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
|
/*
* Copyright (C) 2007 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "AudioResampler"
//#define LOG_NDEBUG 0
#include <stdint.h>
#include <stdlib.h>
#include <sys/types.h>
#include <cutils/log.h>
#include <cutils/properties.h>
#include "AudioResampler.h"
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
#ifdef __arm__
#include <machine/cpu-features.h>
#endif
namespace android {
#ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option
#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
#endif // __ARM_HAVE_HALFWORD_MULTIPLY
// ----------------------------------------------------------------------------
class AudioResamplerOrder1 : public AudioResampler {
public:
AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
}
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
private:
// number of bits used in interpolation multiply - 15 bits avoids overflow
static const int kNumInterpBits = 15;
// bits to shift the phase fraction down to avoid overflow
static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
void init() {}
void resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
void resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement);
void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement);
#endif // ASM_ARM_RESAMP1
static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
}
static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
*frac += inc;
*index += (size_t)(*frac >> kNumPhaseBits);
*frac &= kPhaseMask;
}
int mX0L;
int mX0R;
};
bool AudioResampler::qualityIsSupported(src_quality quality)
{
switch (quality) {
case DEFAULT_QUALITY:
case LOW_QUALITY:
case MED_QUALITY:
case HIGH_QUALITY:
case VERY_HIGH_QUALITY:
return true;
default:
return false;
}
}
// ----------------------------------------------------------------------------
static pthread_once_t once_control = PTHREAD_ONCE_INIT;
static AudioResampler::src_quality defaultQuality = AudioResampler::DEFAULT_QUALITY;
void AudioResampler::init_routine()
{
char value[PROPERTY_VALUE_MAX];
if (property_get("af.resampler.quality", value, NULL) > 0) {
char *endptr;
unsigned long l = strtoul(value, &endptr, 0);
if (*endptr == '\0') {
defaultQuality = (src_quality) l;
ALOGD("forcing AudioResampler quality to %d", defaultQuality);
if (defaultQuality < DEFAULT_QUALITY || defaultQuality > VERY_HIGH_QUALITY) {
defaultQuality = DEFAULT_QUALITY;
}
}
}
}
uint32_t AudioResampler::qualityMHz(src_quality quality)
{
switch (quality) {
default:
case DEFAULT_QUALITY:
case LOW_QUALITY:
return 3;
case MED_QUALITY:
return 6;
case HIGH_QUALITY:
return 20;
case VERY_HIGH_QUALITY:
return 34;
}
}
static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, should be tunable
static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
static uint32_t currentMHz = 0;
AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
int32_t sampleRate, src_quality quality) {
bool atFinalQuality;
if (quality == DEFAULT_QUALITY) {
// read the resampler default quality property the first time it is needed
int ok = pthread_once(&once_control, init_routine);
if (ok != 0) {
ALOGE("%s pthread_once failed: %d", __func__, ok);
}
quality = defaultQuality;
atFinalQuality = false;
} else {
atFinalQuality = true;
}
// naive implementation of CPU load throttling doesn't account for whether resampler is active
pthread_mutex_lock(&mutex);
for (;;) {
uint32_t deltaMHz = qualityMHz(quality);
uint32_t newMHz = currentMHz + deltaMHz;
if ((qualityIsSupported(quality) && newMHz <= maxMHz) || atFinalQuality) {
ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
currentMHz, newMHz, deltaMHz, quality);
currentMHz = newMHz;
break;
}
// not enough CPU available for proposed quality level, so try next lowest level
switch (quality) {
default:
case DEFAULT_QUALITY:
case LOW_QUALITY:
atFinalQuality = true;
break;
case MED_QUALITY:
quality = LOW_QUALITY;
break;
case HIGH_QUALITY:
quality = MED_QUALITY;
break;
case VERY_HIGH_QUALITY:
quality = HIGH_QUALITY;
break;
}
}
pthread_mutex_unlock(&mutex);
AudioResampler* resampler;
switch (quality) {
default:
case DEFAULT_QUALITY:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
case MED_QUALITY:
ALOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
break;
case HIGH_QUALITY:
ALOGV("Create HIGH_QUALITY sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
case VERY_HIGH_QUALITY:
ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
break;
}
// initialize resampler
resampler->init();
return resampler;
}
AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
int32_t sampleRate, src_quality quality) :
mBitDepth(bitDepth), mChannelCount(inChannelCount),
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
mPhaseFraction(0), mLocalTimeFreq(0),
mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
// sanity check on format
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
inChannelCount);
// ALOG_ASSERT(0);
}
if (sampleRate <= 0) {
ALOGE("Unsupported sample rate %d Hz", sampleRate);
}
// initialize common members
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
}
AudioResampler::~AudioResampler() {
pthread_mutex_lock(&mutex);
src_quality quality = getQuality();
uint32_t deltaMHz = qualityMHz(quality);
int32_t newMHz = currentMHz - deltaMHz;
ALOGV("resampler load %u -> %d MHz due to delta -%u MHz from quality %d",
currentMHz, newMHz, deltaMHz, quality);
LOG_ALWAYS_FATAL_IF(newMHz < 0, "negative resampler load %d MHz", newMHz);
currentMHz = newMHz;
pthread_mutex_unlock(&mutex);
}
void AudioResampler::setSampleRate(int32_t inSampleRate) {
mInSampleRate = inSampleRate;
mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
}
void AudioResampler::setVolume(int16_t left, int16_t right) {
// TODO: Implement anti-zipper filter
mVolume[0] = left;
mVolume[1] = right;
}
void AudioResampler::setLocalTimeFreq(uint64_t freq) {
mLocalTimeFreq = freq;
}
void AudioResampler::setPTS(int64_t pts) {
mPTS = pts;
}
int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
if (mPTS == AudioBufferProvider::kInvalidPTS) {
return AudioBufferProvider::kInvalidPTS;
} else {
return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
}
}
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
mBuffer.frameCount = 0;
