1/*
2 * Copyright (C) 2013 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 "AudioResamplerDyn"
18//#define LOG_NDEBUG 0
19
20#include <malloc.h>
21#include <string.h>
22#include <stdlib.h>
23#include <dlfcn.h>
24#include <math.h>
25
26#include <cutils/compiler.h>
27#include <cutils/properties.h>
28#include <utils/Debug.h>
29#include <utils/Log.h>
30#include <audio_utils/primitives.h>
31
32#include "AudioResamplerFirOps.h" // USE_NEON and USE_INLINE_ASSEMBLY defined here
33#include "AudioResamplerFirProcess.h"
34#include "AudioResamplerFirProcessNeon.h"
35#include "AudioResamplerFirGen.h" // requires math.h
36#include "AudioResamplerDyn.h"
37
38//#define DEBUG_RESAMPLER
39
40namespace android {
41
42/*
43 * InBuffer is a type agnostic input buffer.
44 *
45 * Layout of the state buffer for halfNumCoefs=8.
46 *
47 * [rrrrrrppppppppnnnnnnnnrrrrrrrrrrrrrrrrrrr.... rrrrrrr]
48 *  S            I                                R
49 *
50 * S = mState
51 * I = mImpulse
52 * R = mRingFull
53 * p = past samples, convoluted with the (p)ositive side of sinc()
54 * n = future samples, convoluted with the (n)egative side of sinc()
55 * r = extra space for implementing the ring buffer
56 */
57
58template<typename TC, typename TI, typename TO>
59AudioResamplerDyn<TC, TI, TO>::InBuffer::InBuffer()
60    : mState(NULL), mImpulse(NULL), mRingFull(NULL), mStateCount(0)
61{
62}
63
64template<typename TC, typename TI, typename TO>
65AudioResamplerDyn<TC, TI, TO>::InBuffer::~InBuffer()
66{
67    init();
68}
69
70template<typename TC, typename TI, typename TO>
71void AudioResamplerDyn<TC, TI, TO>::InBuffer::init()
72{
73    free(mState);
74    mState = NULL;
75    mImpulse = NULL;
76    mRingFull = NULL;
77    mStateCount = 0;
78}
79
80// resizes the state buffer to accommodate the appropriate filter length
81template<typename TC, typename TI, typename TO>
82void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs)
83{
84    // calculate desired state size
85    size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;
86
87    // check if buffer needs resizing
88    if (mState
89            && stateCount == mStateCount
90            && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) {
91        return;
92    }
93
94    // create new buffer
95    TI* state = NULL;
96    (void)posix_memalign(reinterpret_cast<void**>(&state), 32, stateCount*sizeof(*state));
97    memset(state, 0, stateCount*sizeof(*state));
98
99    // attempt to preserve state
100    if (mState) {
101        TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
102        TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
103        TI* dst = state;
104
105        if (srcLo < mState) {
106            dst += mState-srcLo;
107            srcLo = mState;
108        }
109        if (srcHi > mState + mStateCount) {
110            srcHi = mState + mStateCount;
111        }
112        memcpy(dst, srcLo, (srcHi - srcLo) * sizeof(*srcLo));
113        free(mState);
114    }
115
116    // set class member vars
117    mState = state;
118    mStateCount = stateCount;
119    mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed
120    mRingFull = state + mStateCount - halfNumCoefs*CHANNELS;
121}
122
123// copy in the input data into the head (impulse+halfNumCoefs) of the buffer.
