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
|
/*
* Copyright (C) 2010-2012 Hendrik Leppkes
* http://www.1f0.de
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "stdafx.h"
#include "PostProcessor.h"
#include "LAVAudio.h"
#include "Media.h"
extern "C" {
#include "libavutil/intreadwrite.h"
};
// PCM Volume Adjustment Factors, both for integer and float math
// Entrys start at 2 channel mixing, half volume
static int pcm_volume_adjust_integer[7] = {
362, 443, 512, 572, 627, 677, 724
};
static float pcm_volume_adjust_float[7] = {
1.41421356f, 1.73205081f, 2.00000000f, 2.23606798f, 2.44948974f, 2.64575131f, 2.82842712f
};
// SCALE_CA helper macro for SampleCopyAdjust
#define SCALE_CA(sample, iFactor, factor) { \
if (iFactor > 0) { \
sample *= factor; \
sample >>= 8; \
} else { \
sample <<= 8; \
sample /= factor; \
} \
}
//
// Helper Function that reads one sample from pIn, applys the scale specified by iFactor, and writes it to pOut
//
static inline void SampleCopyAdjust(BYTE *pOut, const BYTE *pIn, int iFactor, LAVAudioSampleFormat sfSampleFormat)
{
ASSERT(abs(iFactor) > 1 && abs(iFactor) <= 8);
const int factorIndex = abs(iFactor) - 2;
switch(sfSampleFormat) {
case SampleFormat_U8:
{
uint8_t *pOutSample = pOut;
int32_t sample = *pIn + INT8_MIN;
SCALE_CA(sample, iFactor, pcm_volume_adjust_integer[factorIndex]);
*pOutSample = av_clip_uint8(sample - INT8_MIN) ;
}
break;
case SampleFormat_16:
{
int16_t *pOutSample = (int16_t *)pOut;
int32_t sample = *((int16_t *)pIn);
SCALE_CA(sample, iFactor, pcm_volume_adjust_integer[factorIndex]);
*pOutSample = av_clip_int16(sample);
}
break;
case SampleFormat_24:
{
int32_t sample = (pIn[0] << 8) + (pIn[1] << 16) + (pIn[2] << 24);
sample >>= 8;
SCALE_CA(sample, iFactor, pcm_volume_adjust_integer[factorIndex]);
sample = av_clip(sample, INT24_MIN, INT24_MAX);
pOut[0] = sample & 0xff;
pOut[1] = (sample >> 8) & 0xff;
pOut[2] = (sample >> 16) & 0xff;
}
break;
case SampleFormat_32:
{
int32_t *pOutSample = (int32_t *)pOut;
int64_t sample = *((int32_t *)pIn);
SCALE_CA(sample, iFactor, pcm_volume_adjust_integer[factorIndex]);
*pOutSample = av_clipl_int32(sample);
}
break;
case SampleFormat_FP32:
{
float *pOutSample = (float *)pOut;
float sample = *((float *)pIn);
if (iFactor > 0) {
sample *= pcm_volume_adjust_float[factorIndex];
} else {
sample /= pcm_volume_adjust_float[factorIndex];
}
*pOutSample = av_clipf(sample, -1.0f, 1.0f);
}
break;
default:
ASSERT(0);
break;
}
}
//
// Writes one sample of silence into the buffer
//
static inline void Silence(BYTE *pBuffer, LAVAudioSampleFormat sfSampleFormat)
{
switch (sfSampleFormat) {
case SampleFormat_16:
case SampleFormat_24:
case SampleFormat_32:
case SampleFormat_FP32:
memset(pBuffer, 0, get_byte_per_sample(sfSampleFormat));
break;
case SampleFormat_U8:
*pBuffer = 128U;
break;
default:
ASSERT(0);
}
}
//
// Channel Remapping Processor
// This function can process a PCM buffer of any sample format, and remap the channels
// into any arbitrary layout and channel count.
//
// The samples are copied byte-by-byte, without any conversion or loss.
//
// The ChannelMap is assumed to always have at least uOutChannels valid entries.
// Its layout is in output format:
// Map[0] is the first output channel, and should contain the index in the source stream (or -1 for silence)
// Map[1] is the second output channel
//
// Source channels can be applied multiple times to the Destination, however Volume Adjustments are not performed.
// Furthermore, multiple Source channels cannot be merged into one channel.
// Note that when copying one source channel into multiple destinations, you always want to reduce its volume.
