/
SDL_audiocvt.c
975 lines (827 loc) · 31.5 KB
1
2
/*
Simple DirectMedia Layer
3
Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../SDL_internal.h"
/* Functions for audio drivers to perform runtime conversion of audio format */
#include "SDL_audio.h"
#include "SDL_audio_c.h"
28
#include "SDL_loadso.h"
29
#include "SDL_assert.h"
30
#include "../SDL_dataqueue.h"
31
32
33
/* Effectively mix right and left channels into a single channel */
static void SDLCALL
34
SDL_ConvertStereoToMono(SDL_AudioCVT * cvt, SDL_AudioFormat format)
35
{
36
37
float *dst = (float *) cvt->buf;
const float *src = dst;
38
39
int i;
40
41
42
43
LOG_DEBUG_CONVERT("stereo", "mono");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / 8; i; --i, src += 2) {
44
*(dst++) = (src[0] + src[1]) * 0.5f;
45
46
47
48
49
50
51
52
53
}
cvt->len_cvt /= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
54
/* Convert from 5.1 to stereo. Average left and right, discard subwoofer. */
55
static void SDLCALL
56
SDL_Convert51ToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
57
{
58
59
float *dst = (float *) cvt->buf;
const float *src = dst;
60
61
int i;
62
LOG_DEBUG_CONVERT("5.1", "stereo");
63
SDL_assert(format == AUDIO_F32SYS);
64
65
/* this assumes FL+FR+FC+subwoof+BL+BR layout. */
66
for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 2) {
67
68
69
const double front_center = (double) src[2];
dst[0] = (float) ((src[0] + front_center + src[4]) / 3.0); /* left */
dst[1] = (float) ((src[1] + front_center + src[5]) / 3.0); /* right */
70
71
72
73
74
75
76
77
78
}
cvt->len_cvt /= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
79
/* Convert from 5.1 to quad */
80
static void SDLCALL
81
SDL_Convert51ToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
82
{
83
84
float *dst = (float *) cvt->buf;
const float *src = dst;
85
86
int i;
87
LOG_DEBUG_CONVERT("5.1", "quad");
88
SDL_assert(format == AUDIO_F32SYS);
89
90
/* assumes quad is FL+FR+BL+BR layout and 5.1 is FL+FR+FC+subwoof+BL+BR */
91
for (i = cvt->len_cvt / (sizeof (float) * 6); i; --i, src += 6, dst += 4) {
92
93
94
95
96
97
/* FIXME: this is a good candidate for SIMD. */
const double front_center = (double) src[2];
dst[0] = (float) ((src[0] + front_center) * 0.5); /* FL */
dst[1] = (float) ((src[1] + front_center) * 0.5); /* FR */
dst[2] = (float) ((src[4] + front_center) * 0.5); /* BL */
dst[3] = (float) ((src[5] + front_center) * 0.5); /* BR */
98
99
100
101
102
103
104
105
106
}
cvt->len_cvt /= 6;
cvt->len_cvt *= 4;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
107
108
109
/* Duplicate a mono channel to both stereo channels */
static void SDLCALL
110
SDL_ConvertMonoToStereo(SDL_AudioCVT * cvt, SDL_AudioFormat format)
111
{
112
113
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
114
115
int i;
116
117
LOG_DEBUG_CONVERT("mono", "stereo");
SDL_assert(format == AUDIO_F32SYS);
118
119
120
121
122
for (i = cvt->len_cvt / sizeof (float); i; --i) {
src--;
dst -= 2;
dst[0] = dst[1] = *src;
123
124
125
126
127
128
129
130
131
132
133
}
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-5.1 stream */
static void SDLCALL
134
SDL_ConvertStereoTo51(SDL_AudioCVT * cvt, SDL_AudioFormat format)
135
136
{
int i;
137
138
139
140
141
142
143
144
145
146
147
148
float lf, rf, ce;
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 3);
LOG_DEBUG_CONVERT("stereo", "5.1");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 6;
src -= 2;
lf = src[0];
rf = src[1];
149
150
151
152
153
154
155
ce = (lf + rf) * 0.5f;
dst[0] = lf + (lf - ce); /* FL */
