AudioResampler.cpp revision 075abae2a954bf3edf18ad1705c2c0f188454ae0 
1/*
2 * Copyright (C) 2007 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17#define LOG_TAG "AudioResampler"
18//#define LOG_NDEBUG 0
19
20#include <stdint.h>
21#include <stdlib.h>
22#include <sys/types.h>
23#include <cutils/log.h>
24#include <cutils/properties.h>
25#include "AudioResampler.h"
26#include "AudioResamplerSinc.h"
27#include "AudioResamplerCubic.h"
28#include "AudioResamplerDyn.h"
29
30#ifdef __arm__
31#include <machine/cpu-features.h>
32#endif
33
34namespace android {
35
36#ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option
37    #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
38#endif // __ARM_HAVE_HALFWORD_MULTIPLY
39// ----------------------------------------------------------------------------
40
41class AudioResamplerOrder1 : public AudioResampler {
42public:
43    AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :
44        AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
45    }
46    virtual void resample(int32_t* out, size_t outFrameCount,
47            AudioBufferProvider* provider);
48private:
49    // number of bits used in interpolation multiply - 15 bits avoids overflow
50    static const int kNumInterpBits = 15;
51
52    // bits to shift the phase fraction down to avoid overflow
53    static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
54
55    void init() {}
56    void resampleMono16(int32_t* out, size_t outFrameCount,
57            AudioBufferProvider* provider);
58    void resampleStereo16(int32_t* out, size_t outFrameCount,
59            AudioBufferProvider* provider);
60#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
61    void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
62            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
63            uint32_t &phaseFraction, uint32_t phaseIncrement);
64    void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
65            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
66            uint32_t &phaseFraction, uint32_t phaseIncrement);
67#endif  // ASM_ARM_RESAMP1
68
69    static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
70        return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
71    }
72    static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) {
73        *frac += inc;
74        *index += (size_t)(*frac >> kNumPhaseBits);
75        *frac &= kPhaseMask;
76    }
77    int mX0L;
78    int mX0R;
79};
80
81/*static*/
82const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
83
84bool AudioResampler::qualityIsSupported(src_quality quality)
85{
86    switch (quality) {
87    case DEFAULT_QUALITY:
88    case LOW_QUALITY:
89    case MED_QUALITY:
90    case HIGH_QUALITY:
91    case VERY_HIGH_QUALITY:
92    case DYN_LOW_QUALITY:
93    case DYN_MED_QUALITY:
94    case DYN_HIGH_QUALITY:
95        return true;
96    default:
97        return false;
98    }
99}
100
101// ----------------------------------------------------------------------------
102
103static pthread_once_t once_control = PTHREAD_ONCE_INIT;
104static AudioResampler::src_quality defaultQuality = AudioResampler::DEFAULT_QUALITY;
105
106void AudioResampler::init_routine()
107{
108    char value[PROPERTY_VALUE_MAX];
109    if (property_get("af.resampler.quality", value, NULL) > 0) {
110        char *endptr;
111        unsigned long l = strtoul(value, &endptr, 0);
112        if (*endptr == '\0') {
113            defaultQuality = (src_quality) l;
114            ALOGD("forcing AudioResampler quality to %d", defaultQuality);
115            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
116                defaultQuality = DEFAULT_QUALITY;
117            }
118        }
119    }
120}
121
122uint32_t AudioResampler::qualityMHz(src_quality quality)
123{
124    switch (quality) {
125    default:
126    case DEFAULT_QUALITY:
127    case LOW_QUALITY:
128        return 3;
129    case MED_QUALITY:
130        return 6;
131    case HIGH_QUALITY:
132        return 20;
133    case VERY_HIGH_QUALITY:
134        return 34;
135    case DYN_LOW_QUALITY:
136        return 4;
137    case DYN_MED_QUALITY:
138        return 6;
139    case DYN_HIGH_QUALITY:
140        return 12;
141    }
142}
143
144static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, should be tunable
145static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
146static uint32_t currentMHz = 0;
147
148AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
149        int32_t sampleRate, src_quality quality) {
150
151    bool atFinalQuality;
152    if (quality == DEFAULT_QUALITY) {
153        // read the resampler default quality property the first time it is needed
154        int ok = pthread_once(&once_control, init_routine);
155        if (ok != 0) {
156            ALOGE("%s pthread_once failed: %d", __func__, ok);
157        }
158        quality = defaultQuality;
159        atFinalQuality = false;
160    } else {
161        atFinalQuality = true;
162    }
163
164    /* if the caller requests DEFAULT_QUALITY and af.resampler.property
165     * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
166     * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
167     * due to estimated CPU load of having too many active resamplers
168     * (the code below the if).