// You can either do this in a second step, or use ExtendedChannelMapping
//
// Examples:
// 5.1 Input Buffer, following map will extract the Center channel, and return it as Mono:
// uOutChannels == 1; map = {2}
//
// Mono Input Buffer, Convert to Stereo
// uOutChannels == 2; map = {0, 0}
//
HRESULT ChannelMapping(BufferDetails *pcm, const unsigned uOutChannels, const ChannelMap map)
{
ExtendedChannelMap extMap;
for (unsigned ch = 0; ch < uOutChannels; ++ch) {
ASSERT(map[ch] >= -1 && map[ch] < pcm->wChannels);
extMap[ch].idx = map[ch];
extMap[ch].factor = 0;
}
return ExtendedChannelMapping(pcm, uOutChannels, extMap);
}
//
// Extended Channel Remapping Processor
// Same functionality as ChannelMapping, except that a factor can be applied to all PCM samples.
//
// For optimization, the factor cannot be freely specified.
// Factors -1, 0, 1 are ignored.
// A Factor of 2 doubles the volume, 3 trippled, etc.
// A Factor of -2 will produce half volume, -3 one third, etc.
// The limit is a factor of 8/-8
//
// Otherwise, see ChannelMapping
HRESULT ExtendedChannelMapping(BufferDetails *pcm, const unsigned uOutChannels, const ExtendedChannelMap extMap)
{
#ifdef DEBUG
ASSERT(pcm && pcm->bBuffer);
ASSERT(uOutChannels > 0 && uOutChannels <= 8);
for(unsigned idx = 0; idx < uOutChannels; ++idx) {
ASSERT(extMap[idx].idx >= -1 && extMap[idx].idx < pcm->wChannels);
}
#endif
// Sample Size
const unsigned uSampleSize = get_byte_per_sample(pcm->sfFormat);
// New Output Buffer
GrowableArray<BYTE> *out = new GrowableArray<BYTE>();
out->SetSize(uOutChannels * pcm->nSamples * uSampleSize);
const BYTE *pIn = pcm->bBuffer->Ptr();
BYTE *pOut = out->Ptr();
for(unsigned i = 0; i < pcm->nSamples; ++i) {
for (unsigned ch = 0; ch < uOutChannels; ++ch) {
if (extMap[ch].idx >= 0) {
if (!extMap[ch].factor || abs(extMap[ch].factor) == 1)
memcpy(pOut, pIn + (extMap[ch].idx * uSampleSize), uSampleSize);
else
SampleCopyAdjust(pOut, pIn + (extMap[ch].idx * uSampleSize), extMap[ch].factor, pcm->sfFormat);
} else
Silence(pOut, pcm->sfFormat);
pOut += uSampleSize;
}
pIn += uSampleSize * pcm->wChannels;
}
// Apply changes to buffer
delete pcm->bBuffer;
pcm->bBuffer = out;
pcm->wChannels = uOutChannels;
return S_OK;
}
#define CHL_CONTAINS_ALL(l, m) (((l) & (m)) == (m))
#define CHL_ALL_OR_NONE(l, m) (((l) & (m)) == (m) || ((l) & (m)) == 0)
HRESULT CLAVAudio::CheckChannelLayoutConformity(DWORD dwLayout)
{
int channels = av_get_channel_layout_nb_channels(dwLayout);
// We require multi-channel and at least containing stereo
if (!CHL_CONTAINS_ALL(dwLayout, AV_CH_LAYOUT_STEREO) || channels == 2)
goto noprocessing;
// We do not know what to do with "top" channels
if (dwLayout & (AV_CH_TOP_CENTER|AV_CH_TOP_FRONT_LEFT|AV_CH_TOP_FRONT_CENTER|AV_CH_TOP_FRONT_RIGHT|AV_CH_TOP_BACK_LEFT|AV_CH_TOP_BACK_CENTER|AV_CH_TOP_BACK_RIGHT)) {
DbgLog((LOG_ERROR, 10, L"::CheckChannelLayoutConformity(): Layout with top channels is not supported (mask: 0x%x)", dwLayout));
goto noprocessing;
}
// We need either both surround channels, or none. One of a type is not supported
