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>
43 #include <vlc_filter.h>
44 #include <vlc_block.h>
46 #include "bandlimited.h"
48 /*****************************************************************************
50 *****************************************************************************/
53 static int OpenFilter ( vlc_object_t * );
54 static void CloseFilter( vlc_object_t * );
55 static block_t *Resample( filter_t *, block_t * );
58 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
59 float *f_in, float *f_out, uint32_t ui_remainder,
60 uint32_t ui_output_rate, int16_t Inc,
63 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
64 float *f_in, float *f_out, uint32_t ui_remainder,
65 uint32_t ui_output_rate, uint32_t ui_input_rate,
66 int16_t Inc, int i_nb_channels );
68 /*****************************************************************************
70 *****************************************************************************/
73 int32_t *p_buf; /* this filter introduces a delay */
79 unsigned int i_remainder; /* remainder of previous sample */
85 /*****************************************************************************
87 *****************************************************************************/
89 set_category( CAT_AUDIO )
90 set_subcategory( SUBCAT_AUDIO_MISC )
91 set_description( N_("Audio filter for band-limited interpolation resampling") )
92 set_capability( "audio filter2", 20 )
93 set_callbacks( OpenFilter, CloseFilter )
96 /*****************************************************************************
97 * Resample: convert a buffer
98 *****************************************************************************/
99 static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
101 if( !p_in_buf || !p_in_buf->i_nb_samples )
104 block_Release( p_in_buf );
108 filter_sys_t *p_sys = p_filter->p_sys;
109 unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
110 int i_nb_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
112 /* Check if we really need to run the resampler */
113 if( i_out_rate == p_filter->fmt_in.audio.i_rate )
115 if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
118 /* output the whole thing with the samples from last time */
119 p_in_buf = block_Realloc( p_in_buf,
120 p_sys->i_old_wing * p_filter->fmt_in.audio.i_bytes_per_frame,
121 p_in_buf->i_buffer );
124 memcpy( p_in_buf->p_buffer, p_sys->p_buf +
125 i_nb_channels * p_sys->i_old_wing,
127 p_filter->fmt_in.audio.i_bytes_per_frame );
129 p_in_buf->i_nb_samples += p_sys->i_old_wing;
131 p_in_buf->i_pts = date_Get( &p_sys->end_date );
133 date_Increment( &p_sys->end_date,
134 p_in_buf->i_nb_samples ) - p_in_buf->i_pts;
136 p_in_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
137 p_sys->i_old_wing = 0;
141 unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
142 p_filter->fmt_out.audio.i_bitspersample / 8;
143 size_t i_out_size = i_bytes_per_frame * ( 1 + ( p_in_buf->i_nb_samples *
144 p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate) )
145 + p_filter->p_sys->i_buf_size;
146 block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
149 float *p_out = (float *)p_out_buf->p_buffer;
151 if( (p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) || p_sys->b_first )
153 /* Continuity in sound samples has been broken, we'd better reset
155 p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
156 p_sys->i_remainder = 0;
157 date_Init( &p_sys->end_date, i_out_rate, 1 );
158 date_Set( &p_sys->end_date, p_in_buf->i_pts );
159 p_sys->d_old_factor = 1;
160 p_sys->i_old_wing = 0;
161 p_sys->b_first = false;
164 int i_in_nb = p_in_buf->i_nb_samples;
166 double d_factor, d_scale_factor, d_old_scale_factor;
170 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
171 p_sys->i_old_rate, p_sys->d_old_factor,
172 p_sys->i_old_wing, i_in_nb );
175 /* Prepare the source buffer */
176 i_in_nb += (p_sys->i_old_wing * 2);
178 float p_in_orig[i_in_nb * p_filter->fmt_in.audio.i_bytes_per_frame / 4],
181 /* Copy all our samples in p_in */
182 if( p_sys->i_old_wing )
184 vlc_memcpy( p_in, p_sys->p_buf,
185 p_sys->i_old_wing * 2 *
186 p_filter->fmt_in.audio.i_bytes_per_frame );
188 /* XXX: why i_nb_channels instead of i_bytes_per_frame??? */
189 vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
191 p_in_buf->i_nb_samples * p_filter->fmt_in.audio.i_bytes_per_frame );
192 block_Release( p_in_buf );
194 /* Make sure the output buffer is reset */
195 memset( p_out, 0, p_out_buf->i_buffer );
197 /* Calculate the new length of the filter wing */
198 d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
199 i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
