AudioResamplerDyn.cpp revision 24781fff62a4cf7279d3dac83c33e2ac612712ba 
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/Log.h>
29
30#include "AudioResamplerFirOps.h" // USE_NEON and USE_INLINE_ASSEMBLY defined here
31#include "AudioResamplerFirProcess.h"
32#include "AudioResamplerFirProcessNeon.h"
33#include "AudioResamplerFirGen.h" // requires math.h
34#include "AudioResamplerDyn.h"
35
36//#define DEBUG_RESAMPLER
37
38namespace android {
39
40// generate a unique resample type compile-time constant (constexpr)
41#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE, COEFTYPE) \
42    ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 | (COEFTYPE)<<2 \
43    | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<3)
44
45/*
46 * InBuffer is a type agnostic input buffer.
47 *
48 * Layout of the state buffer for halfNumCoefs=8.
49 *
50 * [rrrrrrppppppppnnnnnnnnrrrrrrrrrrrrrrrrrrr.... rrrrrrr]
51 *  S            I                                R
52 *
53 * S = mState
54 * I = mImpulse
55 * R = mRingFull
56 * p = past samples, convoluted with the (p)ositive side of sinc()
57 * n = future samples, convoluted with the (n)egative side of sinc()
58 * r = extra space for implementing the ring buffer
59 */
60
61template<typename TI>
62AudioResamplerDyn::InBuffer<TI>::InBuffer()
63    : mState(NULL), mImpulse(NULL), mRingFull(NULL), mStateSize(0) {
64}
65
66template<typename TI>
67AudioResamplerDyn::InBuffer<TI>::~InBuffer() {
68    init();
69}
70
71template<typename TI>
72void AudioResamplerDyn::InBuffer<TI>::init() {
73    free(mState);
74    mState = NULL;
75    mImpulse = NULL;
76    mRingFull = NULL;
77    mStateSize = 0;
78}
79
80// resizes the state buffer to accommodate the appropriate filter length
81template<typename TI>
82void AudioResamplerDyn::InBuffer<TI>::resize(int CHANNELS, int halfNumCoefs) {
83    // calculate desired state size
84    int stateSize = halfNumCoefs * CHANNELS * 2
85            * kStateSizeMultipleOfFilterLength;
86
87    // check if buffer needs resizing
88    if (mState
89            && stateSize == mStateSize
90            && mRingFull-mState == mStateSize-halfNumCoefs*CHANNELS) {
91        return;
92    }
93
94    // create new buffer
95    TI* state = (int16_t*)memalign(32, stateSize*sizeof(*state));
96    memset(state, 0, stateSize*sizeof(*state));
97
98    // attempt to preserve state
99    if (mState) {
100        TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
101        TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
102        TI* dst = state;
103
104        if (srcLo < mState) {
105            dst += mState-srcLo;
106            srcLo = mState;
107        }
108        if (srcHi > mState + mStateSize) {
109            srcHi = mState + mStateSize;
110        }
111        memcpy(dst, srcLo, (srcHi - srcLo) * sizeof(*srcLo));
112        free(mState);
113    }
114
115    // set class member vars
116    mState = state;
117    mStateSize = stateSize;
118    mImpulse = mState + halfNumCoefs*CHANNELS; // actually one sample greater than needed
119    mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS;
120}
121
122// copy in the input data into the head (impulse+halfNumCoefs) of the buffer.