if (!CHL_ALL_OR_NONE(dwLayout, AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)
|| !CHL_ALL_OR_NONE(dwLayout, AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)
|| !CHL_ALL_OR_NONE(dwLayout, AV_CH_FRONT_LEFT_OF_CENTER|AV_CH_FRONT_RIGHT_OF_CENTER)) {
DbgLog((LOG_ERROR, 10, L"::CheckChannelLayoutConformity(): Layout with only one surround channel is not supported (mask: 0x%x)", dwLayout));
goto noprocessing;
}
// No need to process full 5.1/6.1 layouts, or any 8 channel layouts
if (dwLayout == AV_CH_LAYOUT_5POINT1
|| dwLayout == AV_CH_LAYOUT_5POINT1_BACK
|| dwLayout == AV_CH_LAYOUT_6POINT1_BACK
|| channels == 8) {
DbgLog((LOG_TRACE, 10, L"::CheckChannelLayoutConformity(): Layout is already a default layout (mask: 0x%x)", dwLayout));
goto noprocessing;
}
// Check 5.1 channels
if (CHL_CONTAINS_ALL(AV_CH_LAYOUT_5POINT1, dwLayout) /* 5.1 with side channels */
|| CHL_CONTAINS_ALL(AV_CH_LAYOUT_5POINT1_BACK, dwLayout) /* 5.1 with back channels */
|| CHL_CONTAINS_ALL(LAV_CH_LAYOUT_5POINT1_WIDE, dwLayout) /* 5.1 with side-front channels */
|| CHL_CONTAINS_ALL(LAV_CH_LAYOUT_5POINT1_BC, dwLayout)) /* 3/1/x layouts, front channels with a back center */
return Create51Conformity(dwLayout);
// Check 6.1 channels (5.1 layouts + Back Center)
if (CHL_CONTAINS_ALL(AV_CH_LAYOUT_6POINT1, dwLayout) /* 6.1 with side channels */
|| CHL_CONTAINS_ALL(AV_CH_LAYOUT_6POINT1_BACK, dwLayout) /* 6.1 with back channels */
|| CHL_CONTAINS_ALL(LAV_CH_LAYOUT_5POINT1_WIDE|AV_CH_BACK_CENTER, dwLayout)) /* 6.1 with side-front channels */
return Create61Conformity(dwLayout);
// Check 7.1 channels
if (CHL_CONTAINS_ALL(AV_CH_LAYOUT_7POINT1, dwLayout) /* 7.1 with side+back channels */
|| CHL_CONTAINS_ALL(AV_CH_LAYOUT_7POINT1_WIDE, dwLayout) /* 7.1 with front-side+back channels */
|| CHL_CONTAINS_ALL(LAV_CH_LAYOUT_7POINT1_EXTRAWIDE, dwLayout)) /* 7.1 with front-side+side channels */
return Create71Conformity(dwLayout);
noprocessing:
m_bChannelMappingRequired = FALSE;
return S_FALSE;
}
HRESULT CLAVAudio::Create51Conformity(DWORD dwLayout)
{
DbgLog((LOG_TRACE, 10, L"::Create51Conformity(): Creating 5.1 default layout (mask: 0x%x)", dwLayout));
int ch = 0;
ExtChMapClear(&m_ChannelMap);
// All layouts we support have to contain L/R
ExtChMapSet(&m_ChannelMap, 0, ch++, 0);
ExtChMapSet(&m_ChannelMap, 1, ch++, 0);
// Center channel
if (dwLayout & AV_CH_FRONT_CENTER)
ExtChMapSet(&m_ChannelMap, 2, ch++, 0);
// LFE
if (dwLayout & AV_CH_LOW_FREQUENCY)
ExtChMapSet(&m_ChannelMap, 3, ch++, 0);
// Back/Side
if (dwLayout & (AV_CH_SIDE_LEFT|AV_CH_BACK_LEFT|AV_CH_FRONT_LEFT_OF_CENTER)) {
ExtChMapSet(&m_ChannelMap, 4, ch++, 0);
ExtChMapSet(&m_ChannelMap, 5, ch++, 0);
// Back Center
} else if (dwLayout & AV_CH_BACK_CENTER) {
ExtChMapSet(&m_ChannelMap, 4, ch, -2);
ExtChMapSet(&m_ChannelMap, 5, ch++, -2);
}
m_bChannelMappingRequired = TRUE;
m_ChannelMapOutputChannels = 6;
m_ChannelMapOutputLayout = AV_CH_LAYOUT_5POINT1_BACK;
return S_OK;
}
HRESULT CLAVAudio::Create61Conformity(DWORD dwLayout)