dst[1] = rf + (rf - ce); /* FR */
dst[2] = ce; /* FC */
dst[3] = ce; /* !!! FIXME: wrong! This is the subwoofer. */
dst[4] = lf; /* BL */
dst[5] = rf; /* BR */
156
}
157
158
159
160
161
162
163
164
165
166
cvt->len_cvt *= 3;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
/* Duplicate a stereo channel to a pseudo-4.0 stream */
static void SDLCALL
167
SDL_ConvertStereoToQuad(SDL_AudioCVT * cvt, SDL_AudioFormat format)
168
{
169
170
const float *src = (const float *) (cvt->buf + cvt->len_cvt);
float *dst = (float *) (cvt->buf + cvt->len_cvt * 2);
171
float lf, rf;
172
173
int i;
174
175
176
177
178
179
180
181
LOG_DEBUG_CONVERT("stereo", "quad");
SDL_assert(format == AUDIO_F32SYS);
for (i = cvt->len_cvt / 8; i; --i) {
dst -= 4;
src -= 2;
lf = src[0];
rf = src[1];
182
183
184
185
dst[0] = lf; /* FL */
dst[1] = rf; /* FR */
dst[2] = lf; /* BL */
dst[3] = rf; /* BR */
186
}
187
188
189
190
191
192
193
cvt->len_cvt *= 2;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
194
195
196
197
198
static int
SDL_ResampleAudioSimple(const int chans, const double rate_incr,
float *last_sample, const float *inbuf,
const int inbuflen, float *outbuf, const int outbuflen)
{
199
const int framelen = chans * (int)sizeof (float);
200
const int total = (inbuflen / framelen);
201
202
const int finalpos = (total * chans) - chans;
const int dest_samples = (int)(((double)total) * rate_incr);
203
204
const double src_incr = 1.0 / rate_incr;
float *dst = outbuf;
205
206
float *target = (dst + (dest_samples * chans));
double idx = 0.0;
207
208
int i;
209
SDL_assert((dest_samples * framelen) <= outbuflen);
210
211
SDL_assert((inbuflen % framelen) == 0);
212
while (dst < target) {
213
const int pos = ((int)idx) * chans;
214
215
const float *src = &inbuf[pos];
SDL_assert(pos <= finalpos);
216
217
218
219
220
221
222
223
for (i = 0; i < chans; i++) {
const float val = *(src++);
*(dst++) = (val + last_sample[i]) * 0.5f;
last_sample[i] = val;
}
idx += src_incr;
}
224
return (int) ((dst - outbuf) * (int)sizeof(float));
225
226
}
227
228
229
230
231
232
233
234
235
int
SDL_ConvertAudio(SDL_AudioCVT * cvt)
{
/* !!! FIXME: (cvt) should be const; stack-copy it here. */
/* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
/* Make sure there's data to convert */
if (cvt->buf == NULL) {
236
return SDL_SetError("No buffer allocated for conversion");
237
}
238
239
240
241
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if (cvt->filters[0] == NULL) {
242
return 0;
243
244
245
246
247
}
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0] (cvt, cvt->src_format);
248
return 0;
249
250
}
251
252
static void SDLCALL
SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
253
{
254
255
256
#if DEBUG_CONVERT
printf("Converting byte order\n");
#endif
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
switch (SDL_AUDIO_BITSIZE(format)) {
#define CASESWAP(b) \
case b: { \
Uint##b *ptr = (Uint##b *) cvt->buf; \
int i; \
for (i = cvt->len_cvt / sizeof (*ptr); i; --i, ++ptr) { \
*ptr = SDL_Swap##b(*ptr); \
} \
break; \
}
CASESWAP(16);
CASESWAP(32);
CASESWAP(64);
#undef CASESWAP
default: SDL_assert(!"unhandled byteswap datatype!"); break;
}
277
278
279
280
281
282
283
284
285
286
if (cvt->filters[++cvt->filter_index]) {
/* flip endian flag for data. */
if (format & SDL_AUDIO_MASK_ENDIAN) {
format &= ~SDL_AUDIO_MASK_ENDIAN;
} else {
format |= SDL_AUDIO_MASK_ENDIAN;
}
cvt->filters[cvt->filter_index](cvt, format);
}
287
288
289
290
}
static int
291
SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
292
{