123template<typename TI>
124template<int CHANNELS>
125void AudioResamplerDyn::InBuffer<TI>::readAgain(TI*& impulse, const int halfNumCoefs,
126        const TI* const in, const size_t inputIndex) {
127    int16_t* head = impulse + halfNumCoefs*CHANNELS;
128    for (size_t i=0 ; i<CHANNELS ; i++) {
129        head[i] = in[inputIndex*CHANNELS + i];
130    }
131}
132
133// advance the impulse pointer, and load in data into the head (impulse+halfNumCoefs)
134template<typename TI>
135template<int CHANNELS>
136void AudioResamplerDyn::InBuffer<TI>::readAdvance(TI*& impulse, const int halfNumCoefs,
137        const TI* const in, const size_t inputIndex) {
138    impulse += CHANNELS;
139
140    if (CC_UNLIKELY(impulse >= mRingFull)) {
141        const size_t shiftDown = mRingFull - mState - halfNumCoefs*CHANNELS;
142        memcpy(mState, mState+shiftDown, halfNumCoefs*CHANNELS*2*sizeof(TI));
143        impulse -= shiftDown;
144    }
145    readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
146}
147
148void AudioResamplerDyn::Constants::set(
149        int L, int halfNumCoefs, int inSampleRate, int outSampleRate)
150{
151    int bits = 0;
152    int lscale = inSampleRate/outSampleRate < 2 ? L - 1 :
153            static_cast<int>(static_cast<uint64_t>(L)*inSampleRate/outSampleRate);
154    for (int i=lscale; i; ++bits, i>>=1)
155        ;
156    mL = L;
157    mShift = kNumPhaseBits - bits;
158    mHalfNumCoefs = halfNumCoefs;
159}
160
161AudioResamplerDyn::AudioResamplerDyn(int bitDepth,
162        int inChannelCount, int32_t sampleRate, src_quality quality)
163    : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),
164    mResampleType(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),
165    mCoefBuffer(NULL)
166{
167    mVolumeSimd[0] = mVolumeSimd[1] = 0;
168    mConstants.set(128, 8, mSampleRate, mSampleRate); // TODO: set better
169}
170
171AudioResamplerDyn::~AudioResamplerDyn() {
172    free(mCoefBuffer);
173}
174
175void AudioResamplerDyn::init() {
176    mFilterSampleRate = 0; // always trigger new filter generation
177    mInBuffer.init();
178}
179
180void AudioResamplerDyn::setVolume(int16_t left, int16_t right) {
181    AudioResampler::setVolume(left, right);
182    mVolumeSimd[0] = static_cast<int32_t>(left)<<16;
183    mVolumeSimd[1] = static_cast<int32_t>(right)<<16;
184}
185
186template <typename T> T max(T a, T b) {return a > b ? a : b;}
187
188template <typename T> T absdiff(T a, T b) {return a > b ? a - b : b - a;}
189
190template<typename T>
191void AudioResamplerDyn::createKaiserFir(Constants &c, double stopBandAtten,
192        int inSampleRate, int outSampleRate, double tbwCheat) {
193    T* buf = reinterpret_cast<T*>(memalign(32, (c.mL+1)*c.mHalfNumCoefs*sizeof(T)));
194    static const double atten = 0.9998;   // to avoid ripple overflow
195    double fcr;
196    double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
197
198    if (inSampleRate < outSampleRate) { // upsample
199        fcr = max(0.5*tbwCheat - tbw/2, tbw/2);
200    } else { // downsample
201        fcr = max(0.5*tbwCheat*outSampleRate/inSampleRate - tbw/2, tbw/2);
202    }
203    // create and set filter
204    firKaiserGen(buf, c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten);
205    c.setBuf(buf);
206    if (mCoefBuffer) {
207        free(mCoefBuffer);
208    }
209    mCoefBuffer = buf;
210#ifdef DEBUG_RESAMPLER
211    // print basic filter stats
212    printf("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",
213            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, atten, tbw);
214    // test the filter and report results
215    double fp = (fcr - tbw/2)/c.mL;
216    double fs = (fcr + tbw/2)/c.mL;
217    double passMin, passMax, passRipple;
218    double stopMax, stopRipple;
219    testFir(buf, c.mL, c.mHalfNumCoefs, fp, fs, /*passSteps*/ 1000, /*stopSteps*/ 100000,
220            passMin, passMax, passRipple, stopMax, stopRipple);
221    printf("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
222    printf("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
223#endif
224}
225
226// recursive gcd. Using objdump, it appears the tail recursion is converted to a while loop.
227static int gcd(int n, int m) {
228    if (m == 0) {
229        return n;
230    }
231    return gcd(m, n % m);
232}
233
234static bool isClose(int32_t newSampleRate, int32_t prevSampleRate,
235        int32_t filterSampleRate, int32_t outSampleRate) {
236
237    // different upsampling ratios do not need a filter change.