{
if (m_settings.Expand61) {
DbgLog((LOG_TRACE, 10, L"::Create61Conformity(): Expanding to 7.1"));
return Create71Conformity(dwLayout);
}
DbgLog((LOG_TRACE, 10, L"::Create61Conformity(): Creating 6.1 default layout (mask: 0x%x)", dwLayout));
int ch = 0;
ExtChMapClear(&m_ChannelMap);
// All layouts we support have to contain L/R
ExtChMapSet(&m_ChannelMap, 0, ch++, 0);
ExtChMapSet(&m_ChannelMap, 1, ch++, 0);
// Center channel
if (dwLayout & AV_CH_FRONT_CENTER)
ExtChMapSet(&m_ChannelMap, 2, ch++, 0);
// LFE
if (dwLayout & AV_CH_LOW_FREQUENCY)
ExtChMapSet(&m_ChannelMap, 3, ch++, 0);
// Back channels, if before BC
if (dwLayout & (AV_CH_BACK_LEFT|AV_CH_FRONT_LEFT_OF_CENTER)) {
DbgLog((LOG_TRACE, 10, L"::Create61Conformity(): Using surround channels *before* BC"));
ExtChMapSet(&m_ChannelMap, 4, ch++, 0);
ExtChMapSet(&m_ChannelMap, 5, ch++, 0);
}
// Back Center
if (dwLayout & AV_CH_BACK_CENTER)
ExtChMapSet(&m_ChannelMap, 6, ch++, 0);
// Back channels, if after BC
if (dwLayout & AV_CH_SIDE_LEFT) {
DbgLog((LOG_TRACE, 10, L"::Create61Conformity(): Using surround channels *after* BC"));
ExtChMapSet(&m_ChannelMap, 4, ch++, 0);
ExtChMapSet(&m_ChannelMap, 5, ch++, 0);
}
m_bChannelMappingRequired = TRUE;
m_ChannelMapOutputChannels = 7;
m_ChannelMapOutputLayout = AV_CH_LAYOUT_6POINT1_BACK;
return S_OK;
}
HRESULT CLAVAudio::Create71Conformity(DWORD dwLayout)
{
DbgLog((LOG_TRACE, 10, L"::Create71Conformity(): Creating 7.1 default layout (mask: 0x%x)", dwLayout));
int ch = 0;
ExtChMapClear(&m_ChannelMap);
// All layouts we support have to contain L/R
ExtChMapSet(&m_ChannelMap, 0, ch++, 0);
ExtChMapSet(&m_ChannelMap, 1, ch++, 0);
// Center channel
if (dwLayout & AV_CH_FRONT_CENTER)
ExtChMapSet(&m_ChannelMap, 2, ch++, 0);
// LFE
if (dwLayout & AV_CH_LOW_FREQUENCY)
ExtChMapSet(&m_ChannelMap, 3, ch++, 0);
// Back channels
if (dwLayout & AV_CH_BACK_CENTER) {
DbgLog((LOG_TRACE, 10, L"::Create71Conformity(): Usign BC to fill back channels"));
if (dwLayout & AV_CH_SIDE_LEFT) {
DbgLog((LOG_TRACE, 10, L"::Create71Conformity(): Using BC before side-surround channels"));
ExtChMapSet(&m_ChannelMap, 4, ch, -2);
ExtChMapSet(&m_ChannelMap, 5, ch++, -2);
ExtChMapSet(&m_ChannelMap, 6, ch++, 0);
ExtChMapSet(&m_ChannelMap, 7, ch++, 0);
} else {
DbgLog((LOG_TRACE, 10, L"::Create71Conformity(): Using BC after side-surround channels"));
ExtChMapSet(&m_ChannelMap, 6, ch++, 0);
ExtChMapSet(&m_ChannelMap, 7, ch++, 0);
ExtChMapSet(&m_ChannelMap, 4, ch, -2);
ExtChMapSet(&m_ChannelMap, 5, ch++, -2);
}
} else {
DbgLog((LOG_TRACE, 10, L"::Create71Conformity(): Using original 4 surround channels"));
ExtChMapSet(&m_ChannelMap, 4, ch++, 0);
ExtChMapSet(&m_ChannelMap, 5, ch++, 0);
ExtChMapSet(&m_ChannelMap, 6, ch++, 0);
ExtChMapSet(&m_ChannelMap, 7, ch++, 0);
}
m_bChannelMappingRequired = TRUE;
m_ChannelMapOutputChannels = 8;
m_ChannelMapOutputLayout = AV_CH_LAYOUT_7POINT1;
return S_OK;
}
static DWORD sanitize_mask(DWORD mask, AVCodecID codec)
{
DWORD newmask = mask;
// Alot of codecs set 6.1/6.0 wrong..
