1 /*****************************************************************************
2 * bandlimited.c : band-limited interpolation resampler
3 *****************************************************************************
4 * Copyright (C) 2002 VideoLAN
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., 59 Temple Place - Suite 330, Boston, MA 02111, 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 *****************************************************************************/
35 #include <stdlib.h> /* malloc(), free() */
39 #include "audio_output.h"
40 #include "aout_internal.h"
41 #include "bandlimited.h"
43 /*****************************************************************************
45 *****************************************************************************/
46 static int Create ( vlc_object_t * );
47 static void Close ( vlc_object_t * );
48 static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
51 static void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing,
52 float *f_in, float *f_out, uint32_t ui_remainder,
53 uint32_t ui_output_rate, int16_t Inc,
56 static void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing,
57 float *f_in, float *f_out, uint32_t ui_remainder,
58 uint32_t ui_output_rate, uint32_t ui_input_rate,
59 int16_t Inc, int i_nb_channels );
61 /*****************************************************************************
63 *****************************************************************************/
64 struct aout_filter_sys_t
66 int32_t *p_buf; /* this filter introduces a delay */
73 unsigned int i_remainder; /* remainder of previous sample */
75 audio_date_t end_date;
78 /*****************************************************************************
80 *****************************************************************************/
82 set_category( CAT_AUDIO );
83 set_subcategory( SUBCAT_AUDIO_MISC );
84 set_description( _("audio filter for band-limited interpolation resampling") );
85 set_capability( "audio filter", 20 );
86 set_callbacks( Create, Close );
89 /*****************************************************************************
90 * Create: allocate linear resampler
91 *****************************************************************************/
92 static int Create( vlc_object_t *p_this )
94 aout_filter_t * p_filter = (aout_filter_t *)p_this;
98 if ( p_filter->input.i_rate == p_filter->output.i_rate
99 || p_filter->input.i_format != p_filter->output.i_format
100 || p_filter->input.i_physical_channels
101 != p_filter->output.i_physical_channels
102 || p_filter->input.i_original_channels
103 != p_filter->output.i_original_channels
104 || p_filter->input.i_format != VLC_FOURCC('f','l','3','2') )
109 #if !defined( SYS_DARWIN )
110 if( !config_GetInt( p_this, "hq-resampling" ) )
116 /* Allocate the memory needed to store the module's structure */
117 p_filter->p_sys = malloc( sizeof(struct aout_filter_sys_t) );
118 if( p_filter->p_sys == NULL )
120 msg_Err( p_filter, "out of memory" );
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;
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 )
136 msg_Err( p_filter, "out of memory" );
140 p_filter->p_sys->i_old_wing = 0;
141 p_filter->pf_do_work = DoWork;
143 /* We don't want a new buffer to be created because we're not sure we'll
144 * actually need to resample anything. */
145 p_filter->b_in_place = VLC_TRUE;
150 /*****************************************************************************
151 * Close: free our resources
152 *****************************************************************************/
153 static void Close( vlc_object_t * p_this )
155 aout_filter_t * p_filter = (aout_filter_t *)p_this;
156 free( p_filter->p_sys->p_buf );
157 free( p_filter->p_sys );
160 /*****************************************************************************
161 * DoWork: convert a buffer
162 *****************************************************************************/
163 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
164 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
166 float *p_in, *p_in_orig, *p_out = (float *)p_out_buf->p_buffer;
168 int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
169 int i_in_nb = p_in_buf->i_nb_samples;
171 double d_factor, d_scale_factor, d_old_scale_factor;
177 /* Check if we really need to run the resampler */
178 if( p_aout->mixer.mixer.i_rate == p_filter->input.i_rate )
180 if( //p_filter->b_continuity && /* What difference does it make ? :) */
181 p_filter->p_sys->i_old_wing &&
183 p_in_buf->i_nb_bytes + p_filter->p_sys->i_old_wing *
184 p_filter->input.i_bytes_per_frame )
186 /* output the whole thing with the samples from last time */
187 memmove( ((float *)(p_in_buf->p_buffer)) +
188 i_nb_channels * p_filter->p_sys->i_old_wing,
189 p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
190 memcpy( p_in_buf->p_buffer, p_filter->p_sys->p_buf +