169     */
170    if (quality == DEFAULT_QUALITY) {
171        quality = DYN_MED_QUALITY;
172    }
173
174    // naive implementation of CPU load throttling doesn't account for whether resampler is active
175    pthread_mutex_lock(&mutex);
176    for (;;) {
177        uint32_t deltaMHz = qualityMHz(quality);
178        uint32_t newMHz = currentMHz + deltaMHz;
179        if ((qualityIsSupported(quality) && newMHz <= maxMHz) || atFinalQuality) {
180            ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
181                    currentMHz, newMHz, deltaMHz, quality);
182            currentMHz = newMHz;
183            break;
184        }
185        // not enough CPU available for proposed quality level, so try next lowest level
186        switch (quality) {
187        default:
188        case LOW_QUALITY:
189            atFinalQuality = true;
190            break;
191        case MED_QUALITY:
192            quality = LOW_QUALITY;
193            break;
194        case HIGH_QUALITY:
195            quality = MED_QUALITY;
196            break;
197        case VERY_HIGH_QUALITY:
198            quality = HIGH_QUALITY;
199            break;
200        case DYN_LOW_QUALITY:
201            atFinalQuality = true;
202            break;
203        case DYN_MED_QUALITY:
204            quality = DYN_LOW_QUALITY;
205            break;
206        case DYN_HIGH_QUALITY:
207            quality = DYN_MED_QUALITY;
208            break;
209        }
210    }
211    pthread_mutex_unlock(&mutex);
212
213    AudioResampler* resampler;
214
215    switch (quality) {
216    default:
217    case LOW_QUALITY:
218        ALOGV("Create linear Resampler");
219        resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
220        break;
221    case MED_QUALITY:
222        ALOGV("Create cubic Resampler");
223        resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
224        break;
225    case HIGH_QUALITY:
226        ALOGV("Create HIGH_QUALITY sinc Resampler");
227        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
228        break;
229    case VERY_HIGH_QUALITY:
230        ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
231        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
232        break;
233    case DYN_LOW_QUALITY:
234    case DYN_MED_QUALITY:
235    case DYN_HIGH_QUALITY:
236        ALOGV("Create dynamic Resampler = %d", quality);
237        if (bitDepth == 32) { /* bitDepth == 32 signals float precision */
238            resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,
239                    sampleRate, quality);
240        } else if (quality == DYN_HIGH_QUALITY) {
241            resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,
242                    sampleRate, quality);
243        } else {
244            resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,
245                    sampleRate, quality);
246        }
247        break;
248    }
249
250    // initialize resampler
251    resampler->init();
252    return resampler;
253}
254
255AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
256        int32_t sampleRate, src_quality quality) :
257    mBitDepth(bitDepth), mChannelCount(inChannelCount),
258            mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
259            mPhaseFraction(0), mLocalTimeFreq(0),
260            mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {
261    // sanity check on format
262    if ((bitDepth != 16 && (quality < DYN_LOW_QUALITY || bitDepth != 32))
263            || inChannelCount < 1
264            || inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) {
265        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d bits, %d channels",
266                quality, bitDepth, inChannelCount);
267    }
268    if (sampleRate <= 0) {
269        LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
270    }
271
272    // initialize common members
273    mVolume[0] = mVolume[1] = 0;
274    mBuffer.frameCount = 0;
275
276}
277
278AudioResampler::~AudioResampler() {
279    pthread_mutex_lock(&mutex);
280    src_quality quality = getQuality();
281    uint32_t deltaMHz = qualityMHz(quality);
282    int32_t newMHz = currentMHz - deltaMHz;
283    ALOGV("resampler load %u -> %d MHz due to delta -%u MHz from quality %d",
284            currentMHz, newMHz, deltaMHz, quality);
285    LOG_ALWAYS_FATAL_IF(newMHz < 0, "negative resampler load %d MHz", newMHz);