293
int retval = 0; /* 0 == no conversion necessary. */
294
295
296
297
298
if ((SDL_AUDIO_ISBIGENDIAN(src_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
retval = 1; /* added a converter. */
}
299
300
if (!SDL_AUDIO_ISFLOAT(src_fmt)) {
301
302
const Uint16 src_bitsize = SDL_AUDIO_BITSIZE(src_fmt);
const Uint16 dst_bitsize = 32;
303
SDL_AudioFilter filter = NULL;
304
305
306
307
308
switch (src_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
case AUDIO_S8: filter = SDL_Convert_S8_to_F32; break;
case AUDIO_U8: filter = SDL_Convert_U8_to_F32; break;
case AUDIO_S16: filter = SDL_Convert_S16_to_F32; break;
309
case AUDIO_U16: filter = SDL_Convert_U16_to_F32; break;
310
311
case AUDIO_S32: filter = SDL_Convert_S32_to_F32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
312
313
}
314
315
316
317
if (!filter) {
return SDL_SetError("No conversion available for these formats");
}
318
319
320
321
322
323
324
325
cvt->filters[cvt->filter_index++] = filter;
if (src_bitsize < dst_bitsize) {
const int mult = (dst_bitsize / src_bitsize);
cvt->len_mult *= mult;
cvt->len_ratio *= mult;
} else if (src_bitsize > dst_bitsize) {
cvt->len_ratio /= (src_bitsize / dst_bitsize);
}
326
327
retval = 1; /* added a converter. */
328
329
}
330
return retval;
331
332
}
333
334
static int
SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
335
{
336
337
338
int retval = 0; /* 0 == no conversion necessary. */
if (!SDL_AUDIO_ISFLOAT(dst_fmt)) {
339
340
const Uint16 dst_bitsize = SDL_AUDIO_BITSIZE(dst_fmt);
const Uint16 src_bitsize = 32;
341
342
343
344
345
SDL_AudioFilter filter = NULL;
switch (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN) {
case AUDIO_S8: filter = SDL_Convert_F32_to_S8; break;
case AUDIO_U8: filter = SDL_Convert_F32_to_U8; break;
case AUDIO_S16: filter = SDL_Convert_F32_to_S16; break;
346
case AUDIO_U16: filter = SDL_Convert_F32_to_U16; break;
347
348
349
350
351
352
353
case AUDIO_S32: filter = SDL_Convert_F32_to_S32; break;
default: SDL_assert(!"Unexpected audio format!"); break;
}
if (!filter) {
return SDL_SetError("No conversion available for these formats");
}
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
cvt->filters[cvt->filter_index++] = filter;
if (src_bitsize < dst_bitsize) {
const int mult = (dst_bitsize / src_bitsize);
cvt->len_mult *= mult;
cvt->len_ratio *= mult;
} else if (src_bitsize > dst_bitsize) {
cvt->len_ratio /= (src_bitsize / dst_bitsize);
}
retval = 1; /* added a converter. */
}
if ((SDL_AUDIO_ISBIGENDIAN(dst_fmt) != 0) == (SDL_BYTEORDER == SDL_LIL_ENDIAN)) {
cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
retval = 1; /* added a converter. */
}
return retval;
372
373
}
374
375
376
377
378
379
380
static void
SDL_ResampleCVT(SDL_AudioCVT *cvt, const int chans, const SDL_AudioFormat format)
{
const float *src = (const float *) cvt->buf;
const int srclen = cvt->len_cvt;
float *dst = (float *) (cvt->buf + srclen);
const int dstlen = (cvt->len * cvt->len_mult) - srclen;
381
float state[8];
382
383
384
SDL_assert(format == AUDIO_F32SYS);
385
SDL_memcpy(state, src, chans*sizeof(*src));
386
387
cvt->len_cvt = SDL_ResampleAudioSimple(chans, cvt->rate_incr, state, src, srclen, dst, dstlen);
388
389
390
391
392
393
394
395
396
397
398
SDL_memcpy(cvt->buf, dst, cvt->len_cvt);
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* !!! FIXME: We only have this macro salsa because SDL_AudioCVT doesn't
!!! FIXME: store channel info, so we have to have function entry
!!! FIXME: points for each supported channel count and multiple