124template<typename TC, typename TI, typename TO>
125template<int CHANNELS>
126void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAgain(TI*& impulse, const int halfNumCoefs,
127        const TI* const in, const size_t inputIndex)
128{
129    TI* head = impulse + halfNumCoefs*CHANNELS;
130    for (size_t i=0 ; i<CHANNELS ; i++) {
131        head[i] = in[inputIndex*CHANNELS + i];
132    }
133}
134
135// advance the impulse pointer, and load in data into the head (impulse+halfNumCoefs)
136template<typename TC, typename TI, typename TO>
137template<int CHANNELS>
138void AudioResamplerDyn<TC, TI, TO>::InBuffer::readAdvance(TI*& impulse, const int halfNumCoefs,
139        const TI* const in, const size_t inputIndex)
140{
141    impulse += CHANNELS;
142
143    if (CC_UNLIKELY(impulse >= mRingFull)) {
144        const size_t shiftDown = mRingFull - mState - halfNumCoefs*CHANNELS;
145        memcpy(mState, mState+shiftDown, halfNumCoefs*CHANNELS*2*sizeof(TI));
146        impulse -= shiftDown;
147    }
148    readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
149}
150
151template<typename TC, typename TI, typename TO>
152void AudioResamplerDyn<TC, TI, TO>::Constants::set(
153        int L, int halfNumCoefs, int inSampleRate, int outSampleRate)
154{
155    int bits = 0;
156    int lscale = inSampleRate/outSampleRate < 2 ? L - 1 :
157            static_cast<int>(static_cast<uint64_t>(L)*inSampleRate/outSampleRate);
158    for (int i=lscale; i; ++bits, i>>=1)
159        ;
160    mL = L;
161    mShift = kNumPhaseBits - bits;
162    mHalfNumCoefs = halfNumCoefs;
163}
164
165template<typename TC, typename TI, typename TO>
166AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(
167        int inChannelCount, int32_t sampleRate, src_quality quality)
168    : AudioResampler(inChannelCount, sampleRate, quality),
169      mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
170    mCoefBuffer(NULL)
171{
172    mVolumeSimd[0] = mVolumeSimd[1] = 0;
173    // The AudioResampler base class assumes we are always ready for 1:1 resampling.
174    // We reset mInSampleRate to 0, so setSampleRate() will calculate filters for
175    // setSampleRate() for 1:1. (May be removed if precalculated filters are used.)
176    mInSampleRate = 0;
177    mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
178}
179
180template<typename TC, typename TI, typename TO>
181AudioResamplerDyn<TC, TI, TO>::~AudioResamplerDyn()
182{
183    free(mCoefBuffer);
184}
185
186template<typename TC, typename TI, typename TO>
187void AudioResamplerDyn<TC, TI, TO>::init()
188{
189    mFilterSampleRate = 0; // always trigger new filter generation
190    mInBuffer.init();
191}
192
193template<typename TC, typename TI, typename TO>
194void AudioResamplerDyn<TC, TI, TO>::setVolume(float left, float right)
195{
196    AudioResampler::setVolume(left, right);
197    if (is_same<TO, float>::value || is_same<TO, double>::value) {
198        mVolumeSimd[0] = static_cast<TO>(left);
199        mVolumeSimd[1] = static_cast<TO>(right);
200    } else {  // integer requires scaling to U4_28 (rounding down)
201        // integer volumes are clamped to 0 to UNITY_GAIN so there
202        // are no issues with signed overflow.
203        mVolumeSimd[0] = u4_28_from_float(clampFloatVol(left));
204        mVolumeSimd[1] = u4_28_from_float(clampFloatVol(right));
205    }
206}
207
208template<typename T> T max(T a, T b) {return a > b ? a : b;}
209
210template<typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;}
211
212template<typename TC, typename TI, typename TO>
213void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c,
214        double stopBandAtten, int inSampleRate, int outSampleRate, double tbwCheat)
215{
216    TC* buf = NULL;
217    static const double atten = 0.9998;   // to avoid ripple overflow
218    double fcr;
219    double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
220
221    (void)posix_memalign(reinterpret_cast<void**>(&buf), 32, (c.mL+1)*c.mHalfNumCoefs*sizeof(TC));
222    if (inSampleRate < outSampleRate) { // upsample
223        fcr = max(0.5*tbwCheat - tbw/2, tbw/2);
224    } else { // downsample
225        fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2);
226    }
227    // create and set filter
228    firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten);
229    c.mFirCoefs = buf;
230    if (mCoefBuffer) {
231        free(mCoefBuffer);
232    }
233    mCoefBuffer = buf;
234#ifdef DEBUG_RESAMPLER
235    // print basic filter stats
236    printf("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",
237            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw);
238    // test the filter and report results
239    double fp = (fcr - tbw/2)/c.mL;
240    double fs = (fcr + tbw/2)/c.mL;
241    double passMin, passMax, passRipple;
242    double stopMax, stopRipple;
243    testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
244            passMin, passMax, passRipple, stopMax, stopRipple);
245    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
246    printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
247#endif
248}
249
250// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop.
251static int gcd(int n, int m)
252{
253    if (m == 0) {
254        return n;
255    }
256    return gcd(m, n % m);
257}
258
259static bool isClose(int32_t newSampleRate, int32_t prevSampleRate,
260        int32_t filterSampleRate, int32_t outSampleRate)
261{
262
263    // different upsampling ratios do not need a filter change.