// Only these codecs we can trust to properly set BL/BR + BC layouts
if (codec != AV_CODEC_ID_DTS) {
// 6.1
if (mask == (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_BACK_CENTER))
newmask = AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER;
// 6.0
if (mask == (AV_CH_LAYOUT_5POINT0_BACK|AV_CH_BACK_CENTER))
newmask = AV_CH_LAYOUT_5POINT0|AV_CH_BACK_CENTER;
}
// The reverse, TrueHD sets SL/SR when it actually should be BL/BR
if (codec == AV_CODEC_ID_TRUEHD) {
// 6.1
if(mask == (AV_CH_LAYOUT_5POINT1|AV_CH_BACK_CENTER))
newmask = AV_CH_LAYOUT_5POINT1_BACK|AV_CH_BACK_CENTER;
// 6.0
if(mask == (AV_CH_LAYOUT_5POINT0|AV_CH_BACK_CENTER))
newmask = AV_CH_LAYOUT_5POINT0_BACK|AV_CH_BACK_CENTER;
}
// Prefer the 5.1 BACK mask
if (mask == AV_CH_LAYOUT_5POINT1)
newmask = AV_CH_LAYOUT_5POINT1_BACK;
return newmask;
}
template <LAVAudioSampleFormat in, LAVAudioSampleFormat out>
static void SampleConvert(const BYTE *pPCMIn, BYTE *pPCMOut, unsigned samples, unsigned channels)
{
unsigned i, ch;
const int bppIn = get_byte_per_sample(in);
const int bppOut = get_byte_per_sample(out);
for (i = 0; i < samples; i++) {
for (ch = 0; ch < channels; ch++) {
if (in == SampleFormat_16) {
int32_t value = *(int16_t *)pPCMIn;
switch (out) {
case SampleFormat_24: AV_WL24(pPCMOut, ((int32_t)value << 8)); break;
case SampleFormat_32: *(int32_t *)pPCMOut = ((int32_t)value << 16); break;
case SampleFormat_U8: *pPCMOut = (value >> 8) + 0x80; break;
case SampleFormat_FP32: *(float *)pPCMOut = value * (1.0f / (1U<<15)); break;
}
} else if (in == SampleFormat_24 || in == SampleFormat_32) {
int32_t value = 0;
if (in == SampleFormat_24)
value = (pPCMIn[0] << 8) + (pPCMIn[1] << 16) + (pPCMIn[2] << 24);
else
value = *(int32_t *)pPCMIn;
switch (out) {
case SampleFormat_16: *(int16_t *)pPCMOut = value >> 16; break;
case SampleFormat_24: AV_WL24(pPCMOut, value >> 8); break;
case SampleFormat_32: *(int32_t *)pPCMOut = value; break;
case SampleFormat_U8: *pPCMOut = (value >> 24) + 0x80; break;
case SampleFormat_FP32: *(float *)pPCMOut = value * (1.0f / (1U<<31)); break;
}
} else if (in == SampleFormat_U8) {
int32_t value = *pPCMIn - 0x80;
switch (out) {
case SampleFormat_16: *(int16_t *)pPCMOut = value << 8; break;
case SampleFormat_24: AV_WL24(pPCMOut, value << 16); break;
case SampleFormat_32: *(int32_t *)pPCMOut = value << 24; break;
case SampleFormat_FP32: *(float *)pPCMOut = value * (1.0f / (1U<<7)); break;
}
} else if (in == SampleFormat_FP32) {
float value = *(float *)pPCMIn;
switch (out) {
case SampleFormat_16: *(int16_t *)pPCMOut = av_clip_int16(int(value * (1U<<15))); break;
case SampleFormat_24: AV_WL24(pPCMOut, av_clip(int(value * (1U<<23)), INT24_MIN, INT24_MAX)); break;
case SampleFormat_32: *(int32_t *)pPCMOut = av_clipl_int32(int64_t(value * (1U<<31))); break;
case SampleFormat_U8: *pPCMOut = av_clip_uint8(int(value * (1U<<7)) + 0x80); break;
}
}
pPCMIn += bppIn;
pPCMOut += bppOut;
}
}
}
#define CONV_MAP(in, out) \
if (pcm->sfFormat == in && outputFormat == out) { \
SampleConvert<in, out>(pcm->bBuffer->Ptr(), pcmOut->Ptr(), pcm->nSamples, pcm->wChannels); \
}
HRESULT CLAVAudio::ConvertSampleFormat(BufferDetails *pcm, LAVAudioSampleFormat outputFormat)