!!! FIXME: vs arbitrary. When we rev the ABI, clean this up. */
399
400
#define RESAMPLER_FUNCS(chans) \
static void SDLCALL \
401
402
SDL_ResampleCVT_c##chans(SDL_AudioCVT *cvt, SDL_AudioFormat format) { \
SDL_ResampleCVT(cvt, chans, format); \
403
404
405
406
407
408
409
410
}
RESAMPLER_FUNCS(1)
RESAMPLER_FUNCS(2)
RESAMPLER_FUNCS(4)
RESAMPLER_FUNCS(6)
RESAMPLER_FUNCS(8)
#undef RESAMPLER_FUNCS
411
static SDL_AudioFilter
412
ChooseCVTResampler(const int dst_channels)
413
{
414
415
416
417
418
419
420
switch (dst_channels) {
case 1: return SDL_ResampleCVT_c1;
case 2: return SDL_ResampleCVT_c2;
case 4: return SDL_ResampleCVT_c4;
case 6: return SDL_ResampleCVT_c6;
case 8: return SDL_ResampleCVT_c8;
default: break;
421
422
}
423
return NULL;
424
425
426
427
428
429
430
431
432
433
434
435
}
static int
SDL_BuildAudioResampleCVT(SDL_AudioCVT * cvt, const int dst_channels,
const int src_rate, const int dst_rate)
{
SDL_AudioFilter filter;
if (src_rate == dst_rate) {
return 0; /* no conversion necessary. */
}
436
filter = ChooseCVTResampler(dst_channels);
437
438
439
if (filter == NULL) {
return SDL_SetError("No conversion available for these rates");
}
440
441
442
443
444
445
446
447
448
/* Update (cvt) with filter details... */
cvt->filters[cvt->filter_index++] = filter;
if (src_rate < dst_rate) {
const double mult = ((double) dst_rate) / ((double) src_rate);
cvt->len_mult *= (int) SDL_ceil(mult);
cvt->len_ratio *= mult;
} else {
cvt->len_ratio /= ((double) src_rate) / ((double) dst_rate);
449
450
}
451
452
453
454
/* the buffer is big enough to hold the destination now, but
we need it large enough to hold a separate scratch buffer. */
cvt->len_mult *= 2;
455
return 1; /* added a converter. */
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
}
/* Creates a set of audio filters to convert from one format to another.
Returns -1 if the format conversion is not supported, 0 if there's
no conversion needed, or 1 if the audio filter is set up.
*/
int
SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate,
SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
{
/* Sanity check target pointer */
if (cvt == NULL) {
return SDL_InvalidParamError("cvt");
}
474
475
476
/* Make sure we zero out the audio conversion before error checking */
SDL_zerop(cvt);
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
/* there are no unsigned types over 16 bits, so catch this up front. */
if ((SDL_AUDIO_BITSIZE(src_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(src_fmt))) {
return SDL_SetError("Invalid source format");
}
if ((SDL_AUDIO_BITSIZE(dst_fmt) > 16) && (!SDL_AUDIO_ISSIGNED(dst_fmt))) {
return SDL_SetError("Invalid destination format");
}
/* prevent possible divisions by zero, etc. */
if ((src_channels == 0) || (dst_channels == 0)) {
return SDL_SetError("Source or destination channels is zero");
}
if ((src_rate == 0) || (dst_rate == 0)) {
return SDL_SetError("Source or destination rate is zero");
}
492
#if DEBUG_CONVERT
493
494
495
496
497
498
499
500
501
502
503
504
505
506
printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
#endif
/* Start off with no conversion necessary */
cvt->src_format = src_fmt;
cvt->dst_format = dst_fmt;
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
507
508
509
510
511
512
513
514
515
/* Type conversion goes like this now:
- byteswap to CPU native format first if necessary.
- convert to native Float32 if necessary.
- resample and change channel count if necessary.
- convert back to native format.
- byteswap back to foreign format if necessary.
