-
Notifications
You must be signed in to change notification settings - Fork 1
/
alt_audio_convert.c
1057 lines (911 loc) · 31.7 KB
/
alt_audio_convert.c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* Extended Audio Converter for SDL (Simple DirectMedia Layer)
* Copyright (C) 2002 Frank Ranostaj
* Institute of Applied Physik
* Johann Wolfgang Goethe-Universität
* Frankfurt am Main, Germany
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* Frank Ranostaj
* ranostaj@stud.uni-frankfurt.de
*
* (This code blatantly abducted for SDL_sound. Thanks, Frank! --ryan.)
*/
#if HAVE_CONFIG_H
# include <config.h>
#endif
#if SOUND_USE_ALTCVT
#include "alt_audio_convert.h"
#include <math.h>
/* just to make sure this is defined... */
#ifndef min
#define min(x, y) ( ((x) < (y)) ? (x) : (y) )
#endif
#ifndef max
#define max(x, y) ( ((x) > (y)) ? (x) : (y) )
#endif
#ifndef abs
#define abs(x) ( ((x) > (0)) ? (x) : -(x) )
#endif
/* some macros for "parsing" format */
#define IS_8BIT(x) ((x).format & 0x0008)
#define IS_16BIT(x) ((x).format & 0x0010)
#define IS_FLOAT(x) ((x).format & 0x0020)
#define IS_SIGNED(x) ((x).format & 0x8000)
#define IS_SYSENDIAN(x) ((~AUDIO_U16SYS ^ (x).format) & 0x1000)
#define SDL_MSB_POSITION_IN_SHORT ((0x1000 & AUDIO_U16SYS)>>12)
/*-------------------------------------------------------------------------*/
/* the purpose of the RateConverterBuffer is to provide a continous storage
for head and tail of the (sample)-buffer. This allows a simple and
perfomant implemantation of the sample rate converters. Depending of the
operation mode, two layouts for the RateConverterBuffer.inbuffer are
possible:
in the Loop Mode:
... T-4 T-3 T-2 T-1 H+0 H+1 H+2 H+3 H+4 ...
|
linp, finp
in the Single Mode (non Loop):
... T-4 T-3 T-2 T-1 0 0 0 ... 0 0 0 H+0 H+1 H+2 H+3 H+4 ...
| |
linp finp
The RateConverterBuffer allows an accurate attack and decay of the
filters in the rate Converters.
The pointer finp are actually shifted against the depicted position so
that on the first invocation of the rate converter the input of the
filter is nearly complete in the zero region, only one input value is
used. After the calculation of the first output value, the pointer are
incremented or decremented depending on down or up conversion and the
first two input value are taken into account. This procedure repeats
until the filter has processed all zeroes. The distance of the pointer
movement is stored in flength, always positive.
Further a pointer cinp to the sample buffer itself is stored. The pointer
to the sample buffer is shifted too, so that on the first use of this
pointer the filter is complete in the sample buffer. The pointer moves
over the sample buffer until it reaches the other end. The distance of
the movement is stored in clength.
Finally the decay of the filter is done by linp and llength like finp,
flength, but in reverse order.
buffer denotes the start or the end of the output buffer, depending
on direction of the rate conversion.
