services/audioflinger/tests/resampler_tests.cpp - android_frameworks_av - Gitiles

 /*
  * Copyright (C) 2014 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_NDEBUG 0
 #define LOG_TAG "audioflinger_resampler_tests"

 #include <unistd.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <fcntl.h>
 #include <string.h>
 #include <sys/mman.h>
 #include <sys/stat.h>
 #include <errno.h>
 #include <time.h>
 #include <math.h>
 #include <vector>
 #include <utility>
 #include <cutils/log.h>
 #include <gtest/gtest.h>
 #include <media/AudioBufferProvider.h>
 #include "AudioResampler.h"

 #define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))

 template<typename T, typename U>
 struct is_same
 {
     static const bool value = false;
 };

 template<typename T>
 struct is_same<T, T>  // partial specialization
 {
     static const bool value = true;
 };

 template<typename T>
 static inline T convertValue(double val)
 {
     if (is_same<T, int16_t>::value) {
         return floor(val * 32767.0 + 0.5);
     } else if (is_same<T, int32_t>::value) {
         return floor(val * (1UL<<31) + 0.5);
     }
     return val; // assume float or double
 }

 /* Creates a type-independent audio buffer provider from
  * a buffer base address, size, framesize, and input increment array.
  *
  * No allocation or deallocation of the provided buffer is done.
  */
 class TestProvider : public android::AudioBufferProvider {
 public:
     TestProvider(const void* addr, size_t frames, size_t frameSize,
             const std::vector<size_t>& inputIncr)
     : mAddr(addr),
       mNumFrames(frames),
       mFrameSize(frameSize),
       mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
     {
     }

     virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS )
     {
         size_t requestedFrames = buffer->frameCount;
         if (requestedFrames > mNumFrames - mNextFrame) {
             buffer->frameCount = mNumFrames - mNextFrame;
         }
         if (!mInputIncr.empty()) {
             size_t provided = mInputIncr[mNextIdx++];
             ALOGV("getNextBuffer() mValue[%d]=%u not %u",
                     mNextIdx-1, provided, buffer->frameCount);
             if (provided < buffer->frameCount) {
                 buffer->frameCount = provided;
             }
             if (mNextIdx >= mInputIncr.size()) {
                 mNextIdx = 0;
             }
         }
         ALOGV("getNextBuffer() requested %u frames out of %u frames available"
                 " and returned %u frames\n",
                 requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
         mUnrel = buffer->frameCount;
         if (buffer->frameCount > 0) {
             buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
             return android::NO_ERROR;
         } else {
             buffer->raw = NULL;
             return android::NOT_ENOUGH_DATA;
         }
     }

     virtual void releaseBuffer(Buffer* buffer)
     {
         if (buffer->frameCount > mUnrel) {
             ALOGE("releaseBuffer() released %u frames but only %u available "
                     "to release\n", buffer->frameCount, mUnrel);
             mNextFrame += mUnrel;
             mUnrel = 0;
         } else {

             ALOGV("releaseBuffer() released %u frames out of %u frames available "
                     "to release\n", buffer->frameCount, mUnrel);
             mNextFrame += buffer->frameCount;
             mUnrel -= buffer->frameCount;
         }
         buffer->frameCount = 0;
         buffer->raw = NULL;
     }

     void reset()
     {
         mNextFrame = 0;
     }

     size_t getNumFrames()
     {
         return mNumFrames;
     }

     void setIncr(const std::vector<size_t> inputIncr)
     {
         mNextIdx = 0;
         mInputIncr = inputIncr;
     }

 protected:
     const void* mAddr;   // base address
     size_t mNumFrames;   // total frames
     int mFrameSize;      // frame size (# channels * bytes per sample)
     size_t mNextFrame;   // index of next frame to provide
     size_t mUnrel;       // number of frames not yet released
     std::vector<size_t> mInputIncr; // number of frames provided per call
     size_t mNextIdx;     // index of next entry in mInputIncr to use
 };

 /* Creates a buffer filled with a sine wave.
  *
  * Returns a pair consisting of the sine signal buffer and the number of frames.
  * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
  */
 template<typename T>
 static std::pair<T*, size_t> createSine(size_t channels,
         double freq, double samplingRate, double time)
 {
     double tscale = 1. / samplingRate;
     size_t frames = static_cast<size_t>(samplingRate * time);
     T* buffer = new T[frames * channels];
     for (size_t i = 0; i < frames; ++i) {
         double t = i * tscale;
         double y = sin(2. * M_PI * freq * t);
         T yt = convertValue<T>(y);

         for (size_t j = 0; j < channels; ++j) {
             buffer[i*channels + j] = yt / (j + 1);
         }
     }
     return std::make_pair(buffer, frames);
 }

