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 *****************************************************************************/
51 static int Create ( vlc_object_t * );
52 static void Close ( vlc_object_t * );
53 static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
57 static int OpenFilter ( vlc_object_t * );
58 static void CloseFilter( vlc_object_t * );
59 static block_t *Resample( filter_t *, block_t * );
62 static void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing,
63 float *f_in, float *f_out, uint32_t ui_remainder,
64 uint32_t ui_output_rate, int16_t Inc,
67 static void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing,
68 float *f_in, float *f_out, uint32_t ui_remainder,
69 uint32_t ui_output_rate, uint32_t ui_input_rate,
70 int16_t Inc, int i_nb_channels );
72 /*****************************************************************************
74 *****************************************************************************/
77 int32_t *p_buf; /* this filter introduces a delay */
84 unsigned int i_remainder; /* remainder of previous sample */
92 /*****************************************************************************
94 *****************************************************************************/
96 set_category( CAT_AUDIO )
97 set_subcategory( SUBCAT_AUDIO_MISC )
98 set_description( N_("Audio filter for band-limited interpolation resampling") )
99 set_capability( "audio filter", 20 )
100 set_callbacks( Create, Close )
103 set_description( N_("Audio filter for band-limited interpolation resampling") )
104 set_capability( "audio filter2", 20 )
105 set_callbacks( OpenFilter, CloseFilter )
108 /*****************************************************************************
109 * Create: allocate linear resampler
110 *****************************************************************************/
111 static int Create( vlc_object_t *p_this )
113 aout_filter_t * p_filter = (aout_filter_t *)p_this;
114 struct filter_sys_t * p_sys;
118 if ( p_filter->input.i_rate == p_filter->output.i_rate
119 || p_filter->input.i_format != p_filter->output.i_format
120 || p_filter->input.i_physical_channels
121 != p_filter->output.i_physical_channels
122 || p_filter->input.i_original_channels
123 != p_filter->output.i_original_channels
124 || p_filter->input.i_format != VLC_CODEC_FL32 )
129 #if !defined( __APPLE__ )
130 if( !config_GetInt( p_this, "hq-resampling" ) )
136 /* Allocate the memory needed to store the module's structure */
137 p_sys = malloc( sizeof(filter_sys_t) );
140 p_filter->p_sys = (struct aout_filter_sys_t *)p_sys;
142 /* Calculate worst case for the length of the filter wing */
143 d_factor = (double)p_filter->output.i_rate
144 / p_filter->input.i_rate / AOUT_MAX_INPUT_RATE;
145 i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
146 * __MAX(1.0, 1.0/d_factor) + 10;
147 p_sys->i_buf_size = aout_FormatNbChannels( &p_filter->input ) *
148 sizeof(int32_t) * 2 * i_filter_wing;
150 /* Allocate enough memory to buffer previous samples */
151 p_sys->p_buf = malloc( p_sys->i_buf_size );
152 if( p_sys->p_buf == NULL )
158 p_sys->i_old_wing = 0;
159 p_sys->b_filter2 = false; /* It seams to be a good valuefor this module */
160 p_filter->pf_do_work = DoWork;
162 /* We don't want a new buffer to be created because we're not sure we'll
163 * actually need to resample anything. */
164 p_filter->b_in_place = true;
169 /*****************************************************************************
170 * Close: free our resources
171 *****************************************************************************/