All pointer and length referring the buffer as Sint16. All length
are refering to the input buffer */
typedef struct
{
Sint16 inbuffer[24*_fsize];
Sint16 *finp, *cinp, *linp;
int flength, clength, llength;
Sint16 *buffer;
VarFilter *filter;
} RateConverterBuffer;
typedef struct
{
Sint16 carry;
Sint16 pos;
} RateAux;
/* Mono (1 channel ) */
#define Suffix(x) x##1
#include "filter_templates.h"
#undef Suffix
/* Stereo (2 channel ) */
#define Suffix(x) x##2
#include "filter_templates.h"
#undef Suffix
/*-------------------------------------------------------------------------*/
int Sound_estimateBufferSize( Sound_AudioCVT *Data, int size )
{
size *= Data->len_mult;
size += Data->len_add;
return ( size + 3 ) & -4; /* force Size in multipels of 4 Byte */
}
/*-------------------------------------------------------------------------*/
int Sound_AltConvertAudio( Sound_AudioCVT *Data,
Uint8* buffer, int length, int mode )
{
AdapterC Temp;
int i;
/* Make sure there's a converter */
if( Data == NULL ) {
SDL_SetError("No converter given");
return(-1);
}
/* Make sure there's data to convert */
if( buffer == NULL ) {
SDL_SetError("No buffer allocated for conversion");
return(-1);
}
if( length < 0 ) {
SDL_SetError("Lenght < 0");
return(-1);
}
/* Set up the conversion and go! */
Temp.buffer = buffer;
Temp.mode = mode;
Temp.filter = &Data->filter;
for( i = 0; Data->adapter[i] != NULL; i++ )
length = (*Data->adapter[i])( Temp, length);
return length;
}
int Sound_ConvertAudio( Sound_AudioCVT *Data )
{
int length;
/* !!! FIXME: Try the looping stuff under certain circumstances? --ryan. */
length = Sound_AltConvertAudio( Data, Data->buf, Data->len, 0 );
Data->len_cvt = length;
return length;
}
/*-------------------------------------------------------------------------*/
static int expand8BitTo16BitSys( AdapterC Data, int length )
{
int i;
Uint8* inp = Data.buffer - 1;
Uint16* buffer = (Uint16*)Data.buffer - 1;
for( i = length + 1; --i; )
buffer[i] = inp[i]<<8;
return 2*length;
}
static int expand8BitTo16BitWrong( AdapterC Data, int length )
{
int i;
Uint8* inp = Data.buffer - 1;
Uint16* buffer = (Uint16*)Data.buffer - 1;
for( i = length + 1; --i; )
buffer[i] = inp[i];
return 2*length;
}
/*-------------------------------------------------------------------------*/
static int expand16BitToFloat( AdapterC Data, int length )
{
int i;
Sint16* inp = (Sint16*)Data.buffer - 1;
float* buffer = (float*)Data.buffer - 1;
for( i = length>>1 + 1; --i; )
buffer[i] = inp[i]*(1./32767);
return 2*length;
}
/*-------------------------------------------------------------------------*/
static int swapBytes( AdapterC Data, int length )
{
/*
* !!! FIXME !!!
*
*
* Use the faster SDL-Macros to swap
* - Frank
*/
int i;
Uint16 a,b;
Uint16* buffer = (Uint16*) Data.buffer - 1;
for( i = length>>1 + 1; --i; )
{
a = b = buffer[i];
buffer[i] = ( a << 8 ) | ( b >> 8 );
}
return length;
}
/*-------------------------------------------------------------------------*/
static int cutFloatTo16Bit( AdapterC Data, int length )
{
int i;
float* inp = (float*) Data.buffer;
Sint16* buffer = (Sint16*) Data.buffer;
length>>=2;
for( i = 0; i < length; i++ )
{
if( inp[i] > 1. )
buffer[i] = 32767;
else if( inp[i] < -1. )
buffer[i] = -32768;
else
buffer[i] = 32767 * inp[i];
}
return 2*length;
}