264    if (filterSampleRate != 0
265            && filterSampleRate < outSampleRate
266            && newSampleRate < outSampleRate)
267        return true;
268
269    // check design criteria again if downsampling is detected.
270    int pdiff = absdiff(newSampleRate, prevSampleRate);
271    int adiff = absdiff(newSampleRate, filterSampleRate);
272
273    // allow up to 6% relative change increments.
274    // allow up to 12% absolute change increments (from filter design)
275    return pdiff < prevSampleRate>>4 && adiff < filterSampleRate>>3;
276}
277
278template<typename TC, typename TI, typename TO>
279void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate)
280{
281    if (mInSampleRate == inSampleRate) {
282        return;
283    }
284    int32_t oldSampleRate = mInSampleRate;
285    uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift;
286    bool useS32 = false;
287
288    mInSampleRate = inSampleRate;
289
290    // TODO: Add precalculated Equiripple filters
291
292    if (mFilterQuality != getQuality() ||
293            !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) {
294        mFilterSampleRate = inSampleRate;
295        mFilterQuality = getQuality();
296
297        // Begin Kaiser Filter computation
298        //
299        // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB.
300        // Keep the stop band attenuation no greater than 84-85dB for 32 length S16 filters
301        //
302        // For s32 we keep the stop band attenuation at the same as 16b resolution, about
303        // 96-98dB
304        //
305
306        double stopBandAtten;
307        double tbwCheat = 1.; // how much we "cheat" into aliasing
308        int halfLength;
309        if (mFilterQuality == DYN_HIGH_QUALITY) {
310            // 32b coefficients, 64 length
311            useS32 = true;
312            stopBandAtten = 98.;
313            if (inSampleRate >= mSampleRate * 4) {
314                halfLength = 48;
315            } else if (inSampleRate >= mSampleRate * 2) {
316                halfLength = 40;
317            } else {
318                halfLength = 32;
319            }
320        } else if (mFilterQuality == DYN_LOW_QUALITY) {
321            // 16b coefficients, 16-32 length
322            useS32 = false;
323            stopBandAtten = 80.;
324            if (inSampleRate >= mSampleRate * 4) {
325                halfLength = 24;
326            } else if (inSampleRate >= mSampleRate * 2) {
327                halfLength = 16;
328            } else {
329                halfLength = 8;
330            }
331            if (inSampleRate <= mSampleRate) {
332                tbwCheat = 1.05;
333            } else {
334                tbwCheat = 1.03;
335            }
336        } else { // DYN_MED_QUALITY
337            // 16b coefficients, 32-64 length
338            // note: > 64 length filters with 16b coefs can have quantization noise problems
339            useS32 = false;
340            stopBandAtten = 84.;
341            if (inSampleRate >= mSampleRate * 4) {
342                halfLength = 32;
343            } else if (inSampleRate >= mSampleRate * 2) {
344                halfLength = 24;
345            } else {
346                halfLength = 16;
347            }
348            if (inSampleRate <= mSampleRate) {
349                tbwCheat = 1.03;
350            } else {
351                tbwCheat = 1.01;
352            }
353        }
354
355        // determine the number of polyphases in the filterbank.
356        // for 16b, it is desirable to have 2^(16/2) = 256 phases.
357        // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html
358        //
359        // We are a bit more lax on this.
360
361        int phases = mSampleRate / gcd(mSampleRate, inSampleRate);
362
363        // TODO: Once dynamic sample rate change is an option, the code below
364        // should be modified to execute only when dynamic sample rate change is enabled.
365        //
366        // as above, #phases less than 63 is too few phases for accurate linear interpolation.
367        // we increase the phases to compensate, but more phases means more memory per
368        // filter and more time to compute the filter.
369        //
370        // if we know that the filter will be used for dynamic sample rate changes,
371        // that would allow us skip this part for fixed sample rate resamplers.
372        //
373        while (phases<63) {
374            phases *= 2; // this code only needed to support dynamic rate changes
375        }
376
377        if (phases>=256) {  // too many phases, always interpolate
378            phases = 127;
379        }
380
381        // create the filter
382        mConstants.set(phases, halfLength, inSampleRate, mSampleRate);
383        createKaiserFir(mConstants, stopBandAtten,
384                inSampleRate, mSampleRate, tbwCheat);
385    } // End Kaiser filter
386
387    // update phase and state based on the new filter.