}
// ----------------------------------------------------------------------------
void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
// should never happen, but we overflow if it does
// ALOG_ASSERT(outFrameCount < 32767);
// select the appropriate resampler
switch (mChannelCount) {
case 1:
resampleMono16(out, outFrameCount, provider);
break;
case 2:
resampleStereo16(out, outFrameCount, provider);
break;
}
}
void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
int32_t vr = mVolume[1];
size_t inputIndex = mInputIndex;
uint32_t phaseFraction = mPhaseFraction;
uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto resampleStereo16_exit;
}
// ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
provider->releaseBuffer(&mBuffer);
// mBuffer.frameCount == 0 now so we reload a new buffer
}
int16_t *in = mBuffer.i16;
// handle boundary case
while (inputIndex == 0) {
// ALOGE("boundary case");
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
if (outputIndex == outputSampleCount)
break;
}
// process input samples
// ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
int32_t* maxOutPt;
int32_t maxInIdx;
maxOutPt = out + (outputSampleCount - 2); // 2 because 2 frames per loop
maxInIdx = mBuffer.frameCount - 2;
AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
phaseFraction, phaseIncrement);
}
#endif // ASM_ARM_RESAMP1
while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
in[inputIndex*2], phaseFraction);
out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
in[inputIndex*2+1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
// ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
// ALOGE("buffer done, new input index %d", inputIndex);
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
provider->releaseBuffer(&mBuffer);
// verify that the releaseBuffer resets the buffer frameCount
// ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
// ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleStereo16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
}
void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) {
int32_t vl = mVolume[0];
int32_t vr = mVolume[1];
size_t inputIndex = mInputIndex;
uint32_t phaseFraction = mPhaseFraction;
uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
// ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
mX0L = mBuffer.i16[mBuffer.frameCount-1];
provider->releaseBuffer(&mBuffer);
// mBuffer.frameCount == 0 now so we reload a new buffer
}
int16_t *in = mBuffer.i16;
// handle boundary case
while (inputIndex == 0) {
// ALOGE("boundary case");
int32_t sample = Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
Advance(&inputIndex, &phaseFraction, phaseIncrement);
if (outputIndex == outputSampleCount)
break;
}
// process input samples
// ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
int32_t* maxOutPt;
int32_t maxInIdx;
maxOutPt = out + (outputSampleCount - 2);
maxInIdx = (int32_t)mBuffer.frameCount - 2;
AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
phaseFraction, phaseIncrement);
}
#endif // ASM_ARM_RESAMP1
while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
// ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
// ALOGE("buffer done, new input index %d", inputIndex);
mX0L = mBuffer.i16[mBuffer.frameCount-1];
provider->releaseBuffer(&mBuffer);
// verify that the releaseBuffer resets the buffer frameCount
// ALOG_ASSERT(mBuffer.frameCount == 0);
}
}
// ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleMono16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
}
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
/*******************************************************************
*
* AsmMono16Loop
* asm optimized monotonic loop version; one loop is 2 frames
* Input:
* in : pointer on input samples
* maxOutPt : pointer on first not filled
* maxInIdx : index on first not used
* outputIndex : pointer on current output index
* out : pointer on output buffer
* inputIndex : pointer on current input index
* vl, vr : left and right gain
* phaseFraction : pointer on current phase fraction
* phaseIncrement
* Ouput:
* outputIndex :
* out : updated buffer
* inputIndex : index of next to use
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
__attribute__((noinline))
void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
asm(
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
// get parameters
" ldr r6, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
" ldr r6, [r6]\n" // phaseFraction
" ldr r7, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
" ldr r7, [r7]\n" // inputIndex
" ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out
" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
" ldr r0, [r0]\n" // outputIndex
" add r8, r8, r0, asl #2\n" // curOut
" ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement
" ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl
" ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr
// r0 pin, x0, Samp
// r1 in
// r2 maxOutPt
// r3 maxInIdx
// r4 x1, i1, i3, Out1
// r5 out0
// r6 frac
// r7 inputIndex
// r8 curOut
// r9 inc
// r10 vl
// r11 vr
// r12
// r13 sp
// r14
// the following loop works on 2 frames
"1:\n"
" cmp r8, r2\n" // curOut - maxCurOut
" bcs 2f\n"
#define MO_ONE_FRAME \
" add r0, r1, r7, asl #1\n" /* in + inputIndex */\
" ldrsh r4, [r0]\n" /* in[inputIndex] */\
" ldr r5, [r8]\n" /* out[outputIndex] */\
" ldrsh r0, [r0, #-2]\n" /* in[inputIndex-1] */\
" bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
" sub r4, r4, r0\n" /* in[inputIndex] - in[inputIndex-1] */\
" mov r4, r4, lsl #2\n" /* <<2 */\
" smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
" add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
" add r0, r0, r4\n" /* x0 - (..) */\
" mla r5, r0, r10, r5\n" /* vl*interp + out[] */\
" ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
" str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
" mla r4, r0, r11, r4\n" /* vr*interp + out[] */\
" add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */\
" str r4, [r8], #4\n" /* out[outputIndex++] = ... */
MO_ONE_FRAME // frame 1
MO_ONE_FRAME // frame 2
" cmp r7, r3\n" // inputIndex - maxInIdx
" bcc 1b\n"
"2:\n"
" bic r6, r6, #0xC0000000\n" // phaseFraction & ...