/*-------------------------------------------------------------------------*/
static int cut16BitTo8Bit( AdapterC Data, int length, int off )
{
int i;
Uint8* inp = Data.buffer + off;
Uint8* buffer = Data.buffer;
length >>= 1;
for( i = 0; i < length; i++ )
buffer[i] = inp[2*i];
return length;
}
static int cut16BitSysTo8Bit( AdapterC Data, int length )
{
return cut16BitTo8Bit( Data, length, SDL_MSB_POSITION_IN_SHORT );
}
static int cut16BitWrongTo8Bit( AdapterC Data, int length )
{
return cut16BitTo8Bit( Data, length, 1-SDL_MSB_POSITION_IN_SHORT );
}
/*-------------------------------------------------------------------------*/
/* poor mans mmx :-) */
static int changeSigned( AdapterC Data, int length, Uint32 XOR )
{
int i;
Uint32* buffer = (Uint32*) Data.buffer - 1;
for( i = ( length + 7 ) >> 2; --i; )
buffer[i] ^= XOR;
return length;
}
static int changeSigned16BitSys( AdapterC Data, int length )
{
return changeSigned( Data, length, 0x80008000 );
}
static int changeSigned16BitWrong( AdapterC Data, int length )
{
return changeSigned( Data, length, 0x00800080 );
}
static int changeSigned8Bit( AdapterC Data, int length )
{
return changeSigned( Data, length, 0x80808080 );
}
/*-------------------------------------------------------------------------*/
static int convertStereoToMonoS16Bit( AdapterC Data, int length )
{
int i;
Sint16* buffer = (Sint16*) Data.buffer;
Sint16* src = (Sint16*) Data.buffer;
length >>= 2;
for( i = 0; i < length; i++, src+=2 )
buffer[i] = ((int) src[0] + src[1] ) >> 1;
return 2*length;
}
static int convertStereoToMonoU16Bit( AdapterC Data, int length )
{
int i;
Uint16* buffer = (Uint16*) Data.buffer;
Uint16* src = (Uint16*) Data.buffer;
length >>= 2;
for( i = 0; i < length; i++, src+=2 )
buffer[i] = ((int) src[0] + src[1] ) >> 1;
return 2*length;
}
static int convertStereoToMonoS8Bit( AdapterC Data, int length )
{
int i;
Sint8* buffer = (Sint8*) Data.buffer;
Sint8* src = (Sint8*) Data.buffer;
length >>= 1;
for( i = 0; i < length; i++, src+=2 )
buffer[i] = ((int) src[0] + src[1] ) >> 1;
return length;
}
static int convertStereoToMonoU8Bit( AdapterC Data, int length )
{
int i;
Uint8* buffer = (Uint8*) Data.buffer;
Uint8* src = (Uint8*) Data.buffer;
length >>= 1;
for( i = 0; i < length; i++, src+=2 )
buffer[i] = ((int) src[0] + src[1] ) >> 1;
return length;
}
/*-------------------------------------------------------------------------*/
static int convertMonoToStereo16Bit( AdapterC Data, int length )
{
int i;
Uint16* buffer;
Uint16* dst;
length >>=1;
buffer = (Uint16*)Data.buffer - 1;
dst = (Uint16*)Data.buffer + 2*length - 2;
for( i = length + 1; --i; dst-=2 )
dst[0] = dst[1] = buffer[i];
return 4*length;
}
static int convertMonoToStereo8Bit( AdapterC Data, int length )
{
int i;
Uint8* buffer = Data.buffer - 1;
Uint8* dst = Data.buffer + 2*length - 2;
for( i = length + 1; --i; dst-=2 )
dst[0] = dst[1] = buffer[i];
return 2*length;
}
/*-------------------------------------------------------------------------*/
static int minus5dB( AdapterC Data, int length )
{
int i;
Sint16* buffer = (Sint16*) Data.buffer;
for(i = length>>1 + 1; --i; )
buffer[i] = (38084 * (int)buffer[i]) >> 16;
return length;
}
/*-------------------------------------------------------------------------*/
const Fraction Half = {1, 2};
const Fraction Double = {2, 1};
const Fraction One = {1, 1};