191 i_nb_channels * p_filter->p_sys->i_old_wing,
192 p_filter->p_sys->i_old_wing *
193 p_filter->input.i_bytes_per_frame );
195 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
196 p_filter->p_sys->i_old_wing;
198 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
199 p_out_buf->end_date =
200 aout_DateIncrement( &p_filter->p_sys->end_date,
201 p_out_buf->i_nb_samples );
203 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
204 p_filter->input.i_bytes_per_frame;
206 p_filter->b_continuity = VLC_FALSE;
207 p_filter->p_sys->i_old_wing = 0;
211 if( !p_filter->b_continuity )
213 /* Continuity in sound samples has been broken, we'd better reset
215 p_filter->b_continuity = VLC_TRUE;
216 p_filter->p_sys->i_remainder = 0;
217 aout_DateInit( &p_filter->p_sys->end_date, p_filter->output.i_rate );
218 aout_DateSet( &p_filter->p_sys->end_date, p_in_buf->start_date );
219 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
220 p_filter->p_sys->d_old_factor = 1;
221 p_filter->p_sys->i_old_wing = 0;
225 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
226 p_filter->p_sys->i_old_rate, p_filter->p_sys->d_old_factor,
227 p_filter->p_sys->i_old_wing, i_in_nb );
230 /* Prepare the source buffer */
231 i_in_nb += (p_filter->p_sys->i_old_wing * 2);
233 p_in = p_in_orig = (float *)alloca( i_in_nb *
234 p_filter->input.i_bytes_per_frame );
236 p_in = p_in_orig = (float *)malloc( i_in_nb *
237 p_filter->input.i_bytes_per_frame );
244 /* Copy all our samples in p_in */
245 if( p_filter->p_sys->i_old_wing )
247 p_aout->p_vlc->pf_memcpy( p_in, p_filter->p_sys->p_buf,
248 p_filter->p_sys->i_old_wing * 2 *
249 p_filter->input.i_bytes_per_frame );
251 p_aout->p_vlc->pf_memcpy( p_in + p_filter->p_sys->i_old_wing * 2 *
252 i_nb_channels, p_in_buf->p_buffer,
253 p_in_buf->i_nb_samples *
254 p_filter->input.i_bytes_per_frame );
256 /* Make sure the output buffer is reset */
257 memset( p_out, 0, p_out_buf->i_size );
259 /* Calculate the new length of the filter wing */
260 d_factor = (double)p_aout->mixer.mixer.i_rate / p_filter->input.i_rate;
261 i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
263 /* Account for increased filter gain when using factors less than 1 */
264 d_old_scale_factor = SMALL_FILTER_SCALE *
265 p_filter->p_sys->d_old_factor + 0.5;
266 d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
268 /* Apply the old rate until we have enough samples for the new one */
269 i_in = p_filter->p_sys->i_old_wing;
270 p_in += p_filter->p_sys->i_old_wing * i_nb_channels;
271 for( ; i_in < i_filter_wing &&
272 (i_in + p_filter->p_sys->i_old_wing) < i_in_nb; i_in++ )
274 if( p_filter->p_sys->d_old_factor == 1 )
276 /* Just copy the samples */
278 p_filter->input.i_bytes_per_frame );
279 p_in += i_nb_channels;
280 p_out += i_nb_channels;
285 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
288 if( p_filter->p_sys->d_old_factor >= 1 )
290 /* FilterFloatUP() is faster if we can use it */
292 /* Perform left-wing inner product */
293 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
294 SMALL_FILTER_NWING, p_in, p_out,
295 p_filter->p_sys->i_remainder,
296 p_filter->output.i_rate,
298 /* Perform right-wing inner product */
299 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
300 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
301 p_filter->output.i_rate -
302 p_filter->p_sys->i_remainder,
303 p_filter->output.i_rate,
307 /* Normalize for unity filter gain */
308 for( i = 0; i < i_nb_channels; i++ )
310 *(p_out+i) *= d_old_scale_factor;
315 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
316 <= (unsigned int)i_out+1 )
318 p_out += i_nb_channels;
320 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
326 /* Perform left-wing inner product */
327 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
328 SMALL_FILTER_NWING, p_in, p_out,
329 p_filter->p_sys->i_remainder,
330 p_filter->output.i_rate, p_filter->input.i_rate,
332 /* Perform right-wing inner product */
333 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
334 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
335 p_filter->output.i_rate -
336 p_filter->p_sys->i_remainder,
337 p_filter->output.i_rate, p_filter->input.i_rate,
341 p_out += i_nb_channels;
344 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
347 p_in += i_nb_channels;
348 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
351 /* Apply the new rate for the rest of the samples */
352 if( i_in < i_in_nb - i_filter_wing )
354 p_filter->p_sys->i_old_rate = p_filter->input.i_rate;
355 p_filter->p_sys->d_old_factor = d_factor;
356 p_filter->p_sys->i_old_wing = i_filter_wing;
358 for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
360 while( p_filter->p_sys->i_remainder < p_filter->output.i_rate )
365 /* FilterFloatUP() is faster if we can use it */
367 /* Perform left-wing inner product */
368 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