286    currentMHz = newMHz;
287    pthread_mutex_unlock(&mutex);
288}
289
290void AudioResampler::setSampleRate(int32_t inSampleRate) {
291    mInSampleRate = inSampleRate;
292    mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate);
293}
294
295void AudioResampler::setVolume(int16_t left, int16_t right) {
296    // TODO: Implement anti-zipper filter
297    mVolume[0] = left;
298    mVolume[1] = right;
299}
300
301void AudioResampler::setLocalTimeFreq(uint64_t freq) {
302    mLocalTimeFreq = freq;
303}
304
305void AudioResampler::setPTS(int64_t pts) {
306    mPTS = pts;
307}
308
309int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
310
311    if (mPTS == AudioBufferProvider::kInvalidPTS) {
312        return AudioBufferProvider::kInvalidPTS;
313    } else {
314        return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
315    }
316}
317
318void AudioResampler::reset() {
319    mInputIndex = 0;
320    mPhaseFraction = 0;
321    mBuffer.frameCount = 0;
322}
323
324// ----------------------------------------------------------------------------
325
326void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
327        AudioBufferProvider* provider) {
328
329    // should never happen, but we overflow if it does
330    // ALOG_ASSERT(outFrameCount < 32767);
331
332    // select the appropriate resampler
333    switch (mChannelCount) {
334    case 1:
335        resampleMono16(out, outFrameCount, provider);
336        break;
337    case 2:
338        resampleStereo16(out, outFrameCount, provider);
339        break;
340    }
341}
342
343void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
344        AudioBufferProvider* provider) {
345
346    int32_t vl = mVolume[0];
347    int32_t vr = mVolume[1];
348
349    size_t inputIndex = mInputIndex;
350    uint32_t phaseFraction = mPhaseFraction;
351    uint32_t phaseIncrement = mPhaseIncrement;
352    size_t outputIndex = 0;
353    size_t outputSampleCount = outFrameCount * 2;
354    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
355
356    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
357    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
358
359    while (outputIndex < outputSampleCount) {
360
361        // buffer is empty, fetch a new one
362        while (mBuffer.frameCount == 0) {
363            mBuffer.frameCount = inFrameCount;
364            provider->getNextBuffer(&mBuffer,
365                                    calculateOutputPTS(outputIndex / 2));
366            if (mBuffer.raw == NULL) {
367                goto resampleStereo16_exit;
368            }
369
370            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
371            if (mBuffer.frameCount > inputIndex) break;
372
373            inputIndex -= mBuffer.frameCount;
374            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
375            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
376            provider->releaseBuffer(&mBuffer);
377            // mBuffer.frameCount == 0 now so we reload a new buffer
378        }
379
380        int16_t *in = mBuffer.i16;
381
382        // handle boundary case
383        while (inputIndex == 0) {
384            // ALOGE("boundary case");
385            out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
386            out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
387            Advance(&inputIndex, &phaseFraction, phaseIncrement);
388            if (outputIndex == outputSampleCount) {
389                break;
390            }
391        }
392
393        // process input samples
394        // ALOGE("general case");
395
396#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
397        if (inputIndex + 2 < mBuffer.frameCount) {
398            int32_t* maxOutPt;
399            int32_t maxInIdx;
400
401            maxOutPt = out + (outputSampleCount - 2);   // 2 because 2 frames per loop
402            maxInIdx = mBuffer.frameCount - 2;
403            AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
404                    phaseFraction, phaseIncrement);
405        }
406#endif  // ASM_ARM_RESAMP1
407
408        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
409            out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
410                    in[inputIndex*2], phaseFraction);
411            out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
412                    in[inputIndex*2+1], phaseFraction);