static void initStraigthBuffer( RateConverterBuffer *rcb,
int length, Fraction r )
{
int i, size, minsize;
size = 8 * _fsize;
minsize = min( size, length );
for( i = 0; i < minsize; i++ )
{
rcb->inbuffer[i] = rcb->buffer[length-size+i];
rcb->inbuffer[i+size] = 0;
rcb->inbuffer[i+2*size] = rcb->buffer[i];
}
for( ; i < size; i++ )
{
rcb->inbuffer[i] = 0;
rcb->inbuffer[i+size] = 0;
rcb->inbuffer[i+2*size] = 0;
}
length = max( length, size );
rcb->flength = rcb->llength = size;
rcb->clength = length - size;
if( r.numerator < r.denominator )
{
rcb->finp = rcb->inbuffer + 5*size/2;
rcb->cinp = rcb->buffer + length - size/2;
rcb->linp = rcb->inbuffer + 3*size/2;
rcb->buffer += ( 1 + r.denominator * ( length + size )
/ r.numerator ) & -2;
}
else
{
rcb->finp = rcb->inbuffer + size/2;
rcb->cinp = rcb->buffer + size/2;
rcb->linp = rcb->inbuffer + 3*size/2;
}
}
static void initLoopBuffer( RateConverterBuffer *rcb,
int length, Fraction r )
{
/* !!!FIXME: modulo length, take scale into account,
check against the Straight part -frank */
int i, size;
size = 8 * _fsize;
for( i = 0; i < size; i++ )
{
rcb->inbuffer[i] = rcb->buffer[length-size+i];
rcb->inbuffer[i+size] = rcb->buffer[i];
}
rcb->finp = rcb->linp = rcb->inbuffer + size;
if( size < 0 )
rcb->buffer += r.numerator * ( length + 2 * size )
/ r.denominator;
}
static void initRateConverterBuffer( RateConverterBuffer *rcb,
AdapterC* Data, int length, Fraction ratio )
{
length >>= 1;
rcb->buffer = (Sint16*)( Data->buffer );
rcb->filter = Data->filter;
if( Data->mode & SDL_SOUND_Loop )
initLoopBuffer( rcb, length, ratio );
else
initStraigthBuffer( rcb, length, ratio );
fprintf( stderr, " finp: %8x length: %8x\n", rcb->finp, rcb->flength );
fprintf( stderr, " cinp: %8x length: %8x\n", rcb->cinp, rcb->clength );
fprintf( stderr, " linp: %8x length: %8x\n", rcb->linp, rcb->llength );
}
static void nextRateConverterBuffer( RateConverterBuffer *rcb )
{
rcb->buffer++;
rcb->finp++;
rcb->cinp++;
rcb->linp++;
}
typedef Sint16* (*RateConverter)( Sint16*, Sint16*, int,
VarFilter*, RateAux* );
static Sint16* doRateConversion( RateConverterBuffer* rcb, RateConverter rc )
{
RateAux aux = {0,0};
Sint16 *outp = rcb->buffer;
VarFilter* filter = rcb->filter;
outp = (*rc)( outp, rcb->finp, rcb->flength, filter, &aux );
fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
outp = (*rc)( outp, rcb->cinp, rcb->clength, filter, &aux );
fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
outp = (*rc)( outp, rcb->linp, rcb->llength, filter, &aux );
fprintf( stderr, " outp: %8x aux.carry: %8x\n", outp, aux.carry );
return outp;
}
/*-------------------------------------------------------------------------*/
static void clearSint16Buffer( Sint8* buffer, Sint16*r )
{
while( r >= (Sint16*)buffer ) *r-- = 0;
}
/*-------------------------------------------------------------------------*/
static int doubleRateMono( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Half );
r = 1 + doRateConversion( &rcb, doubleRate1 );
clearSint16Buffer( Data.buffer, r );
return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 2 );
}
static int doubleRateStereo( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
fprintf( stderr, "\n Buffer: %8x length: %8x\n", Data.buffer, length );
initRateConverterBuffer( &rcb, &Data, length, Half );