388    const Constants& c(mConstants);
389    mInBuffer.resize(mChannelCount, c.mHalfNumCoefs);
390    const uint32_t phaseWrapLimit = c.mL << c.mShift;
391    // try to preserve as much of the phase fraction as possible for on-the-fly changes
392    mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)
393            * phaseWrapLimit / oldPhaseWrapLimit;
394    mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.
395    mPhaseIncrement = static_cast<uint32_t>(static_cast<uint64_t>(phaseWrapLimit)
396            * inSampleRate / mSampleRate);
397
398    // determine which resampler to use
399    // check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
400    int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
401    if (locked) {
402        mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
403    }
404
405    // stride is the minimum number of filter coefficients processed per loop iteration.
406    // We currently only allow a stride of 16 to match with SIMD processing.
407    // This means that the filter length must be a multiple of 16,
408    // or half the filter length (mHalfNumCoefs) must be a multiple of 8.
409    //
410    // Note: A stride of 2 is achieved with non-SIMD processing.
411    int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2;
412    LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more");
413    LOG_ALWAYS_FATAL_IF(mChannelCount < 1 || mChannelCount > 8,
414            "Resampler channels(%d) must be between 1 to 8", mChannelCount);
415    // stride 16 (falls back to stride 2 for machines that do not support NEON)
416    if (locked) {
417        switch (mChannelCount) {
418        case 1:
419            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
420            break;
421        case 2:
422            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
423            break;
424        case 3:
425            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>;
426            break;
427        case 4:
428            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>;
429            break;
430        case 5:
431            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>;
432            break;
433        case 6:
434            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>;
435            break;
436        case 7:
437            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>;
438            break;
439        case 8:
440            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>;
441            break;
442        }
443    } else {
444        switch (mChannelCount) {
445        case 1:
446            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
447            break;
448        case 2:
449            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
450            break;
451        case 3:
452            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>;
453            break;
454        case 4:
455            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>;
456            break;
457        case 5:
458            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>;
459            break;
460        case 6:
461            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>;
462            break;
463        case 7:
464            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>;
465            break;
466        case 8:
467            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>;
468            break;
469        }
470    }
471#ifdef DEBUG_RESAMPLER
472    printf("channels:%d  %s  stride:%d  %s  coef:%d  shift:%d\n",
473            mChannelCount, locked ? "locked" : "interpolated",
474            stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift);
475#endif
476}
477
478template<typename TC, typename TI, typename TO>
479size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
480            AudioBufferProvider* provider)
481{
482    return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
483}
484
485template<typename TC, typename TI, typename TO>
486template<int CHANNELS, bool LOCKED, int STRIDE>
487size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
488        AudioBufferProvider* provider)
489{
490    // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
491    const int OUTPUT_CHANNELS = (CHANNELS < 2) ? 2 : CHANNELS;
492    const Constants& c(mConstants);
493    const TC* const coefs = mConstants.mFirCoefs;
494    TI* impulse = mInBuffer.getImpulse();
495    size_t inputIndex = 0;
496    uint32_t phaseFraction = mPhaseFraction;
497    const uint32_t phaseIncrement = mPhaseIncrement;
498    size_t outputIndex = 0;
499    size_t outputSampleCount = outFrameCount * OUTPUT_CHANNELS;
500    const uint32_t phaseWrapLimit = c.mL << c.mShift;
501    size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction)
502            / phaseWrapLimit;
503    // sanity check that inFrameCount is in signed 32 bit integer range.
504    ALOG_ASSERT(0 <= inFrameCount && inFrameCount < (1U << 31));
505
506    //ALOGV("inFrameCount:%d  outFrameCount:%d"
507    //        "  phaseIncrement:%u  phaseFraction:%u  phaseWrapLimit:%u",
508    //        inFrameCount, outFrameCount, phaseIncrement, phaseFraction, phaseWrapLimit);
509
510    // NOTE: be very careful when modifying the code here. register
511    // pressure is very high and a small change might cause the compiler
512    // to generate far less efficient code.
513    // Always sanity check the result with objdump or test-resample.
514
515    // the following logic is a bit convoluted to keep the main processing loop
516    // as tight as possible with register allocation.
517    while (outputIndex < outputSampleCount) {
518        //ALOGV("LOOP: inFrameCount:%d  outputIndex:%d  outFrameCount:%d"
519        //        "  phaseFraction:%u  phaseWrapLimit:%u",
520        //        inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
521
522        // check inputIndex overflow
523        ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d",
524                inputIndex, mBuffer.frameCount);
525        // Buffer is empty, fetch a new one if necessary (inFrameCount > 0).