The expectation is we can process data faster in float32
(possibly with SIMD), and making several passes over the same
516
buffer is likely to be CPU cache-friendly, avoiding the
517
518
519
520
biggest performance hit in modern times. Previously we had
(script-generated) custom converters for every data type and
it was a bloat on SDL compile times and final library size. */
521
522
523
524
525
526
527
528
529
530
531
532
/* see if we can skip float conversion entirely. */
if (src_rate == dst_rate && src_channels == dst_channels) {
if (src_fmt == dst_fmt) {
return 0;
}
/* just a byteswap needed? */
if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
cvt->filters[cvt->filter_index++] = SDL_Convert_Byteswap;
cvt->needed = 1;
return 1;
}
533
534
}
535
/* Convert data types, if necessary. Updates (cvt). */
536
if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
537
538
539
540
541
542
return -1; /* shouldn't happen, but just in case... */
}
/* Channel conversion */
if (src_channels != dst_channels) {
if ((src_channels == 1) && (dst_channels > 1)) {
543
cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
544
545
546
547
548
cvt->len_mult *= 2;
src_channels = 2;
cvt->len_ratio *= 2;
}
if ((src_channels == 2) && (dst_channels == 6)) {
549
cvt->filters[cvt->filter_index++] = SDL_ConvertStereoTo51;
550
551
552
553
554
src_channels = 6;
cvt->len_mult *= 3;
cvt->len_ratio *= 3;
}
if ((src_channels == 2) && (dst_channels == 4)) {
555
cvt->filters[cvt->filter_index++] = SDL_ConvertStereoToQuad;
556
557
558
559
560
src_channels = 4;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
}
while ((src_channels * 2) <= dst_channels) {
561
cvt->filters[cvt->filter_index++] = SDL_ConvertMonoToStereo;
562
563
564
565
566
cvt->len_mult *= 2;
src_channels *= 2;
cvt->len_ratio *= 2;
}
if ((src_channels == 6) && (dst_channels <= 2)) {
567
cvt->filters[cvt->filter_index++] = SDL_Convert51ToStereo;
568
569
570
571
src_channels = 2;
cvt->len_ratio /= 3;
}
if ((src_channels == 6) && (dst_channels == 4)) {
572
cvt->filters[cvt->filter_index++] = SDL_Convert51ToQuad;
573
574
575
576
577
578
579
580
581
src_channels = 4;
cvt->len_ratio /= 2;
}
/* This assumes that 4 channel audio is in the format:
Left {front/back} + Right {front/back}
so converting to L/R stereo works properly.
*/
while (((src_channels % 2) == 0) &&
((src_channels / 2) >= dst_channels)) {
582
cvt->filters[cvt->filter_index++] = SDL_ConvertStereoToMono;
583
584
585
586
587
588
589
590
591
src_channels /= 2;
cvt->len_ratio /= 2;
}
if (src_channels != dst_channels) {
/* Uh oh.. */ ;
}
}
/* Do rate conversion, if necessary. Updates (cvt). */
592
if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
593
594
595
return -1; /* shouldn't happen, but just in case... */
}
596
/* Move to final data type. */
597
if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
598
return -1; /* shouldn't happen, but just in case... */
599
}
600
601
cvt->needed = (cvt->filter_index != 0);
602
603
604
return (cvt->needed);
}
605
606
607
typedef int (*SDL_ResampleAudioStreamFunc)(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen);
typedef void (*SDL_ResetAudioStreamResamplerFunc)(SDL_AudioStream *stream);
typedef void (*SDL_CleanupAudioStreamResamplerFunc)(SDL_AudioStream *stream);
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
struct SDL_AudioStream
{
SDL_AudioCVT cvt_before_resampling;
SDL_AudioCVT cvt_after_resampling;
SDL_DataQueue *queue;
Uint8 *work_buffer;
int work_buffer_len;
Uint8 *resample_buffer;
int resample_buffer_len;
int src_sample_frame_size;
SDL_AudioFormat src_format;
Uint8 src_channels;
int src_rate;
int dst_sample_frame_size;
SDL_AudioFormat dst_format;
Uint8 dst_channels;
int dst_rate;
double rate_incr;
Uint8 pre_resample_channels;
int packetlen;
629
630
631
632
void *resampler_state;
SDL_ResampleAudioStreamFunc resampler_func;
SDL_ResetAudioStreamResamplerFunc reset_resampler_func;
SDL_CleanupAudioStreamResamplerFunc cleanup_resampler_func;
633
634
};
635
#ifdef HAVE_LIBSAMPLERATE_H
636
637
638
static int
SDL_ResampleAudioStream_SRC(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
{
639
const int framelen = sizeof(float) * stream->pre_resample_channels;
640
SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
641
642
643
SRC_DATA data;
int result;
644
data.data_in = (float *)inbuf; /* Older versions of libsamplerate had a non-const pointer, but didn't write to it */
645
data.input_frames = inbuflen / framelen;
646
647
648
data.input_frames_used = 0;
data.data_out = outbuf;
649
data.output_frames = outbuflen / framelen;
650
651
652
653