238    if (filterSampleRate != 0
239            && filterSampleRate < outSampleRate
240            && newSampleRate < outSampleRate)
241        return true;
242
243    // check design criteria again if downsampling is detected.
244    int pdiff = absdiff(newSampleRate, prevSampleRate);
245    int adiff = absdiff(newSampleRate, filterSampleRate);
246
247    // allow up to 6% relative change increments.
248    // allow up to 12% absolute change increments (from filter design)
249    return pdiff < prevSampleRate>>4 && adiff < filterSampleRate>>3;
250}
251
252void AudioResamplerDyn::setSampleRate(int32_t inSampleRate) {
253    if (mInSampleRate == inSampleRate) {
254        return;
255    }
256    int32_t oldSampleRate = mInSampleRate;
257    int32_t oldHalfNumCoefs = mConstants.mHalfNumCoefs;
258    uint32_t oldPhaseWrapLimit = mConstants.mL << mConstants.mShift;
259    bool useS32 = false;
260
261    mInSampleRate = inSampleRate;
262
263    // TODO: Add precalculated Equiripple filters
264
265    if (mFilterQuality != getQuality() ||
266            !isClose(inSampleRate, oldSampleRate, mFilterSampleRate, mSampleRate)) {
267        mFilterSampleRate = inSampleRate;
268        mFilterQuality = getQuality();
269
270        // Begin Kaiser Filter computation
271        //
272        // The quantization floor for S16 is about 96db - 10*log_10(#length) + 3dB.
273        // Keep the stop band attenuation no greater than 84-85dB for 32 length S16 filters
274        //
275        // For s32 we keep the stop band attenuation at the same as 16b resolution, about
276        // 96-98dB
277        //
278
279        double stopBandAtten;
280        double tbwCheat = 1.; // how much we "cheat" into aliasing
281        int halfLength;
282        if (mFilterQuality == DYN_HIGH_QUALITY) {
283            // 32b coefficients, 64 length
284            useS32 = true;
285            stopBandAtten = 98.;
286            if (inSampleRate >= mSampleRate * 4) {
287                halfLength = 48;
288            } else if (inSampleRate >= mSampleRate * 2) {
289                halfLength = 40;
290            } else {
291                halfLength = 32;
292            }
293        } else if (mFilterQuality == DYN_LOW_QUALITY) {
294            // 16b coefficients, 16-32 length
295            useS32 = false;
296            stopBandAtten = 80.;
297            if (inSampleRate >= mSampleRate * 4) {
298                halfLength = 24;
299            } else if (inSampleRate >= mSampleRate * 2) {
300                halfLength = 16;
301            } else {
302                halfLength = 8;
303            }
304            if (inSampleRate <= mSampleRate) {
305                tbwCheat = 1.05;
306            } else {
307                tbwCheat = 1.03;
308            }
309        } else { // DYN_MED_QUALITY
310            // 16b coefficients, 32-64 length
311            // note: > 64 length filters with 16b coefs can have quantization noise problems
312            useS32 = false;
313            stopBandAtten = 84.;
314            if (inSampleRate >= mSampleRate * 4) {
315                halfLength = 32;
316            } else if (inSampleRate >= mSampleRate * 2) {
317                halfLength = 24;
318            } else {
319                halfLength = 16;
320            }
321            if (inSampleRate <= mSampleRate) {
322                tbwCheat = 1.03;
323            } else {
324                tbwCheat = 1.01;
325            }
326        }
327
328        // determine the number of polyphases in the filterbank.
329        // for 16b, it is desirable to have 2^(16/2) = 256 phases.
330        // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html
331        //
332        // We are a bit more lax on this.
333
334        int phases = mSampleRate / gcd(mSampleRate, inSampleRate);
335
336        // TODO: Once dynamic sample rate change is an option, the code below
337        // should be modified to execute only when dynamic sample rate change is enabled.
338        //
339        // as above, #phases less than 63 is too few phases for accurate linear interpolation.
340        // we increase the phases to compensate, but more phases means more memory per
341        // filter and more time to compute the filter.
342        //
343        // if we know that the filter will be used for dynamic sample rate changes,
344        // that would allow us skip this part for fixed sample rate resamplers.