// save modified values
" ldr r0, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
" str r6, [r0]\n" // phaseFraction
" ldr r0, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
" str r7, [r0]\n" // inputIndex
" ldr r0, [sp, #" MO_PARAM5 " + 4]\n" // out
" sub r8, r0\n" // curOut - out
" asr r8, #2\n" // new outputIndex
" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
" str r8, [r0]\n" // save outputIndex
" ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
);
}
/*******************************************************************
*
* AsmStereo16Loop
* asm optimized stereo loop version; one loop is 2 frames
* Input:
* in : pointer on input samples
* maxOutPt : pointer on first not filled
* maxInIdx : index on first not used
* outputIndex : pointer on current output index
* out : pointer on output buffer
* inputIndex : pointer on current input index
* vl, vr : left and right gain
* phaseFraction : pointer on current phase fraction
* phaseIncrement
* Ouput:
* outputIndex :
* out : updated buffer
* inputIndex : index of next to use
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
__attribute__((noinline))
void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex)
asm(
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
// get parameters
" ldr r6, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
" ldr r6, [r6]\n" // phaseFraction
" ldr r7, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
" ldr r7, [r7]\n" // inputIndex
" ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out
" ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
" ldr r0, [r0]\n" // outputIndex
" add r8, r8, r0, asl #2\n" // curOut
" ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement
" ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl
" ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr
// r0 pin, x0, Samp
// r1 in
// r2 maxOutPt
// r3 maxInIdx
// r4 x1, i1, i3, out1
// r5 out0
// r6 frac
// r7 inputIndex
// r8 curOut
// r9 inc
// r10 vl
// r11 vr
// r12 temporary
// r13 sp
// r14
"3:\n"
" cmp r8, r2\n" // curOut - maxCurOut
" bcs 4f\n"
#define ST_ONE_FRAME \
" bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
\
" add r0, r1, r7, asl #2\n" /* in + 2*inputIndex */\
\
" ldrsh r4, [r0]\n" /* in[2*inputIndex] */\
" ldr r5, [r8]\n" /* out[outputIndex] */\
" ldrsh r12, [r0, #-4]\n" /* in[2*inputIndex-2] */\
" sub r4, r4, r12\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
" mov r4, r4, lsl #2\n" /* <<2 */\
" smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
" add r12, r12, r4\n" /* x0 - (..) */\
" mla r5, r12, r10, r5\n" /* vl*interp + out[] */\
" ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
" str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
\
" ldrsh r12, [r0, #+2]\n" /* in[2*inputIndex+1] */\
" ldrsh r0, [r0, #-2]\n" /* in[2*inputIndex-1] */\
" sub r12, r12, r0\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
" mov r12, r12, lsl #2\n" /* <<2 */\
" smulwt r12, r12, r6\n" /* (x1-x0)*.. */\
" add r12, r0, r12\n" /* x0 - (..) */\
" mla r4, r12, r11, r4\n" /* vr*interp + out[] */\
" str r4, [r8], #4\n" /* out[outputIndex++] = ... */\
\
" add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
" add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */
ST_ONE_FRAME // frame 1
ST_ONE_FRAME // frame 1
" cmp r7, r3\n" // inputIndex - maxInIdx
" bcc 3b\n"
"4:\n"
" bic r6, r6, #0xC0000000\n" // phaseFraction & ...
// save modified values
" ldr r0, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
" str r6, [r0]\n" // phaseFraction
" ldr r0, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
" str r7, [r0]\n" // inputIndex
" ldr r0, [sp, #" ST_PARAM5 " + 4]\n" // out
" sub r8, r0\n" // curOut - out
" asr r8, #2\n" // new outputIndex
" ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
" str r8, [r0]\n" // save outputIndex
" ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
);
}
#endif // ASM_ARM_RESAMP1
// ----------------------------------------------------------------------------
} // namespace android
|