413            Advance(&inputIndex, &phaseFraction, phaseIncrement);
414        }
415
416        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
417
418        // if done with buffer, save samples
419        if (inputIndex >= mBuffer.frameCount) {
420            inputIndex -= mBuffer.frameCount;
421
422            // ALOGE("buffer done, new input index %d", inputIndex);
423
424            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
425            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
426            provider->releaseBuffer(&mBuffer);
427
428            // verify that the releaseBuffer resets the buffer frameCount
429            // ALOG_ASSERT(mBuffer.frameCount == 0);
430        }
431    }
432
433    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
434
435resampleStereo16_exit:
436    // save state
437    mInputIndex = inputIndex;
438    mPhaseFraction = phaseFraction;
439}
440
441void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
442        AudioBufferProvider* provider) {
443
444    int32_t vl = mVolume[0];
445    int32_t vr = mVolume[1];
446
447    size_t inputIndex = mInputIndex;
448    uint32_t phaseFraction = mPhaseFraction;
449    uint32_t phaseIncrement = mPhaseIncrement;
450    size_t outputIndex = 0;
451    size_t outputSampleCount = outFrameCount * 2;
452    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
453
454    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
455    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
456    while (outputIndex < outputSampleCount) {
457        // buffer is empty, fetch a new one
458        while (mBuffer.frameCount == 0) {
459            mBuffer.frameCount = inFrameCount;
460            provider->getNextBuffer(&mBuffer,
461                                    calculateOutputPTS(outputIndex / 2));
462            if (mBuffer.raw == NULL) {
463                mInputIndex = inputIndex;
464                mPhaseFraction = phaseFraction;
465                goto resampleMono16_exit;
466            }
467            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
468            if (mBuffer.frameCount >  inputIndex) break;
469
470            inputIndex -= mBuffer.frameCount;
471            mX0L = mBuffer.i16[mBuffer.frameCount-1];
472            provider->releaseBuffer(&mBuffer);
473            // mBuffer.frameCount == 0 now so we reload a new buffer
474        }
475        int16_t *in = mBuffer.i16;
476
477        // handle boundary case
478        while (inputIndex == 0) {
479            // ALOGE("boundary case");
480            int32_t sample = Interp(mX0L, in[0], phaseFraction);
481            out[outputIndex++] += vl * sample;
482            out[outputIndex++] += vr * sample;
483            Advance(&inputIndex, &phaseFraction, phaseIncrement);
484            if (outputIndex == outputSampleCount) {
485                break;
486            }
487        }
488
489        // process input samples
490        // ALOGE("general case");
491
492#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
493        if (inputIndex + 2 < mBuffer.frameCount) {
494            int32_t* maxOutPt;
495            int32_t maxInIdx;
496
497            maxOutPt = out + (outputSampleCount - 2);
498            maxInIdx = (int32_t)mBuffer.frameCount - 2;
499                AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
500                        phaseFraction, phaseIncrement);
501        }
502#endif  // ASM_ARM_RESAMP1
503
504        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
505            int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
506                    phaseFraction);
507            out[outputIndex++] += vl * sample;
508            out[outputIndex++] += vr * sample;
509            Advance(&inputIndex, &phaseFraction, phaseIncrement);
510        }
511
512
513        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
514
515        // if done with buffer, save samples
516        if (inputIndex >= mBuffer.frameCount) {
517            inputIndex -= mBuffer.frameCount;
518
519            // ALOGE("buffer done, new input index %d", inputIndex);
520
521            mX0L = mBuffer.i16[mBuffer.frameCount-1];
522            provider->releaseBuffer(&mBuffer);
523
524            // verify that the releaseBuffer resets the buffer frameCount
525            // ALOG_ASSERT(mBuffer.frameCount == 0);
526        }
527    }