172 static void Close( vlc_object_t * p_this )
174 aout_filter_t * p_filter = (aout_filter_t *)p_this;
175 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
176 free( p_sys->p_buf );
180 /*****************************************************************************
181 * DoWork: convert a buffer
182 *****************************************************************************/
183 static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
184 aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
186 filter_sys_t *p_sys = (filter_sys_t *)p_filter->p_sys;
187 float *p_out = (float *)p_out_buf->p_buffer;
189 int i_nb_channels = aout_FormatNbChannels( &p_filter->input );
190 int i_in_nb = p_in_buf->i_nb_samples;
192 unsigned int i_out_rate;
193 double d_factor, d_scale_factor, d_old_scale_factor;
196 if( p_sys->b_filter2 )
197 i_out_rate = p_filter->output.i_rate;
199 i_out_rate = p_aout->mixer_format.i_rate;
201 /* Check if we really need to run the resampler */
202 if( i_out_rate == p_filter->input.i_rate )
204 if( /*p_filter->b_continuity && /--* What difference does it make ? :) */
207 p_in_buf->i_nb_bytes + p_sys->i_old_wing *
208 p_filter->input.i_bytes_per_frame )
210 /* output the whole thing with the samples from last time */
211 memmove( ((float *)(p_in_buf->p_buffer)) +
212 i_nb_channels * p_sys->i_old_wing,
213 p_in_buf->p_buffer, p_in_buf->i_nb_bytes );
214 memcpy( p_in_buf->p_buffer, p_sys->p_buf +
215 i_nb_channels * p_sys->i_old_wing,
217 p_filter->input.i_bytes_per_frame );
219 p_out_buf->i_nb_samples = p_in_buf->i_nb_samples +
222 p_out_buf->start_date = date_Get( &p_sys->end_date );
223 p_out_buf->end_date =
224 date_Increment( &p_sys->end_date,
225 p_out_buf->i_nb_samples );
227 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
228 p_filter->input.i_bytes_per_frame;
230 p_filter->b_continuity = false;
231 p_sys->i_old_wing = 0;
235 if( !p_filter->b_continuity )
237 /* Continuity in sound samples has been broken, we'd better reset
239 p_filter->b_continuity = true;
240 p_sys->i_remainder = 0;
241 date_Init( &p_sys->end_date, i_out_rate, 1 );
242 date_Set( &p_sys->end_date, p_in_buf->start_date );
243 p_sys->i_old_rate = p_filter->input.i_rate;
244 p_sys->d_old_factor = 1;
245 p_sys->i_old_wing = 0;
249 msg_Err( p_filter, "old rate: %i, old factor: %f, old wing: %i, i_in: %i",
250 p_sys->i_old_rate, p_sys->d_old_factor,
251 p_sys->i_old_wing, i_in_nb );
254 /* Prepare the source buffer */
255 i_in_nb += (p_sys->i_old_wing * 2);
257 float p_in_orig[i_in_nb * p_filter->input.i_bytes_per_frame / 4],
260 /* Copy all our samples in p_in */
261 if( p_sys->i_old_wing )
263 vlc_memcpy( p_in, p_sys->p_buf,
264 p_sys->i_old_wing * 2 *
265 p_filter->input.i_bytes_per_frame );
267 vlc_memcpy( p_in + p_sys->i_old_wing * 2 * i_nb_channels,
269 p_in_buf->i_nb_samples * p_filter->input.i_bytes_per_frame );
271 /* Make sure the output buffer is reset */
272 memset( p_out, 0, p_out_buf->i_size );
274 /* Calculate the new length of the filter wing */
275 d_factor = (double)i_out_rate / p_filter->input.i_rate;
276 i_filter_wing = ((SMALL_FILTER_NMULT+1)/2.0) * __MAX(1.0,1.0/d_factor) + 1;
278 /* Account for increased filter gain when using factors less than 1 */
279 d_old_scale_factor = SMALL_FILTER_SCALE *
280 p_sys->d_old_factor + 0.5;
281 d_scale_factor = SMALL_FILTER_SCALE * d_factor + 0.5;
283 /* Apply the old rate until we have enough samples for the new one */
284 i_in = p_sys->i_old_wing;
285 p_in += p_sys->i_old_wing * i_nb_channels;
286 for( ; i_in < i_filter_wing &&
287 (i_in + p_sys->i_old_wing) < i_in_nb; i_in++ )