345        //
346        while (phases<63) {
347            phases *= 2; // this code only needed to support dynamic rate changes
348        }
349
350        if (phases>=256) {  // too many phases, always interpolate
351            phases = 127;
352        }
353
354        // create the filter
355        mConstants.set(phases, halfLength, inSampleRate, mSampleRate);
356        if (useS32) {
357            createKaiserFir<int32_t>(mConstants, stopBandAtten,
358                    inSampleRate, mSampleRate, tbwCheat);
359        } else {
360            createKaiserFir<int16_t>(mConstants, stopBandAtten,
361                    inSampleRate, mSampleRate, tbwCheat);
362        }
363    } // End Kaiser filter
364
365    // update phase and state based on the new filter.
366    const Constants& c(mConstants);
367    mInBuffer.resize(mChannelCount, c.mHalfNumCoefs);
368    const uint32_t phaseWrapLimit = c.mL << c.mShift;
369    // try to preserve as much of the phase fraction as possible for on-the-fly changes
370    mPhaseFraction = static_cast<unsigned long long>(mPhaseFraction)
371            * phaseWrapLimit / oldPhaseWrapLimit;
372    mPhaseFraction %= phaseWrapLimit; // should not do anything, but just in case.
373    mPhaseIncrement = static_cast<uint32_t>(static_cast<double>(phaseWrapLimit)
374            * inSampleRate / mSampleRate);
375
376    // determine which resampler to use
377    // check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
378    int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
379    int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2;
380    if (locked) {
381        mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
382    }
383
384    mResampleType = RESAMPLETYPE(mChannelCount, locked, stride, !!useS32);
385#ifdef DEBUG_RESAMPLER
386    printf("channels:%d  %s  stride:%d  %s  coef:%d  shift:%d\n",
387            mChannelCount, locked ? "locked" : "interpolated",
388            stride, useS32 ? "S32" : "S16", 2*c.mHalfNumCoefs, c.mShift);
389#endif
390}
391
392void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
393            AudioBufferProvider* provider)
394{
395    // TODO:
396    // 24 cases - this perhaps can be reduced later, as testing might take too long
397    switch (mResampleType) {
398
399    // stride 16 (falls back to stride 2 for machines that do not support NEON)
400    case RESAMPLETYPE(1, true, 16, 0):
401        return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
402    case RESAMPLETYPE(2, true, 16, 0):
403        return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
404    case RESAMPLETYPE(1, false, 16, 0):
405        return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
406    case RESAMPLETYPE(2, false, 16, 0):
407        return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
408    case RESAMPLETYPE(1, true, 16, 1):
409        return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
410    case RESAMPLETYPE(2, true, 16, 1):
411        return resample<2, true, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
412    case RESAMPLETYPE(1, false, 16, 1):
413        return resample<1, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
414    case RESAMPLETYPE(2, false, 16, 1):
415        return resample<2, false, 16>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
416#if 0
417    // TODO: Remove these?
418    // stride 8
419    case RESAMPLETYPE(1, true, 8, 0):
420        return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
421    case RESAMPLETYPE(2, true, 8, 0):
422        return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
423    case RESAMPLETYPE(1, false, 8, 0):
424        return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
425    case RESAMPLETYPE(2, false, 8, 0):
426        return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
427    case RESAMPLETYPE(1, true, 8, 1):
428        return resample<1, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
429    case RESAMPLETYPE(2, true, 8, 1):
430        return resample<2, true, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
431    case RESAMPLETYPE(1, false, 8, 1):
432        return resample<1, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
433    case RESAMPLETYPE(2, false, 8, 1):
434        return resample<2, false, 8>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
435    // stride 2 (can handle any filter length)
436    case RESAMPLETYPE(1, true, 2, 0):
437        return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
438    case RESAMPLETYPE(2, true, 2, 0):
439        return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
440    case RESAMPLETYPE(1, false, 2, 0):
441        return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
442    case RESAMPLETYPE(2, false, 2, 0):
443        return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
444    case RESAMPLETYPE(1, true, 2, 1):
445        return resample<1, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
446    case RESAMPLETYPE(2, true, 2, 1):
447        return resample<2, true, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
448    case RESAMPLETYPE(1, false, 2, 1):
449        return resample<1, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
450    case RESAMPLETYPE(2, false, 2, 1):
451        return resample<2, false, 2>(out, outFrameCount, mConstants.mFirCoefsS32, provider);
452#endif
453    default:
454        ; // error
455    }
456}
457
458template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
459void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount,
460        const TC* const coefs,  AudioBufferProvider* provider)
461{
462    const Constants& c(mConstants);
463    int16_t* impulse = mInBuffer.getImpulse();
464    size_t inputIndex = mInputIndex;
465    uint32_t phaseFraction = mPhaseFraction;
466    const uint32_t phaseIncrement = mPhaseIncrement;
467    size_t outputIndex = 0;
468    size_t outputSampleCount = outFrameCount * 2;   // stereo output
469    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
470    const uint32_t phaseWrapLimit = c.mL << c.mShift;
471
472    // NOTE: be very careful when modifying the code here. register
473    // pressure is very high and a small change might cause the compiler
474    // to generate far less efficient code.