{
const unsigned uSampleSize = get_byte_per_sample(outputFormat);
// New Output Buffer
GrowableArray<BYTE> *pcmOut = new GrowableArray<BYTE>();
pcmOut->SetSize(pcm->wChannels * pcm->nSamples * uSampleSize);
CONV_MAP(SampleFormat_16, SampleFormat_24)
else CONV_MAP(SampleFormat_16, SampleFormat_32)
else CONV_MAP(SampleFormat_16, SampleFormat_U8)
else CONV_MAP(SampleFormat_16, SampleFormat_FP32)
else CONV_MAP(SampleFormat_24, SampleFormat_16)
else CONV_MAP(SampleFormat_24, SampleFormat_32)
else CONV_MAP(SampleFormat_24, SampleFormat_U8)
else CONV_MAP(SampleFormat_24, SampleFormat_FP32)
else CONV_MAP(SampleFormat_32, SampleFormat_16)
else CONV_MAP(SampleFormat_32, SampleFormat_24)
else CONV_MAP(SampleFormat_32, SampleFormat_U8)
else CONV_MAP(SampleFormat_32, SampleFormat_FP32)
else CONV_MAP(SampleFormat_FP32, SampleFormat_16)
else CONV_MAP(SampleFormat_FP32, SampleFormat_24)
else CONV_MAP(SampleFormat_FP32, SampleFormat_32)
else CONV_MAP(SampleFormat_FP32, SampleFormat_U8)
else CONV_MAP(SampleFormat_U8, SampleFormat_16)
else CONV_MAP(SampleFormat_U8, SampleFormat_24)
else CONV_MAP(SampleFormat_U8, SampleFormat_32)
else CONV_MAP(SampleFormat_U8, SampleFormat_FP32)
else ASSERT(0);
// Apply changes to buffer
delete pcm->bBuffer;
pcm->bBuffer = pcmOut;
pcm->sfFormat = outputFormat;
pcm->wBitsPerSample = get_byte_per_sample(outputFormat) << 3;
return S_OK;
}
HRESULT CLAVAudio::Truncate32Buffer(BufferDetails *buffer)
{
ASSERT(buffer->sfFormat == SampleFormat_32 && buffer->wBitsPerSample <= 24);
// Determine the bytes per sample to keep. Cut a 16-bit sample to 24 if 16-bit is disabled for some reason
const int bytes_per_sample = buffer->wBitsPerSample <= 16 && GetSampleFormat(SampleFormat_16) ? 2 : 3;
if (bytes_per_sample == 3 && !GetSampleFormat(SampleFormat_24))
return S_FALSE;
const int skip = 4 - bytes_per_sample;
const DWORD size = (buffer->nSamples * buffer->wChannels) * bytes_per_sample;
GrowableArray<BYTE> *pcmOut = new GrowableArray<BYTE>();
pcmOut->SetSize(size);
const BYTE *pDataIn = buffer->bBuffer->Ptr();
BYTE *pDataOut = pcmOut->Ptr();
pDataIn += skip;
for (unsigned int i = 0; i < buffer->nSamples; ++i) {
for(int ch = 0; ch < buffer->wChannels; ++ch) {
memcpy(pDataOut, pDataIn, bytes_per_sample);
pDataOut += bytes_per_sample;
pDataIn += 4;
}
}
delete buffer->bBuffer;
buffer->bBuffer = pcmOut;
buffer->sfFormat = bytes_per_sample == 3 ? SampleFormat_24 : SampleFormat_16;
return S_OK;
}
HRESULT CLAVAudio::PerformMixing(BufferDetails *buffer)
{
int ret = 0;
DWORD dwMixingLayout = m_dwOverrideMixer ? m_dwOverrideMixer : m_settings.MixingLayout;
// Check if we need mixing, either already in target mask or in stereo (no downmixing from stereo)
if (buffer->dwChannelMask == dwMixingLayout || (buffer->wChannels <= 2 && (m_settings.MixingFlags & LAV_MIXING_FLAG_UNTOUCHED_STEREO) && !m_dwOverrideMixer))
return S_FALSE;
// Sadly, we need to convert this, avresample has no 24-bit mode
if (buffer->sfFormat == SampleFormat_24) {
ConvertSampleFormat(buffer, SampleFormat_32);
}
if (buffer->dwChannelMask != m_DecodeLayoutSanified || (!m_avrContext && !m_bAVResampleFailed) || m_bMixingSettingsChanged || m_dwRemixLayout != dwMixingLayout) {