528
529    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
530
531resampleMono16_exit:
532    // save state
533    mInputIndex = inputIndex;
534    mPhaseFraction = phaseFraction;
535}
536
537#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
538
539/*******************************************************************
540*
541*   AsmMono16Loop
542*   asm optimized monotonic loop version; one loop is 2 frames
543*   Input:
544*       in : pointer on input samples
545*       maxOutPt : pointer on first not filled
546*       maxInIdx : index on first not used
547*       outputIndex : pointer on current output index
548*       out : pointer on output buffer
549*       inputIndex : pointer on current input index
550*       vl, vr : left and right gain
551*       phaseFraction : pointer on current phase fraction
552*       phaseIncrement
553*   Ouput:
554*       outputIndex :
555*       out : updated buffer
556*       inputIndex : index of next to use
557*       phaseFraction : phase fraction for next interpolation
558*
559*******************************************************************/
560__attribute__((noinline))
561void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
562            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
563            uint32_t &phaseFraction, uint32_t phaseIncrement)
564{
565    (void)maxOutPt; // remove unused parameter warnings
566    (void)maxInIdx;
567    (void)outputIndex;
568    (void)out;
569    (void)inputIndex;
570    (void)vl;
571    (void)vr;
572    (void)phaseFraction;
573    (void)phaseIncrement;
574    (void)in;
575#define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
576
577    asm(
578        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
579        // get parameters
580        "   ldr r6, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
581        "   ldr r6, [r6]\n"                         // phaseFraction
582        "   ldr r7, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
583        "   ldr r7, [r7]\n"                         // inputIndex
584        "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
585        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
586        "   ldr r0, [r0]\n"                         // outputIndex
587        "   add r8, r8, r0, asl #2\n"               // curOut
588        "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
589        "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
590        "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
591
592        // r0 pin, x0, Samp
593
594        // r1 in
595        // r2 maxOutPt
596        // r3 maxInIdx
597
598        // r4 x1, i1, i3, Out1
599        // r5 out0
600
601        // r6 frac
602        // r7 inputIndex
603        // r8 curOut
604
605        // r9 inc
606        // r10 vl
607        // r11 vr
608
609        // r12
610        // r13 sp
611        // r14
612
613        // the following loop works on 2 frames
614
615        "1:\n"
616        "   cmp r8, r2\n"                   // curOut - maxCurOut
617        "   bcs 2f\n"
618
619#define MO_ONE_FRAME \
620    "   add r0, r1, r7, asl #1\n"       /* in + inputIndex */\
621    "   ldrsh r4, [r0]\n"               /* in[inputIndex] */\
622    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
623    "   ldrsh r0, [r0, #-2]\n"          /* in[inputIndex-1] */\
624    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
625    "   sub r4, r4, r0\n"               /* in[inputIndex] - in[inputIndex-1] */\
626    "   mov r4, r4, lsl #2\n"           /* <<2 */\
627    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
628    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
629    "   add r0, r0, r4\n"               /* x0 - (..) */\
630    "   mla r5, r0, r10, r5\n"          /* vl*interp + out[] */\
631    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
632    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
633    "   mla r4, r0, r11, r4\n"          /* vr*interp + out[] */\
634    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */\
635    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */
636
637        MO_ONE_FRAME    // frame 1
638        MO_ONE_FRAME    // frame 2
639
640        "   cmp r7, r3\n"                   // inputIndex - maxInIdx
641        "   bcc 1b\n"
642        "2:\n"
643
644        "   bic r6, r6, #0xC0000000\n"             // phaseFraction & ...