data.end_of_input = 0;
data.src_ratio = stream->rate_incr;
654
result = SRC_src_process(state, &data);
655
if (result != 0) {
656
SDL_SetError("src_process() failed: %s", SRC_src_strerror(result));
657
658
659
660
661
662
663
664
665
666
667
668
return 0;
}
/* If this fails, we need to store them off somewhere */
SDL_assert(data.input_frames_used == data.input_frames);
return data.output_frames_gen * (sizeof(float) * stream->pre_resample_channels);
}
static void
SDL_ResetAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
669
SRC_src_reset((SRC_STATE *)stream->resampler_state);
670
671
672
673
674
}
static void
SDL_CleanupAudioStreamResampler_SRC(SDL_AudioStream *stream)
{
675
SRC_STATE *state = (SRC_STATE *)stream->resampler_state;
676
if (state) {
677
SRC_src_delete(state);
678
679
680
681
682
683
684
685
686
687
688
}
stream->resampler_state = NULL;
stream->resampler_func = NULL;
stream->reset_resampler_func = NULL;
stream->cleanup_resampler_func = NULL;
}
static SDL_bool
SetupLibSampleRateResampling(SDL_AudioStream *stream)
{
689
690
int result = 0;
SRC_STATE *state = NULL;
691
692
693
694
695
696
if (SRC_available) {
state = SRC_src_new(SRC_SINC_FASTEST, stream->pre_resample_channels, &result);
if (!state) {
SDL_SetError("src_new() failed: %s", SRC_src_strerror(result));
}
697
698
}
699
700
if (!state) {
SDL_CleanupAudioStreamResampler_SRC(stream);
701
702
703
704
705
706
707
708
709
710
return SDL_FALSE;
}
stream->resampler_state = state;
stream->resampler_func = SDL_ResampleAudioStream_SRC;
stream->reset_resampler_func = SDL_ResetAudioStreamResampler_SRC;
stream->cleanup_resampler_func = SDL_CleanupAudioStreamResampler_SRC;
return SDL_TRUE;
}
711
#endif /* HAVE_LIBSAMPLERATE_H */
712
713
714
715
716
717
718
719
720
721
722
723
724
725
typedef struct
{
SDL_bool resampler_seeded;
float resampler_state[8];
} SDL_AudioStreamResamplerState;
static int
SDL_ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
{
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
const int chans = (int)stream->pre_resample_channels;
726
SDL_assert(chans <= SDL_arraysize(state->resampler_state));
727
728
if (!state->resampler_seeded) {
729
int i;
730
731
732
733
734
735
for (i = 0; i < chans; i++) {
state->resampler_state[i] = inbuf[i];
}
state->resampler_seeded = SDL_TRUE;
}
736
return SDL_ResampleAudioSimple(chans, stream->rate_incr, state->resampler_state, inbuf, inbuflen, outbuf, outbuflen);
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
}
static void
SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
{
SDL_AudioStreamResamplerState *state = (SDL_AudioStreamResamplerState*)stream->resampler_state;
state->resampler_seeded = SDL_FALSE;
}
static void
SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
{
SDL_free(stream->resampler_state);
}
752
753
754
755
756
757
758
SDL_AudioStream *
SDL_NewAudioStream(const SDL_AudioFormat src_format,
const Uint8 src_channels,
const int src_rate,
const SDL_AudioFormat dst_format,
const Uint8 dst_channels,
const int dst_rate)
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
{
const int packetlen = 4096; /* !!! FIXME: good enough for now. */
Uint8 pre_resample_channels;
SDL_AudioStream *retval;
retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
if (!retval) {
return NULL;
}
/* If increasing channels, do it after resampling, since we'd just
do more work to resample duplicate channels. If we're decreasing, do
it first so we resample the interpolated data instead of interpolating
the resampled data (!!! FIXME: decide if that works in practice, though!). */
pre_resample_channels = SDL_min(src_channels, dst_channels);
retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
retval->src_format = src_format;
retval->src_channels = src_channels;
retval->src_rate = src_rate;
retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
retval->dst_format = dst_format;
retval->dst_channels = dst_channels;
retval->dst_rate = dst_rate;
retval->pre_resample_channels = pre_resample_channels;
retval->packetlen = packetlen;
retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
/* Not resampling? It's an easy conversion (and maybe not even that!). */
if (src_rate == dst_rate) {
retval->cvt_before_resampling.needed = SDL_FALSE;
retval->cvt_before_resampling.len_mult = 1;
791
792
if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
SDL_FreeAudioStream(retval);
793
794
795
796
797
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
}
} else {
/* Don't resample at first. Just get us to Float32 format. */
/* !!! FIXME: convert to int32 on devices without hardware float. */
798
799