289 if( p_sys->d_old_factor == 1 )
291 /* Just copy the samples */
293 p_filter->input.i_bytes_per_frame );
294 p_in += i_nb_channels;
295 p_out += i_nb_channels;
300 while( p_sys->i_remainder < p_filter->output.i_rate )
303 if( p_sys->d_old_factor >= 1 )
305 /* FilterFloatUP() is faster if we can use it */
307 /* Perform left-wing inner product */
308 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
309 SMALL_FILTER_NWING, p_in, p_out,
311 p_filter->output.i_rate,
313 /* Perform right-wing inner product */
314 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
315 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
316 p_filter->output.i_rate -
318 p_filter->output.i_rate,
322 /* Normalize for unity filter gain */
323 for( i = 0; i < i_nb_channels; i++ )
325 *(p_out+i) *= d_old_scale_factor;
330 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
331 <= (unsigned int)i_out+1 )
333 p_out += i_nb_channels;
335 p_sys->i_remainder += p_filter->input.i_rate;
341 /* Perform left-wing inner product */
342 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
343 SMALL_FILTER_NWING, p_in, p_out,
345 p_filter->output.i_rate, p_filter->input.i_rate,
347 /* Perform right-wing inner product */
348 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
349 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
350 p_filter->output.i_rate -
352 p_filter->output.i_rate, p_filter->input.i_rate,
356 p_out += i_nb_channels;
359 p_sys->i_remainder += p_filter->input.i_rate;
362 p_in += i_nb_channels;
363 p_sys->i_remainder -= p_filter->output.i_rate;
366 /* Apply the new rate for the rest of the samples */
367 if( i_in < i_in_nb - i_filter_wing )
369 p_sys->i_old_rate = p_filter->input.i_rate;
370 p_sys->d_old_factor = d_factor;
371 p_sys->i_old_wing = i_filter_wing;
373 for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
375 while( p_sys->i_remainder < p_filter->output.i_rate )
380 /* FilterFloatUP() is faster if we can use it */
382 /* Perform left-wing inner product */
383 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
384 SMALL_FILTER_NWING, p_in, p_out,
386 p_filter->output.i_rate,
389 /* Perform right-wing inner product */
390 FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
391 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
392 p_filter->output.i_rate -
394 p_filter->output.i_rate,
398 /* Normalize for unity filter gain */
399 for( int i = 0; i < i_nb_channels; i++ )
401 *(p_out+i) *= d_old_scale_factor;
405 if( p_out_buf->i_size/p_filter->input.i_bytes_per_frame
406 <= (unsigned int)i_out+1 )
408 p_out += i_nb_channels;
410 p_sys->i_remainder += p_filter->input.i_rate;
416 /* Perform left-wing inner product */
417 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
418 SMALL_FILTER_NWING, p_in, p_out,
420 p_filter->output.i_rate, p_filter->input.i_rate,
422 /* Perform right-wing inner product */
423 FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
424 SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
425 p_filter->output.i_rate -
427 p_filter->output.i_rate, p_filter->input.i_rate,
431 p_out += i_nb_channels;
434 p_sys->i_remainder += p_filter->input.i_rate;
437 p_in += i_nb_channels;
438 p_sys->i_remainder -= p_filter->output.i_rate;
441 /* Buffer i_filter_wing * 2 samples for next time */
442 if( p_sys->i_old_wing )
444 memcpy( p_sys->p_buf,
445 p_in_orig + (i_in_nb - 2 * p_sys->i_old_wing) *
446 i_nb_channels, (2 * p_sys->i_old_wing) *
447 p_filter->input.i_bytes_per_frame );
451 msg_Err( p_filter, "p_out size: %i, nb bytes out: %i", p_out_buf->i_size,
452 i_out * p_filter->input.i_bytes_per_frame );