doRateConversion( &rcb, doubleRate2 );
nextRateConverterBuffer( &rcb );
r = 2 + doRateConversion( &rcb, doubleRate2 );
clearSint16Buffer( Data.buffer, r );
return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 3 );
}
/*-------------------------------------------------------------------------*/
static int halfRateMono( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Double );
r = doRateConversion( &rcb, halfRate1 );
return 2 * ( r - (Sint16*)Data.buffer );
}
static int halfRateStereo( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Double );
doRateConversion( &rcb, halfRate2 );
nextRateConverterBuffer( &rcb );
r = doRateConversion( &rcb, halfRate2 );
return 2 * ( r - (Sint16*)Data.buffer );
}
/*-------------------------------------------------------------------------*/
static int increaseRateMono( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
r = doRateConversion( &rcb, increaseRate1 );
clearSint16Buffer( Data.buffer, r );
return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 1 );
}
static int increaseRateStereo( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
fprintf( stderr, "\n Buffer: %8x length: %8x\n", Data.buffer, length );
initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
doRateConversion( &rcb, increaseRate2 );
nextRateConverterBuffer( &rcb );
r = doRateConversion( &rcb, increaseRate2 );
clearSint16Buffer( Data.buffer, r );
return 2 * ( rcb.buffer - (Sint16*)Data.buffer + 1 );
}
/*-------------------------------------------------------------------------*/
static int decreaseRateMono( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
r = doRateConversion( &rcb, decreaseRate1 );
return 2 * ( r - (Sint16*)Data.buffer );
}
static int decreaseRateStereo( AdapterC Data, int length )
{
Sint16* r;
RateConverterBuffer rcb;
initRateConverterBuffer( &rcb, &Data, length, Data.filter->ratio );
doRateConversion( &rcb, decreaseRate2 );
nextRateConverterBuffer( &rcb );
r = doRateConversion( &rcb, decreaseRate2 );
return 2 * ( r - (Sint16*)Data.buffer );
}
/*-------------------------------------------------------------------------*/
/* gives a maximal error of 3% and typical less than 0.2% */
static Fraction findFraction( float Value )
{
const Sint8 frac[95]={
2, -1, /* /1 */
1, 3, -1, /* /2 */
2, 4, 5, -1, /* /3 */
3, 5, 7, -1, /* /4 */
3, 4, 6, 7, 8, 9, -1, /* /5 */
5, 7, 11, -1, /* /6 */
4, 5, 6, 8, 9, 10, 11, 12, 13, -1, /* /7 */
5, 7, 9, 11, 13, 15, -1, /* /8 */
5, 7, 8, 10, 11, 13, 14, 16, -1, /* /9 */
7, 9, 11, 13, -1, /* /10 */
6, 7, 8, 9, 10, 12, 13, 14, 15, 16, -1, /* /11 */
7, 11, 13, -1, /* /12 */
7, 8, 9, 10, 11, 12, 14, 15, 16, -1, /* /13 */
9, 11, 13, 15, -1, /* /14 */
8, 11, 13, 14, 16, -1, /* /15 */
9, 11, 13, 15 }; /* /16 */
Fraction Result = {0,0};
int i,num,den=1;
float RelErr, BestErr = 0;
if( Value < 31/64. || Value > 64/31. ) return Result;
for( i = 0; i < SDL_TABLESIZE(frac); i++ )
{
num = frac[i];
if( num < 0 ) den++;
RelErr = Value * num / den;
RelErr = min( RelErr, 1/RelErr );
if( RelErr > BestErr )
{
BestErr = RelErr;
Result.denominator = den;
Result.numerator = num;
}
}
return Result;
}
/*-------------------------------------------------------------------------*/
static float sinc( float x )
{
if( x > -1e-24 && x < 1e-24 ) return 1.;
else return sin(x)/x;
}
static float calculateVarFilter( Sint16* dst,