201 /* Account for increased filter gain when using factors less than 1 */
202 d_old_scale_factor = SMALL_FILTER_SCALE *
203 p_sys->d_old_factor + 0.5;
204 d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
206 /* Apply the old rate until we have enough samples for the new one */
207 i_in = p_sys->i_old_wing;
208 p_in += p_sys->i_old_wing * i_nb_channels;
209 for( ; i_in < i_filter_wing &&
210 (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
212 if( p_sys->d_old_factor == 1 )
214 /* Just copy the samples */
216 p_filter->fmt_in.audio.i_bytes_per_frame );
217 p_in += i_nb_channels;
218 p_out += i_nb_channels;
223 while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
226 if( p_sys->d_old_factor >= 1 )
228 /* FilterFloatUP() is faster if we can use it */
230 /* Perform left-wing inner product */
231 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
232 SMALL_FILTER_NWING, p_in, p_out,
234 p_filter->fmt_out.audio.i_rate,
236 /* Perform right-wing inner product */
237 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
238 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
239 p_filter->fmt_out.audio.i_rate -
241 p_filter->fmt_out.audio.i_rate,
245 /* Normalize for unity filter gain */
246 for( i = 0; i < i_nb_channels; i++ )
248 *(p_out+i) *= d_old_scale_factor;
253 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
254 <= (unsigned int)i_out+1 )
256 p_out += i_nb_channels;
258 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
264 /* Perform left-wing inner product */
265 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
266 SMALL_FILTER_NWING, p_in, p_out,
268 p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
270 /* Perform right-wing inner product */
271 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
272 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
273 p_filter->fmt_out.audio.i_rate -
275 p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
279 p_out += i_nb_channels;
282 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
285 p_in += i_nb_channels;
286 p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
289 /* Apply the new rate for the rest of the samples */
290 if( i_in < i_in_nb - i_filter_wing )
292 p_sys->d_old_factor = d_factor;
293 p_sys->i_old_wing = i_filter_wing;
295 for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
297 while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
302 /* FilterFloatUP() is faster if we can use it */
304 /* Perform left-wing inner product */
305 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
306 SMALL_FILTER_NWING, p_in, p_out,
308 p_filter->fmt_out.audio.i_rate,
311 /* Perform right-wing inner product */
312 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
313 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
314 p_filter->fmt_out.audio.i_rate -
316 p_filter->fmt_out.audio.i_rate,
320 /* Normalize for unity filter gain */
321 for( int i = 0; i < i_nb_channels; i++ )
323 *(p_out+i) *= d_old_scale_factor;
327 if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
328 <= (unsigned int)i_out+1 )
330 p_out += i_nb_channels;
332 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
338 /* Perform left-wing inner product */
339 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
340 SMALL_FILTER_NWING, p_in, p_out,
342 p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
344 /* Perform right-wing inner product */
345 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
346 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
347 p_filter->fmt_out.audio.i_rate -
349 p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
353 p_out += i_nb_channels;
356 p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
359 p_in += i_nb_channels;
360 p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
363 /* Buffer i_filter_wing * 2 samples for next time */
364 if( p_sys->i_old_wing )
366 memcpy( p_sys->p_buf,
367 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
368 i_nb_channels, (2 * p_sys->i_old_wing) *
369 p_filter->fmt_in.audio.i_bytes_per_frame );
373 msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_buffer,
374 i_out * p_filter->fmt_in.audio.i_bytes_per_frame );
377 /* Finalize aout buffer */
378 p_out_buf->i_nb_samples = i_out;
379 p_out_buf->i_pts = date_Get( &p_sys->end_date );
380 p_out_buf->i_length = date_Increment( &p_sys->end_date,
381 p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
383 p_out_buf->i_buffer = p_out_buf->i_nb_samples *
384 i_nb_channels * sizeof(int32_t);
388 /*****************************************************************************
