filter coeff differences step */
490 Hp += Npc; /* Filter coeff step */
491 p_in += (Inc * i_nb_channels); /* Input signal step */
495 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
496 float *p_out, uint32_t ui_remainder,
497 uint32_t ui_output_rate, uint32_t ui_input_rate,
498 int16_t Inc, int i_nb_channels )
500 const float *Hp, *Hdp, *End;
502 uint32_t ui_linear_remainder;
503 int i, ui_counter = 0;
505 Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
506 Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
510 if (Inc == 1) /* If doing right wing... */
511 { /* ...drop extra coeff, so when Ph is */
512 End--; /* 0.5, we don't do too many mult's */
513 if (ui_remainder == 0) /* If the phase is zero... */
514 { /* ...then we've already skipped the */
515 Hp = Imp + /* first sample, so we must also */
516 (ui_output_rate << Nhc) / ui_input_rate;
517 Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
518 (ui_output_rate << Nhc) / ui_input_rate;
524 t = *Hp; /* Get filter coeff */
525 /* t is now interp'd filter coeff */
526 ui_linear_remainder =
527 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
528 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
529 ui_input_rate * ui_input_rate;
530 t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
531 for( i = 0; i < i_nb_channels; i++ )
534 temp *= *(p_in+i); /* Mult coeff by input sample */
535 *(p_out+i) += temp; /* The filter output */
540 /* Filter coeff step */
541 Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
543 /* Filter coeff differences step */
544 Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
547 p_in += (Inc * i_nb_channels); /* Input signal step */