m_bAVResampleFailed = FALSE;
m_bMixingSettingsChanged = FALSE;
if (m_avrContext) {
avresample_close(m_avrContext);
avresample_free(&m_avrContext);
}
m_DecodeLayoutSanified = buffer->dwChannelMask;
// We prefer to mix to FP32, so try to use that
m_sfRemixFormat = GetBestAvailableSampleFormat(SampleFormat_FP32);
// avresample has no 24-bit mode
if (m_sfRemixFormat == SampleFormat_24)
m_sfRemixFormat = SampleFormat_32;
BOOL bNormalize = !!(m_settings.MixingFlags & LAV_MIXING_FLAG_NORMALIZE_MATRIX);
// Force FP32 output from remixing when not performing normalization
if (!bNormalize)
m_sfRemixFormat = SampleFormat_FP32;
m_avrContext = avresample_alloc_context();
av_opt_set_int(m_avrContext, "in_channel_layout", buffer->dwChannelMask, 0);
av_opt_set_int(m_avrContext, "in_sample_fmt", get_ff_sample_fmt(buffer->sfFormat), 0);
av_opt_set_int(m_avrContext, "out_channel_layout", dwMixingLayout, 0);
av_opt_set_int(m_avrContext, "out_sample_fmt", get_ff_sample_fmt(m_sfRemixFormat), 0);
// Open Resample Context
ret = avresample_open(m_avrContext);
if (ret < 0) {
DbgLog((LOG_ERROR, 10, L"avresample_open failed, layout in: %x, out: %x, sample fmt in: %d, out: %d", buffer->dwChannelMask, dwMixingLayout, buffer->sfFormat, m_sfRemixFormat));
goto setuperr;
}
// Create Matrix
int in_ch = buffer->wChannels;
int out_ch = av_get_channel_layout_nb_channels(dwMixingLayout);
double *matrix_dbl = (double *)av_mallocz(in_ch * out_ch * sizeof(*matrix_dbl));
const double center_mix_level = (double)m_settings.MixingCenterLevel / 10000.0 / M_SQRT1_2;
const double surround_mix_level = (double)m_settings.MixingSurroundLevel / 10000.0;
const double lfe_mix_level = (double)m_settings.MixingLFELevel / 10000.0 / (dwMixingLayout == AV_CH_LAYOUT_MONO ? 1.0 : M_SQRT1_2);
ret = avresample_build_matrix(buffer->dwChannelMask, dwMixingLayout, center_mix_level, surround_mix_level, lfe_mix_level, bNormalize, matrix_dbl, in_ch, (AVMatrixEncoding)m_settings.MixingMode);
if (ret < 0) {
DbgLog((LOG_ERROR, 10, L"avresample_build_matrix failed, layout in: %x, out: %x, sample fmt in: %d, out: %d", buffer->dwChannelMask, dwMixingLayout, buffer->sfFormat, m_sfRemixFormat));
av_free(matrix_dbl);
goto setuperr;
}
// Set Matrix on the context
ret = avresample_set_matrix(m_avrContext, matrix_dbl, in_ch);
av_free(matrix_dbl);
if (ret < 0) {
DbgLog((LOG_ERROR, 10, L"avresample_set_matrix failed, layout in: %x, out: %x, sample fmt in: %d, out: %d", buffer->dwChannelMask, dwMixingLayout, buffer->sfFormat, m_sfRemixFormat));
goto setuperr;
}
m_dwRemixLayout = dwMixingLayout;
}
if (!m_avrContext)
return S_FALSE;
GrowableArray<BYTE> *pcmOut = new GrowableArray<BYTE>();
pcmOut->Allocate(FFALIGN(buffer->nSamples, 32) * av_get_channel_layout_nb_channels(m_dwRemixLayout) * get_byte_per_sample(m_sfRemixFormat));
BYTE *pOut = pcmOut->Ptr();
BYTE *pIn = buffer->bBuffer->Ptr();
ret = avresample_convert(m_avrContext, &pOut, pcmOut->GetAllocated(), buffer->nSamples, &pIn, buffer->bBuffer->GetAllocated(), buffer->nSamples);
if (ret < 0) {
DbgLog((LOG_ERROR, 10, L"avresample_convert failed"));
delete pcmOut;