645        // save modified values
646        "   ldr r0, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
647        "   str r6, [r0]\n"                         // phaseFraction
648        "   ldr r0, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
649        "   str r7, [r0]\n"                         // inputIndex
650        "   ldr r0, [sp, #" MO_PARAM5 " + 4]\n"     // out
651        "   sub r8, r0\n"                           // curOut - out
652        "   asr r8, #2\n"                           // new outputIndex
653        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
654        "   str r8, [r0]\n"                         // save outputIndex
655
656        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
657    );
658}
659
660/*******************************************************************
661*
662*   AsmStereo16Loop
663*   asm optimized stereo loop version; one loop is 2 frames
664*   Input:
665*       in : pointer on input samples
666*       maxOutPt : pointer on first not filled
667*       maxInIdx : index on first not used
668*       outputIndex : pointer on current output index
669*       out : pointer on output buffer
670*       inputIndex : pointer on current input index
671*       vl, vr : left and right gain
672*       phaseFraction : pointer on current phase fraction
673*       phaseIncrement
674*   Ouput:
675*       outputIndex :
676*       out : updated buffer
677*       inputIndex : index of next to use
678*       phaseFraction : phase fraction for next interpolation
679*
680*******************************************************************/
681__attribute__((noinline))
682void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
683            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
684            uint32_t &phaseFraction, uint32_t phaseIncrement)
685{
686    (void)maxOutPt; // remove unused parameter warnings
687    (void)maxInIdx;
688    (void)outputIndex;
689    (void)out;
690    (void)inputIndex;
691    (void)vl;
692    (void)vr;
693    (void)phaseFraction;
694    (void)phaseIncrement;
695    (void)in;
696#define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
697    asm(
698        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
699        // get parameters
700        "   ldr r6, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
701        "   ldr r6, [r6]\n"                         // phaseFraction
702        "   ldr r7, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
703        "   ldr r7, [r7]\n"                         // inputIndex
704        "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
705        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
706        "   ldr r0, [r0]\n"                         // outputIndex
707        "   add r8, r8, r0, asl #2\n"               // curOut
708        "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
709        "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
710        "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr
711
712        // r0 pin, x0, Samp
713
714        // r1 in
715        // r2 maxOutPt
716        // r3 maxInIdx
717
718        // r4 x1, i1, i3, out1
719        // r5 out0
720
721        // r6 frac
722        // r7 inputIndex
723        // r8 curOut
724
725        // r9 inc
726        // r10 vl
727        // r11 vr
728
729        // r12 temporary
730        // r13 sp
731        // r14
732
733        "3:\n"
734        "   cmp r8, r2\n"                   // curOut - maxCurOut
735        "   bcs 4f\n"
736
737#define ST_ONE_FRAME \
738    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
739\
740    "   add r0, r1, r7, asl #2\n"       /* in + 2*inputIndex */\
741\
742    "   ldrsh r4, [r0]\n"               /* in[2*inputIndex] */\
743    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
744    "   ldrsh r12, [r0, #-4]\n"         /* in[2*inputIndex-2] */\
745    "   sub r4, r4, r12\n"              /* in[2*InputIndex] - in[2*InputIndex-2] */\
746    "   mov r4, r4, lsl #2\n"           /* <<2 */\
747    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
748    "   add r12, r12, r4\n"             /* x0 - (..) */\
749    "   mla r5, r12, r10, r5\n"         /* vl*interp + out[] */\
750    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
751    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
752\
753    "   ldrsh r12, [r0, #+2]\n"         /* in[2*inputIndex+1] */\
754    "   ldrsh r0, [r0, #-2]\n"          /* in[2*inputIndex-1] */\
755    "   sub r12, r12, r0\n"             /* in[2*InputIndex] - in[2*InputIndex-2] */\
756    "   mov r12, r12, lsl #2\n"         /* <<2 */\
757    "   smulwt r12, r12, r6\n"          /* (x1-x0)*.. */\
758    "   add r12, r0, r12\n"             /* x0 - (..) */\
759    "   mla r4, r12, r11, r4\n"         /* vr*interp + out[] */\
760    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */\
761\
762    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
763    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */
764
765    ST_ONE_FRAME    // frame 1
766    ST_ONE_FRAME    // frame 1
767
768        "   cmp r7, r3\n"                       // inputIndex - maxInIdx
769        "   bcc 3b\n"
770        "4:\n"
771
772        "   bic r6, r6, #0xC0000000\n"              // phaseFraction & ...
773        // save modified values
774        "   ldr r0, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
775        "   str r6, [r0]\n"                         // phaseFraction
776        "   ldr r0, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
777        "   str r7, [r0]\n"                         // inputIndex
778        "   ldr r0, [sp, #" ST_PARAM5 " + 4]\n"     // out
779        "   sub r8, r0\n"                           // curOut - out
780        "   asr r8, #2\n"                           // new outputIndex
781        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
782        "   str r8, [r0]\n"                         // save outputIndex
783
784        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
785    );
786}
787
788#endif  // ASM_ARM_RESAMP1
789
790
791// ----------------------------------------------------------------------------
792
793} // namespace android
794