390 *****************************************************************************/
391 static int OpenFilter( vlc_object_t *p_this )
393 filter_t *p_filter = (filter_t *)p_this;
395 unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
399 if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
400 p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
405 #if !defined( SYS_DARWIN )
406 if( !config_GetInt( p_this, "hq-resampling" ) )
412 /* Allocate the memory needed to store the module's structure */
413 p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
417 /* Calculate worst case for the length of the filter wing */
418 d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
419 i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
420 * __MAX(1.0, 1.0/d_factor) + 10;
421 p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
422 sizeof(int32_t) * 2 * i_filter_wing;
424 /* Allocate enough memory to buffer previous samples */
425 p_sys->p_buf = malloc( p_sys->i_buf_size );
426 if( p_sys->p_buf == NULL )
432 p_sys->i_old_wing = 0;
433 p_sys->b_first = true;
434 p_filter->pf_audio_filter = Resample;
436 msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
437 (char *)&p_filter->fmt_in.i_codec,
438 p_filter->fmt_in.audio.i_rate,
439 p_filter->fmt_in.audio.i_channels,
440 (char *)&p_filter->fmt_out.i_codec,
441 p_filter->fmt_out.audio.i_rate,
442 p_filter->fmt_out.audio.i_channels);
444 p_filter->fmt_out = p_filter->fmt_in;
445 p_filter->fmt_out.audio.i_rate = i_out_rate;
450 /*****************************************************************************
451 * CloseFilter : deallocate data structures
452 *****************************************************************************/
453 static void CloseFilter( vlc_object_t *p_this )
455 filter_t *p_filter = (filter_t *)p_this;
456 free( p_filter->p_sys->p_buf );
457 free( p_filter->p_sys );
460 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
461 float *p_out, uint32_t ui_remainder,
462 uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
464 const float *Hp, *Hdp, *End;
466 uint32_t ui_linear_remainder;
469 Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
470 Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
474 ui_linear_remainder = (ui_remainder<<Nhc) -
475 (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
477 if (Inc == 1) /* If doing right wing... */
478 { /* ...drop extra coeff, so when Ph is */
479 End--; /* 0.5, we don't do too many mult's */
480 if (ui_remainder == 0) /* If the phase is zero... */
481 { /* ...then we've already skipped the */
482 Hp += Npc; /* first sample, so we must also */
483 Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
488 t = *Hp; /* Get filter coeff */
489 /* t is now interp'd filter coeff */
490 t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
491 for( i = 0; i < i_nb_channels; i++ )
494 temp *= *(p_in+i); /* Mult coeff by input sample */
495 *(p_out+i) += temp; /* The filter output */
497 Hdp += Npc; /* Filter coeff differences step */
498 Hp += Npc; /* Filter coeff step */
499 p_in += (Inc * i_nb_channels); /* Input signal step */
503 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
504 float *p_out, uint32_t ui_remainder,
505 uint32_t ui_output_rate, uint32_t ui_input_rate,
506 int16_t Inc, int i_nb_channels )
508 const float *Hp, *Hdp, *End;
510 uint32_t ui_linear_remainder;
511 int i, ui_counter = 0;
513 Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
514 Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
518 if (Inc == 1) /* If doing right wing... */
519 { /* ...drop extra coeff, so when Ph is */
520 End--; /* 0.5, we don't do too many mult's */
521 if (ui_remainder == 0) /* If the phase is zero... */
522 { /* ...then we've already skipped the */
523 Hp = Imp + /* first sample, so we must also */
524 (ui_output_rate << Nhc) / ui_input_rate;
525 Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
526 (ui_output_rate << Nhc) / ui_input_rate;
532 t = *Hp; /* Get filter coeff */
533 /* t is now interp'd filter coeff */
534 ui_linear_remainder =
535 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
536 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
537 ui_input_rate * ui_input_rate;
538 t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
539 for( i = 0; i < i_nb_channels; i++ )
542 temp *= *(p_in+i); /* Mult coeff by input sample */
543 *(p_out+i) += temp; /* The filter output */
548 /* Filter coeff step */
549 Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
551 /* Filter coeff differences step */
552 Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
555 p_in += (Inc * i_nb_channels); /* Input signal step */