float Ratio, float phase, float scale )
{
const Uint16 KaiserWindow7[]= {
22930, 16292, 14648, 14288, 14470, 14945, 15608, 16404,
17304, 18289, 19347, 20467, 21644, 22872, 24145, 25460,
26812, 28198, 29612, 31052, 32513, 33991, 35482, 36983,
38487, 39993, 41494, 42986, 44466, 45928, 47368, 48782,
50165, 51513, 52821, 54086, 55302, 56466, 57575, 58624,
59610, 60529, 61379, 62156, 62858, 63483, 64027, 64490,
64870, 65165, 65375, 65498, 65535, 65484, 65347, 65124,
64815, 64422, 63946, 63389, 62753, 62039, 61251, 60391 };
int i;
float w;
const float fg = -.018 + .5 * Ratio;
const float omega = 2 * M_PI * fg;
fprintf( stderr, " phase: %6g \n", phase );
phase += 63;
for( i = 0; i < 64; i++)
{
w = scale * ( KaiserWindow7[i] * ( i + 1 ));
dst[i] = w * sinc( omega * (i-phase) );
dst[127-i] = w * sinc( omega * (127-i-phase) );
}
fprintf( stderr, " center: %6d %6d \n", dst[63], dst[64] );
return fg;
}
static Fraction setupVarFilter( Sound_AudioCVT *Data, float Ratio )
{
int pos,n,d, incr, phase = 0;
float Scale, rd, fg;
Fraction IRatio;
VarFilter* filter = &Data->filter;
IRatio = findFraction( Ratio );
// Scale = Ratio < 1. ? 0.0364733 : 0.0211952;
Scale = 0.0084778;
Ratio = min( Ratio, 0.97 );
filter->ratio = IRatio;
n = IRatio.numerator;
d = IRatio.denominator;
rd = 1. / d;
fprintf( stderr, "Filter:\n" );
for( pos = 0; pos < d; pos++ )
{
fg = calculateVarFilter( filter->c[pos], Ratio, phase*rd, Scale );
phase += n;
filter->incr[pos] = phase / d;
phase %= d;
}
fprintf( stderr, " fg: %6g\n\n", fg );
/* !!!FIXME: get rid of the inversion -Frank*/
IRatio.numerator = d;
IRatio.denominator = n;
return IRatio;
}
/*-------------------------------------------------------------------------*/
static void initAudioCVT( Sound_AudioCVT *Data )
{
Data->len_ratio = 1.;
Data->len_mult = 1;
Data->add = 0;
Data->len_add = 0;
Data->filter_index = 0;
}
static void adjustSize( Sound_AudioCVT *Data, int add, Fraction f )
{
double ratio = f.numerator / (double) f.denominator;
Data->len_ratio *= ratio;
Data->len_mult = max( Data->len_mult, ceil(Data->len_ratio) );
Data->add = ratio * (Data->add + add);
Data->len_add = max( Data->len_add, ceil(Data->add) );
}
static Adapter* addAdapter( Sound_AudioCVT *Data, Adapter a )
{
Data->adapter[Data->filter_index] = a;
return &Data->adapter[Data->filter_index++];
}
static void addHAdapter( Sound_AudioCVT *Data, Adapter a )
{
adjustSize( Data, 0, Half );
addAdapter( Data, a );
}
static void addDAdapter( Sound_AudioCVT *Data, Adapter a )
{
adjustSize( Data, 0, Double );
addAdapter( Data, a );
}
/*-------------------------------------------------------------------------*/
const Adapter doubleRate[2] = { doubleRateMono, doubleRateStereo };
const Adapter halfRate[2] = { halfRateMono, halfRateStereo };
const Adapter increaseRate[2] = { increaseRateMono, increaseRateStereo };
const Adapter decreaseRate[2] = { decreaseRateMono, decreaseRateStereo };
static int createRateConverter( Sound_AudioCVT *Data,
int SrcRate, int DestRate, int channel )
{
const int c = channel - 1;
const int size = 16 * channel * _fsize;
Adapter* AdapterPos;
float Ratio = DestRate;
Fraction f;
if( SrcRate < 1 || SrcRate > 1<<18 ||
DestRate < 1 || DestRate > 1<<18 ) return -1;
Ratio /= SrcRate;
AdapterPos = addAdapter( Data, minus5dB );
while( Ratio > 64./31.)