if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
SDL_FreeAudioStream(retval);
800
801
802
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
}
803
#ifdef HAVE_LIBSAMPLERATE_H
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
SetupLibSampleRateResampling(retval);
#endif
if (!retval->resampler_func) {
retval->resampler_state = SDL_calloc(1, sizeof(SDL_AudioStreamResamplerState));
if (!retval->resampler_state) {
SDL_FreeAudioStream(retval);
SDL_OutOfMemory();
return NULL;
}
retval->resampler_func = SDL_ResampleAudioStream;
retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
}
819
/* Convert us to the final format after resampling. */
820
821
if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
SDL_FreeAudioStream(retval);
822
823
824
825
826
827
return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
}
}
retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
if (!retval->queue) {
828
SDL_FreeAudioStream(retval);
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
return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
}
return retval;
}
static Uint8 *
EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
{
if (*len < newlen) {
void *ptr = SDL_realloc(*buf, newlen);
if (!ptr) {
SDL_OutOfMemory();
return NULL;
}
*buf = (Uint8 *) ptr;
*len = newlen;
}
return *buf;
}
int
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
{
int buflen = (int) _buflen;
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!buf) {
return SDL_InvalidParamError("buf");
} else if (buflen == 0) {
return 0; /* nothing to do. */
} else if ((buflen % stream->src_sample_frame_size) != 0) {
return SDL_SetError("Can't add partial sample frames");
}
if (stream->cvt_before_resampling.needed) {
const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
if (workbuf == NULL) {
return -1; /* probably out of memory. */
}
SDL_memcpy(workbuf, buf, buflen);
stream->cvt_before_resampling.buf = workbuf;
stream->cvt_before_resampling.len = buflen;
if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
return -1; /* uhoh! */
}
buf = workbuf;
buflen = stream->cvt_before_resampling.len_cvt;
}
if (stream->dst_rate != stream->src_rate) {
const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
if (workbuf == NULL) {
return -1; /* probably out of memory. */
}
887
buflen = stream->resampler_func(stream, (float *) buf, buflen, workbuf, workbuflen);
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
buf = workbuf;
}
if (stream->cvt_after_resampling.needed) {
const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
Uint8 *workbuf;
if (buf == stream->resample_buffer) {
workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
} else {
const int inplace = (buf == stream->work_buffer);
workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
if (workbuf && !inplace) {
SDL_memcpy(workbuf, buf, buflen);
}
}
if (workbuf == NULL) {
return -1; /* probably out of memory. */
}
stream->cvt_after_resampling.buf = workbuf;
stream->cvt_after_resampling.len = buflen;
if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
return -1; /* uhoh! */
}
buf = workbuf;
buflen = stream->cvt_after_resampling.len_cvt;
}
return SDL_WriteToDataQueue(stream->queue, buf, buflen);
}
void
SDL_AudioStreamClear(SDL_AudioStream *stream)
{
if (!stream) {
SDL_InvalidParamError("stream");
} else {
SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
928
929
930
if (stream->reset_resampler_func) {
stream->reset_resampler_func(stream);
}
931
932
933
934
935
936
}
}
/* get converted/resampled data from the stream */
int
937
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, const Uint32 len)
938
939
940
941
942
943
944
945
946
947
948
{
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!buf) {
return SDL_InvalidParamError("buf");
} else if (len == 0) {
return 0; /* nothing to do. */
} else if ((len % stream->dst_sample_frame_size) != 0) {
return SDL_SetError("Can't request partial sample frames");
}
949
return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
950
951
952
953
954
955
956
957
958
959
960
961
962
963
}
/* number of converted/resampled bytes available */
int
SDL_AudioStreamAvailable(SDL_AudioStream *stream)
{
return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
}
/* dispose of a stream */
void
SDL_FreeAudioStream(SDL_AudioStream *stream)
{
if (stream) {
964
965
966
if (stream->cleanup_resampler_func) {
stream->cleanup_resampler_func(stream);
}
967
968
969
970
971
972
973
SDL_FreeDataQueue(stream->queue);
SDL_free(stream->work_buffer);
SDL_free(stream->resample_buffer);
SDL_free(stream);
}
}
974
/* vi: set ts=4 sw=4 expandtab: */