455 /* Finalize aout buffer */
456 p_out_buf->i_nb_samples = i_out;
457 p_out_buf->start_date = date_Get( &p_sys->end_date );
458 p_out_buf->end_date = date_Increment( &p_sys->end_date,
459 p_out_buf->i_nb_samples );
461 p_out_buf->i_nb_bytes = p_out_buf->i_nb_samples *
462 i_nb_channels * sizeof(int32_t);
466 /*****************************************************************************
468 *****************************************************************************/
469 static int OpenFilter( vlc_object_t *p_this )
471 filter_t *p_filter = (filter_t *)p_this;
473 unsigned int i_out_rate = p_filter->fmt_out.audio.i_rate;
477 if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
478 p_filter->fmt_in.i_codec != VLC_CODEC_FL32 )
483 #if !defined( SYS_DARWIN )
484 if( !config_GetInt( p_this, "hq-resampling" ) )
490 /* Allocate the memory needed to store the module's structure */
491 p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
495 /* Calculate worst case for the length of the filter wing */
496 d_factor = (double)i_out_rate / p_filter->fmt_in.audio.i_rate;
497 i_filter_wing = ((SMALL_FILTER_NMULT + 1)/2.0)
498 * __MAX(1.0, 1.0/d_factor) + 10;
499 p_filter->p_sys->i_buf_size = p_filter->fmt_in.audio.i_channels *
500 sizeof(int32_t) * 2 * i_filter_wing;
502 /* Allocate enough memory to buffer previous samples */
503 p_filter->p_sys->p_buf = malloc( p_filter->p_sys->i_buf_size );
504 if( p_filter->p_sys->p_buf == NULL )
510 p_filter->p_sys->i_old_wing = 0;
511 p_sys->b_first = true;
512 p_sys->b_filter2 = true;
513 p_filter->pf_audio_filter = Resample;
515 msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
516 (char *)&p_filter->fmt_in.i_codec,
517 p_filter->fmt_in.audio.i_rate,
518 p_filter->fmt_in.audio.i_channels,
519 (char *)&p_filter->fmt_out.i_codec,
520 p_filter->fmt_out.audio.i_rate,
521 p_filter->fmt_out.audio.i_channels);
523 p_filter->fmt_out = p_filter->fmt_in;
524 p_filter->fmt_out.audio.i_rate = i_out_rate;
529 /*****************************************************************************
530 * CloseFilter : deallocate data structures
531 *****************************************************************************/
532 static void CloseFilter( vlc_object_t *p_this )
534 filter_t *p_filter = (filter_t *)p_this;
535 free( p_filter->p_sys->p_buf );
536 free( p_filter->p_sys );
539 /*****************************************************************************
541 *****************************************************************************/
542 static block_t *Resample( filter_t *p_filter, block_t *p_block )
544 aout_filter_t aout_filter;
545 aout_buffer_t in_buf, out_buf;
548 int i_bytes_per_frame;
550 if( !p_block || !p_block->i_samples )
553 block_Release( p_block );
557 i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
558 p_filter->fmt_out.audio.i_bitspersample / 8;
560 i_out_size = i_bytes_per_frame * ( 1 + ( p_block->i_samples *
561 p_filter->fmt_out.audio.i_rate /
562 p_filter->fmt_in.audio.i_rate) ) +
563 p_filter->p_sys->i_buf_size;
565 p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
568 msg_Warn( p_filter, "can't get output buffer" );
569 block_Release( p_block );
573 p_out->i_samples = i_out_size / i_bytes_per_frame;
574 p_out->i_dts = p_block->i_dts;
575 p_out->i_pts = p_block->i_pts;
576 p_out->i_length = p_block->i_length;
578 aout_filter.p_sys = (struct aout_filter_sys_t *)p_filter->p_sys;
579 aout_filter.input = p_filter->fmt_in.audio;
580 aout_filter.input.i_bytes_per_frame = p_filter->fmt_in.audio.i_channels *
581 p_filter->fmt_in.audio.i_bitspersample / 8;