{
Ratio /= 2.;
addAdapter( Data, doubleRate[c] );
adjustSize( Data, size, Double );
}
while( Ratio < 31./64. )
{
Ratio *= 2;
addAdapter( Data, halfRate[c] );
adjustSize( Data, size, Half );
}
if( Ratio > 1. )
{
*AdapterPos = increaseRate[c];
f = setupVarFilter( Data, Ratio );
adjustSize( Data, size, f );
}
else
{
f = setupVarFilter( Data, Ratio );
addAdapter( Data, decreaseRate[c]);
adjustSize( Data, size, f );
}
return 0;
}
/*-------------------------------------------------------------------------*/
static void createFormatConverter16Bit(Sound_AudioCVT *Data,
SDL_AudioSpec src, SDL_AudioSpec dst )
{
if( src.channels == 2 && dst.channels == 1 )
{
if( !IS_SYSENDIAN(src) )
addAdapter( Data, swapBytes );
if( IS_SIGNED(src) )
addHAdapter( Data, convertStereoToMonoS16Bit );
else
addHAdapter( Data, convertStereoToMonoU16Bit );
if( !IS_SYSENDIAN(dst) )
addAdapter( Data, swapBytes );
}
else if( IS_SYSENDIAN(src) != IS_SYSENDIAN(dst) )
addAdapter( Data, swapBytes );
if( IS_SIGNED(src) != IS_SIGNED(dst) )
{
if( IS_SYSENDIAN(dst) )
addAdapter( Data, changeSigned16BitSys );
else
addAdapter( Data, changeSigned16BitWrong );
}
if( src.channels == 1 && dst.channels == 2 )
addDAdapter( Data, convertMonoToStereo16Bit );
}
/*-------------------------------------------------------------------------*/
static void createFormatConverter8Bit(Sound_AudioCVT *Data,
SDL_AudioSpec src, SDL_AudioSpec dst )
{
if( IS_16BIT(src) )
{
if( IS_SYSENDIAN(src) )
addHAdapter( Data, cut16BitSysTo8Bit );
else
addHAdapter( Data, cut16BitWrongTo8Bit );
}
if( src.channels == 2 && dst.channels == 1 )
{
if( IS_SIGNED(src) )
addHAdapter( Data, convertStereoToMonoS8Bit );
else
addHAdapter( Data, convertStereoToMonoU8Bit );
}
if( IS_SIGNED(src) != IS_SIGNED(dst) )
addDAdapter( Data, changeSigned8Bit );
if( src.channels == 1 && dst.channels == 2 )
addDAdapter( Data, convertMonoToStereo8Bit );
if( !IS_8BIT(dst) )
{
if( IS_SYSENDIAN(dst) )
addDAdapter( Data, expand8BitTo16BitSys );
else
addDAdapter( Data, expand8BitTo16BitWrong );
}
}
/*-------------------------------------------------------------------------*/
static void createFormatConverter(Sound_AudioCVT *Data,
SDL_AudioSpec src, SDL_AudioSpec dst )
{
if( IS_FLOAT(src) )
addHAdapter( Data, cutFloatTo16Bit );
if( IS_8BIT(src) || IS_8BIT(dst) )
createFormatConverter8Bit( Data, src, dst);
else
createFormatConverter16Bit( Data, src, dst);
if( IS_FLOAT(dst) )
addDAdapter( Data, expand16BitToFloat );
}
/*-------------------------------------------------------------------------*/
int Sound_AltBuildAudioCVT( Sound_AudioCVT *Data,
SDL_AudioSpec src, SDL_AudioSpec dst )
{
SDL_AudioSpec im;
if( Data == NULL ) return -1;
initAudioCVT( Data );
Data->filter.ratio.denominator = 0;
Data->filter.mask = dst.size - 1;
/* Check channels */
if( src.channels < 1 || src.channels > 2 ||
dst.channels < 1 || dst.channels > 2 ) goto error_exit;
if( src.freq != dst.freq )
{
/* Convert to intermidiate format: signed 16Bit System-Endian */
im.format = AUDIO_S16SYS;
im.channels = min( src.channels, dst.channels );
createFormatConverter( Data, src, im );
/* Do rate conversion */
if( createRateConverter( Data, src.freq, dst.freq, im.channels ) )
goto error_exit;
src = im;
}
/* Convert to final format */
createFormatConverter( Data, src, dst );
/* Finalize adapter list */
addAdapter( Data, NULL );
/* !!! FIXME: Is it okay to assign NULL to a function pointer?