return S_FALSE;
}
if (!(m_settings.MixingFlags & LAV_MIXING_FLAG_NORMALIZE_MATRIX) && (m_settings.MixingFlags & LAV_MIXING_FLAG_CLIP_PROTECTION)) {
ASSERT(m_sfRemixFormat == SampleFormat_FP32);
DWORD dwSamples = av_get_channel_layout_nb_channels(m_dwRemixLayout) * buffer->nSamples;
BOOL bNeedNormalize = (m_fMixingClipThreshold > 1.0f);
float *pfOut = (float *)pOut;
for (DWORD i = 0; i < dwSamples; i++) {
const float fVal = abs(pfOut[i]);
if (fVal > m_fMixingClipThreshold) {
m_fMixingClipThreshold = fVal;
bNeedNormalize = TRUE;
DbgLog((LOG_TRACE, 10, L"PerformMixing: Clipping Protection engaged with coeff of %f", m_fMixingClipThreshold));
}
if (bNeedNormalize) {
pfOut[i] /= m_fMixingClipThreshold;
}
}
}
delete buffer->bBuffer;
buffer->bBuffer = pcmOut;
buffer->dwChannelMask = m_dwRemixLayout;
buffer->sfFormat = m_sfRemixFormat;
buffer->wBitsPerSample = get_byte_per_sample(m_sfRemixFormat) << 3;
buffer->wChannels = av_get_channel_layout_nb_channels(m_dwRemixLayout);
buffer->bBuffer->SetSize(buffer->wChannels * buffer->nSamples * get_byte_per_sample(m_sfRemixFormat));
return S_OK;
setuperr:
avresample_free(&m_avrContext);
m_bAVResampleFailed = TRUE;
return S_FALSE;
}
HRESULT CLAVAudio::PostProcess(BufferDetails *buffer)
{
buffer->dwChannelMask = sanitize_mask(buffer->dwChannelMask, m_nCodecId);
int layout_channels = av_get_channel_layout_nb_channels(buffer->dwChannelMask);
// Validate channel mask
if (!buffer->dwChannelMask || layout_channels != buffer->wChannels) {
buffer->dwChannelMask = get_channel_mask(buffer->wChannels);
if (!buffer->dwChannelMask && buffer->wChannels <= 2) {
buffer->dwChannelMask = buffer->wChannels == 2 ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
}
}
if (m_settings.MixingEnabled || m_dwOverrideMixer) {
PerformMixing(buffer);
}
// Remap to standard configurations, if requested
else if (m_settings.OutputStandardLayout) {
if (buffer->dwChannelMask != m_DecodeLayoutSanified) {
m_DecodeLayoutSanified = buffer->dwChannelMask;
CheckChannelLayoutConformity(buffer->dwChannelMask);
}
if (m_bChannelMappingRequired) {
ExtendedChannelMapping(buffer, m_ChannelMapOutputChannels, m_ChannelMap);
buffer->dwChannelMask = m_ChannelMapOutputLayout;
}
}
// Mono -> Stereo expansion
if (buffer->wChannels == 1 && m_settings.ExpandMono) {
ExtendedChannelMap map = {{0,-2}, {0, -2}};
ExtendedChannelMapping(buffer, 2, map);
buffer->dwChannelMask = AV_CH_LAYOUT_STEREO;
}
// 6.1 -> 7.1 expansion
if (m_settings.Expand61 && buffer->dwChannelMask == (AV_CH_LAYOUT_5POINT1_BACK|AV_CH_BACK_CENTER)) {
ExtendedChannelMap map = {{0,0}, {1,0}, {2,0}, {3,0}, {6,-2}, {6,-2}, {4,0}, {5,0}};
ExtendedChannelMapping(buffer, 8, map);
buffer->dwChannelMask = AV_CH_LAYOUT_7POINT1;
}
if (m_bVolumeStats) {
UpdateVolumeStats(*buffer);
}
// Truncate 24-in-32 to real 24
if (buffer->sfFormat == SampleFormat_32 && buffer->wBitsPerSample && buffer->wBitsPerSample <= 24) {
Truncate32Buffer(buffer);
}
// Convert Sample format, if necessary
LAVAudioSampleFormat outputFormat = GetBestAvailableSampleFormat(buffer->sfFormat);
if (outputFormat != buffer->sfFormat) {
ConvertSampleFormat(buffer, outputFormat);
}
return S_OK;
}
|