369 SMALL_FILTER_NWING, p_in, p_out,
370 p_filter->p_sys->i_remainder,
371 p_filter->output.i_rate,
374 /* Perform right-wing inner product */
375 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
376 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
377 p_filter->output.i_rate -
378 p_filter->p_sys->i_remainder,
379 p_filter->output.i_rate,
383 /* Normalize for unity filter gain */
384 for( i = 0; i < i_nb_channels; i++ )
386 *(p_out+i) *= d_old_scale_factor;
390 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
391 <= (unsigned int)i_out+1 )
393 p_out += i_nb_channels;
395 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
401 /* Perform left-wing inner product */
402 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
403 SMALL_FILTER_NWING, p_in, p_out,
404 p_filter->p_sys->i_remainder,
405 p_filter->output.i_rate, p_filter->input.i_rate,
407 /* Perform right-wing inner product */
408 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
409 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
410 p_filter->output.i_rate -
411 p_filter->p_sys->i_remainder,
412 p_filter->output.i_rate, p_filter->input.i_rate,
416 p_out += i_nb_channels;
419 p_filter->p_sys->i_remainder += p_filter->input.i_rate;
422 p_in += i_nb_channels;
423 p_filter->p_sys->i_remainder -= p_filter->output.i_rate;
426 /* Buffer i_filter_wing * 2 samples for next time */
427 if( p_filter->p_sys->i_old_wing )
429 memcpy( p_filter->p_sys->p_buf,
430 p_in_orig + (i_in_nb - 2 * p_filter->p_sys->i_old_wing) *
431 i_nb_channels, (2 * p_filter->p_sys->i_old_wing) *
432 p_filter->input.i_bytes_per_frame );
436 msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
437 i_out * p_filter->input.i_bytes_per_frame );
440 /* Free the temp buffer */
445 /* Finalize aout buffer */
446 p_out_buf->i_nb_samples = i_out;
447 p_out_buf->start_date = aout_DateGet( &p_filter->p_sys->end_date );
448 p_out_buf->end_date = aout_DateIncrement( &p_filter->p_sys->end_date,
449 p_out_buf->i_nb_samples );
451 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
452 i_nb_channels * sizeof(int32_t);
456 void FilterFloatUP( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
457 float *p_out, uint32_t ui_remainder,
458 uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
460 float *Hp, *Hdp, *End;
462 uint32_t ui_linear_remainder;
465 Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
466 Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
470 ui_linear_remainder = (ui_remainder<<Nhc) -
471 (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
473 if (Inc == 1) /* If doing right wing... */
474 { /* ...drop extra coeff, so when Ph is */
475 End--; /* 0.5, we don't do too many mult's */
476 if (ui_remainder == 0) /* If the phase is zero... */
477 { /* ...then we've already skipped the */
478 Hp += Npc; /* first sample, so we must also */
479 Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
484 t = *Hp; /* Get filter coeff */
485 /* t is now interp'd filter coeff */
486 t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
487 for( i = 0; i < i_nb_channels; i++ )
490 temp *= *(p_in+i); /* Mult coeff by input sample */
491 *(p_out+i) += temp; /* The filter output */
493 Hdp += Npc; /* Filter coeff differences step */
494 Hp += Npc; /* Filter coeff step */
495 p_in += (Inc * i_nb_channels); /* Input signal step */
499 void FilterFloatUD( float Imp[], float ImpD[], uint16_t Nwing, float *p_in,
500 float *p_out, uint32_t ui_remainder,
501 uint32_t ui_output_rate, uint32_t ui_input_rate,
502 int16_t Inc, int i_nb_channels )
504 float *Hp, *Hdp, *End;
506 uint32_t ui_linear_remainder;
507 int i, ui_counter = 0;
509 Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
510 Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
514 if (Inc == 1) /* If doing right wing... */
515 { /* ...drop extra coeff, so when Ph is */
516 End--; /* 0.5, we don't do too many mult's */
517 if (ui_remainder == 0) /* If the phase is zero... */
518 { /* ...then we've already skipped the */
519 Hp = Imp + /* first sample, so we must also */
520 (ui_output_rate << Nhc) / ui_input_rate;
521 Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
522 (ui_output_rate << Nhc) / ui_input_rate;
528 t = *Hp; /* Get filter coeff */
529 /* t is now interp'd filter coeff */
530 ui_linear_remainder =
531 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
532 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
533 ui_input_rate * ui_input_rate;
534 t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
535 for( i = 0; i < i_nb_channels; i++ )
538 temp *= *(p_in+i); /* Mult coeff by input sample */
539 *(p_out+i) += temp; /* The filter output */
544 /* Filter coeff step */
545 Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
547 /* Filter coeff differences step */
548 Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
551 p_in += (Inc * i_nb_channels); /* Input signal step */