Borland says no. -frank */
return 0;
error_exit:
/* !!! FIXME: Is it okay to assign NULL to a function pointer?
Borland says no. -frank */
Data->adapter[0] = NULL;
return -1;
}
/*-------------------------------------------------------------------------*/
static char *fmt_to_str(Uint16 fmt)
{
switch (fmt)
{
case AUDIO_U8: return " U8";
case AUDIO_S8: return " S8";
case AUDIO_U16MSB: return "U16MSB";
case AUDIO_S16MSB: return "S16MSB";
case AUDIO_U16LSB: return "U16LSB";
case AUDIO_S16LSB: return "S16LSB";
}
return "??????";
}
#define AdapterDesc(x) { x, #x }
static void show_AudioCVT( Sound_AudioCVT *Data )
{
int i,j;
const struct{ int (*adapter) ( AdapterC, int); Sint8 *name; }
AdapterDescription[] = {
AdapterDesc(expand8BitTo16BitSys),
AdapterDesc(expand8BitTo16BitWrong),
AdapterDesc(expand16BitToFloat),
AdapterDesc(swapBytes),
AdapterDesc(cut16BitSysTo8Bit),
AdapterDesc(cut16BitWrongTo8Bit),
AdapterDesc(cutFloatTo16Bit),
AdapterDesc(changeSigned16BitSys),
AdapterDesc(changeSigned16BitWrong),
AdapterDesc(changeSigned8Bit),
AdapterDesc(convertStereoToMonoS16Bit),
AdapterDesc(convertStereoToMonoU16Bit),
AdapterDesc(convertStereoToMonoS8Bit),
AdapterDesc(convertStereoToMonoU8Bit),
AdapterDesc(convertMonoToStereo16Bit),
AdapterDesc(convertMonoToStereo8Bit),
AdapterDesc(minus5dB),
AdapterDesc(doubleRateMono),
AdapterDesc(doubleRateStereo),
AdapterDesc(halfRateMono),
AdapterDesc(halfRateStereo),
AdapterDesc(increaseRateMono),
AdapterDesc(increaseRateStereo),
AdapterDesc(decreaseRateMono),
AdapterDesc(decreaseRateStereo),
{ NULL, "----------NULL-----------\n" }
};
fprintf( stderr, "Sound_AudioCVT:\n" );
fprintf( stderr, " needed: %8d\n", Data->needed );
fprintf( stderr, " add: %8g\n", Data->add );
fprintf( stderr, " len_add: %8d\n", Data->len_add );
fprintf( stderr, " len_ratio: %8g\n", Data->len_ratio );
fprintf( stderr, " len_mult: %8d\n", Data->len_mult );
fprintf( stderr, " filter->mask: %#7x\n", Data->filter.mask );
fprintf( stderr, "\n" );
fprintf( stderr, "Adapter List: \n" );
for( i = 0; i < 32; i++ )
{
for( j = 0; j < SDL_TABLESIZE(AdapterDescription); j++ )
{
if( Data->adapter[i] == AdapterDescription[j].adapter )
{
fprintf( stderr, " %s \n", AdapterDescription[j].name );
if( Data->adapter[i] == NULL ) goto sucess_exit;
goto cont;
}
}
fprintf( stderr, " Error: unknown adapter\n" );
cont:
}
fprintf( stderr, " Error: NULL adapter missing\n" );
sucess_exit:
if( Data->filter.ratio.denominator )
{
fprintf( stderr, "Variable Rate Converter:\n"
" numerator: %3d\n"