475    // Always sanity check the result with objdump or test-resample.
476
477    // the following logic is a bit convoluted to keep the main processing loop
478    // as tight as possible with register allocation.
479    while (outputIndex < outputSampleCount) {
480        // buffer is empty, fetch a new one
481        while (mBuffer.frameCount == 0) {
482            mBuffer.frameCount = inFrameCount;
483            provider->getNextBuffer(&mBuffer,
484                    calculateOutputPTS(outputIndex / 2));
485            if (mBuffer.raw == NULL) {
486                goto resample_exit;
487            }
488            if (phaseFraction >= phaseWrapLimit) { // read in data
489                mInBuffer.readAdvance<CHANNELS>(
490                        impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
491                phaseFraction -= phaseWrapLimit;
492                while (phaseFraction >= phaseWrapLimit) {
493                    inputIndex++;
494                    if (inputIndex >= mBuffer.frameCount) {
495                        inputIndex -= mBuffer.frameCount;
496                        provider->releaseBuffer(&mBuffer);
497                        break;
498                    }
499                    mInBuffer.readAdvance<CHANNELS>(
500                            impulse, c.mHalfNumCoefs, mBuffer.i16, inputIndex);
501                    phaseFraction -= phaseWrapLimit;
502                }
503            }
504        }
505        const int16_t* const in = mBuffer.i16;
506        const size_t frameCount = mBuffer.frameCount;
507        const int coefShift = c.mShift;
508        const int halfNumCoefs = c.mHalfNumCoefs;
509        const int32_t* const volumeSimd = mVolumeSimd;
510
511        // reread the last input in.
512        mInBuffer.readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
513
514        // main processing loop
515        while (CC_LIKELY(outputIndex < outputSampleCount)) {
516            // caution: fir() is inlined and may be large.
517            // output will be loaded with the appropriate values
518            //
519            // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs]
520            // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs.
521            //
522            fir<CHANNELS, LOCKED, STRIDE>(
523                    &out[outputIndex],
524                    phaseFraction, phaseWrapLimit,
525                    coefShift, halfNumCoefs, coefs,
526                    impulse, volumeSimd);
527            outputIndex += 2;
528
529            phaseFraction += phaseIncrement;
530            while (phaseFraction >= phaseWrapLimit) {
531                inputIndex++;
532                if (inputIndex >= frameCount) {
533                    goto done;  // need a new buffer
534                }
535                mInBuffer.readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex);
536                phaseFraction -= phaseWrapLimit;
537            }
538        }
539done:
540        // often arrives here when input buffer runs out
541        if (inputIndex >= frameCount) {
542            inputIndex -= frameCount;
543            provider->releaseBuffer(&mBuffer);
544            // mBuffer.frameCount MUST be zero here.
545        }
546    }
547
548resample_exit:
549    mInBuffer.setImpulse(impulse);
550    mInputIndex = inputIndex;
551    mPhaseFraction = phaseFraction;
552}
553
554// ----------------------------------------------------------------------------
555}; // namespace android
556