582 aout_filter.output = p_filter->fmt_out.audio;
583 aout_filter.output.i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
584 p_filter->fmt_out.audio.i_bitspersample / 8;
585 aout_filter.b_continuity = !p_filter->p_sys->b_first;
586 p_filter->p_sys->b_first = false;
588 in_buf.p_buffer = p_block->p_buffer;
589 in_buf.i_nb_bytes = in_buf.i_size = p_block->i_buffer;
590 in_buf.i_nb_samples = p_block->i_samples;
591 out_buf.p_buffer = p_out->p_buffer;
592 out_buf.i_nb_bytes = out_buf.i_size = p_out->i_buffer;
593 out_buf.i_nb_samples = p_out->i_samples;
595 DoWork( (aout_instance_t *)p_filter, &aout_filter, &in_buf, &out_buf );
597 block_Release( p_block );
599 p_out->i_buffer = out_buf.i_nb_bytes;
600 p_out->i_samples = out_buf.i_nb_samples;
605 void FilterFloatUP( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
606 float *p_out, uint32_t ui_remainder,
607 uint32_t ui_output_rate, int16_t Inc, int i_nb_channels )
609 const float *Hp, *Hdp, *End;
611 uint32_t ui_linear_remainder;
614 Hp = &Imp[(ui_remainder<<Nhc)/ui_output_rate];
615 Hdp = &ImpD[(ui_remainder<<Nhc)/ui_output_rate];
619 ui_linear_remainder = (ui_remainder<<Nhc) -
620 (ui_remainder<<Nhc)/ui_output_rate*ui_output_rate;
622 if (Inc == 1) /* If doing right wing... */
623 { /* ...drop extra coeff, so when Ph is */
624 End--; /* 0.5, we don't do too many mult's */
625 if (ui_remainder == 0) /* If the phase is zero... */
626 { /* ...then we've already skipped the */
627 Hp += Npc; /* first sample, so we must also */
628 Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */
633 t = *Hp; /* Get filter coeff */
634 /* t is now interp'd filter coeff */
635 t += *Hdp * ui_linear_remainder / ui_output_rate / Npc;
636 for( i = 0; i < i_nb_channels; i++ )
639 temp *= *(p_in+i); /* Mult coeff by input sample */
640 *(p_out+i) += temp; /* The filter output */
642 Hdp += Npc; /* Filter coeff differences step */
643 Hp += Npc; /* Filter coeff step */
644 p_in += (Inc * i_nb_channels); /* Input signal step */
648 void FilterFloatUD( const float Imp[], const float ImpD[], uint16_t Nwing, float *p_in,
649 float *p_out, uint32_t ui_remainder,
650 uint32_t ui_output_rate, uint32_t ui_input_rate,
651 int16_t Inc, int i_nb_channels )
653 const float *Hp, *Hdp, *End;
655 uint32_t ui_linear_remainder;
656 int i, ui_counter = 0;
658 Hp = Imp + (ui_remainder<<Nhc) / ui_input_rate;
659 Hdp = ImpD + (ui_remainder<<Nhc) / ui_input_rate;
663 if (Inc == 1) /* If doing right wing... */
664 { /* ...drop extra coeff, so when Ph is */
665 End--; /* 0.5, we don't do too many mult's */
666 if (ui_remainder == 0) /* If the phase is zero... */
667 { /* ...then we've already skipped the */
668 Hp = Imp + /* first sample, so we must also */
669 (ui_output_rate << Nhc) / ui_input_rate;
670 Hdp = ImpD + /* skip ahead in Imp[] and ImpD[] */
671 (ui_output_rate << Nhc) / ui_input_rate;
677 t = *Hp; /* Get filter coeff */
678 /* t is now interp'd filter coeff */
679 ui_linear_remainder =
680 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) -
681 ((ui_output_rate * ui_counter + ui_remainder)<< Nhc) /
682 ui_input_rate * ui_input_rate;
683 t += *Hdp * ui_linear_remainder / ui_input_rate / Npc;
684 for( i = 0; i < i_nb_channels; i++ )
687 temp *= *(p_in+i); /* Mult coeff by input sample */
688 *(p_out+i) += temp; /* The filter output */
693 /* Filter coeff step */
694 Hp = Imp + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
696 /* Filter coeff differences step */
697 Hdp = ImpD + ((ui_output_rate * ui_counter + ui_remainder)<< Nhc)
700 p_in += (Inc * i_nb_channels); /* Input signal step */