526        // We may not fetch a new buffer if the existing data is sufficient.
527        while (mBuffer.frameCount == 0 && inFrameCount > 0) {
528            mBuffer.frameCount = inFrameCount;
529            provider->getNextBuffer(&mBuffer);
530            if (mBuffer.raw == NULL) {
531                goto resample_exit;
532            }
533            inFrameCount -= mBuffer.frameCount;
534            if (phaseFraction >= phaseWrapLimit) { // read in data
535                mInBuffer.template readAdvance<CHANNELS>(
536                        impulse, c.mHalfNumCoefs,
537                        reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
538                inputIndex++;
539                phaseFraction -= phaseWrapLimit;
540                while (phaseFraction >= phaseWrapLimit) {
541                    if (inputIndex >= mBuffer.frameCount) {
542                        inputIndex = 0;
543                        provider->releaseBuffer(&mBuffer);
544                        break;
545                    }
546                    mInBuffer.template readAdvance<CHANNELS>(
547                            impulse, c.mHalfNumCoefs,
548                            reinterpret_cast<TI*>(mBuffer.raw), inputIndex);
549                    inputIndex++;
550                    phaseFraction -= phaseWrapLimit;
551                }
552            }
553        }
554        const TI* const in = reinterpret_cast<const TI*>(mBuffer.raw);
555        const size_t frameCount = mBuffer.frameCount;
556        const int coefShift = c.mShift;
557        const int halfNumCoefs = c.mHalfNumCoefs;
558        const TO* const volumeSimd = mVolumeSimd;
559
560        // main processing loop
561        while (CC_LIKELY(outputIndex < outputSampleCount)) {
562            // caution: fir() is inlined and may be large.
563            // output will be loaded with the appropriate values
564            //
565            // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs]
566            // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs.
567            //
568            //ALOGV("LOOP2: inFrameCount:%d  outputIndex:%d  outFrameCount:%d"
569            //        "  phaseFraction:%u  phaseWrapLimit:%u",
570            //        inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit);
571            ALOG_ASSERT(phaseFraction < phaseWrapLimit);
572            fir<CHANNELS, LOCKED, STRIDE>(
573                    &out[outputIndex],
574                    phaseFraction, phaseWrapLimit,
575                    coefShift, halfNumCoefs, coefs,
576                    impulse, volumeSimd);
577
578            outputIndex += OUTPUT_CHANNELS;
579
580            phaseFraction += phaseIncrement;
581            while (phaseFraction >= phaseWrapLimit) {
582                if (inputIndex >= frameCount) {
583                    goto done;  // need a new buffer
584                }
585                mInBuffer.template readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
586                inputIndex++;
587                phaseFraction -= phaseWrapLimit;
588            }
589        }
590done:
591        // We arrive here when we're finished or when the input buffer runs out.
592        // Regardless we need to release the input buffer if we've acquired it.
593        if (inputIndex > 0) {  // we've acquired a buffer (alternatively could check frameCount)
594            ALOG_ASSERT(inputIndex == frameCount, "inputIndex(%d) != frameCount(%d)",
595                    inputIndex, frameCount);  // must have been fully read.
596            inputIndex = 0;
597            provider->releaseBuffer(&mBuffer);
598            ALOG_ASSERT(mBuffer.frameCount == 0);
599        }
600    }
601
602resample_exit:
603    // inputIndex must be zero in all three cases:
604    // (1) the buffer never was been acquired; (2) the buffer was
605    // released at "done:"; or (3) getNextBuffer() failed.
606    ALOG_ASSERT(inputIndex == 0, "Releasing: inputindex:%d frameCount:%d  phaseFraction:%u",
607            inputIndex, mBuffer.frameCount, phaseFraction);
608    ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
609    mInBuffer.setImpulse(impulse);
610    mPhaseFraction = phaseFraction;
611    return outputIndex / OUTPUT_CHANNELS;
612}
613
614/* instantiate templates used by AudioResampler::create */
615template class AudioResamplerDyn<float, float, float>;
616template class AudioResamplerDyn<int16_t, int16_t, int32_t>;
617template class AudioResamplerDyn<int32_t, int16_t, int32_t>;
618
619// ----------------------------------------------------------------------------
620} // namespace android
621