 /* Creates a buffer filled with a chirp signal (a sine wave sweep).
  *
  * Returns a pair consisting of the chirp signal buffer and the number of frames.
  * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
  *
  * When creating the Chirp, note that the frequency is the true sinusoidal
  * frequency not the sampling rate.
  *
  * http://en.wikipedia.org/wiki/Chirp
  */
 template<typename T>
 static std::pair<T*, size_t> createChirp(size_t channels,
         double minfreq, double maxfreq, double samplingRate, double time)
 {
     double tscale = 1. / samplingRate;
     size_t frames = static_cast<size_t>(samplingRate * time);
     T *buffer = new T[frames * channels];
     // note the chirp constant k has a divide-by-two.
     double k = (maxfreq - minfreq) / (2. * time);
     for (size_t i = 0; i < frames; ++i) {
         double t = i * tscale;
         double y = sin(2. * M_PI * (k * t + minfreq) * t);
         T yt = convertValue<T>(y);

         for (size_t j = 0; j < channels; ++j) {
             buffer[i*channels + j] = yt / (j + 1);
         }
     }
     return std::make_pair(buffer, frames);
 }

 /* This derived class creates a buffer provider of datatype T,
  * consisting of an input signal, e.g. from createChirp().
  * The number of frames can be obtained from the base class
  * TestProvider::getNumFrames().
  */
 template <typename T>
 class SignalProvider : public TestProvider {
 public:
     SignalProvider(const std::pair<T*, size_t>& bufferInfo, size_t channels,
             const std::vector<size_t>& values)
     : TestProvider(bufferInfo.first, bufferInfo.second, channels * sizeof(T), values),
       mManagedPtr(bufferInfo.first)
     {
     }

     virtual ~SignalProvider()
     {
         delete[] mManagedPtr;
     }

 protected:
     T* mManagedPtr;
 };

 void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr,
         android::AudioBufferProvider *provider, android::AudioResampler *resampler)
 {
     for (size_t i = 0, j = 0; i < outputFrames; ) {
         size_t thisFrames = outputIncr[j++];
         if (j >= outputIncr.size()) {
             j = 0;
         }
         if (thisFrames == 0 || thisFrames > outputFrames - i) {
             thisFrames = outputFrames - i;
         }
         resampler->resample((int32_t*) output + 2*i, thisFrames, provider);
         i += thisFrames;
     }
 }

 void buffercmp(const void *reference, const void *test,
         size_t outputFrameSize, size_t outputFrames)
 {
     for (size_t i = 0; i < outputFrames; ++i) {
         int check = memcmp((const char*)reference + i * outputFrameSize,
                 (const char*)test + i * outputFrameSize, outputFrameSize);
         if (check) {
             ALOGE("Failure at frame %d", i);
             ASSERT_EQ(check, 0); /* fails */
         }
     }
 }

 void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFreq,
         enum android::AudioResampler::src_quality quality)
 {
     // create the provider
     std::vector<size_t> inputIncr;
     SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
             0., outputFreq/2., outputFreq, outputFreq/2000.),
             channels, inputIncr);

     // calculate the output size
     size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
     size_t outputFrameSize = 2 * sizeof(int32_t);
     size_t outputSize = outputFrameSize * outputFrames;
     outputSize &= ~7;

     // create the resampler
     const int volumePrecision = 12; /* typical unity gain */
     android::AudioResampler* resampler;

     resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);

     // set up the reference run
     std::vector<size_t> refIncr;
     refIncr.push_back(outputFrames);
     void* reference = malloc(outputSize);
     resample(reference, outputFrames, refIncr, &provider, resampler);

     provider.reset();

 #if 0
     /* this test will fail - API interface issue: reset() does not clear internal buffers */
     resampler->reset();
 #else
     delete resampler;
     resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
 #endif

     // set up the test run
     std::vector<size_t> outIncr;
     outIncr.push_back(1);
     outIncr.push_back(2);
     outIncr.push_back(3);
     void* test = malloc(outputSize);
     resample(test, outputFrames, outIncr, &provider, resampler);

     // check
     buffercmp(reference, test, outputFrameSize, outputFrames);

     free(reference);
     free(test);
     delete resampler;
 }

 template <typename T>
 inline double sqr(T v)
 {
     double dv = static_cast<double>(v);
     return dv * dv;
 }

 template <typename T>
 double signalEnergy(T *start, T *end, unsigned stride)
 {
     double accum = 0;

     for (T *p = start; p < end; p += stride) {
         accum += sqr(*p);
     }
     unsigned count = (end - start + stride - 1) / stride;
     return accum / count;
 }

 void testStopbandDownconversion(size_t channels,
         unsigned inputFreq, unsigned outputFreq,
         unsigned passband, unsigned stopband,
         enum android::AudioResampler::src_quality quality)
 {
     // create the provider
     std::vector<size_t> inputIncr;
     SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
             0., inputFreq/2., inputFreq, inputFreq/2000.),
             channels, inputIncr);

     // calculate the output size
     size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
     size_t outputFrameSize = 2 * sizeof(int32_t);
     size_t outputSize = outputFrameSize * outputFrames;
     outputSize &= ~7;

     // create the resampler
     const int volumePrecision = 12; /* typical unity gain */
     android::AudioResampler* resampler;

     resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
     resampler->setSampleRate(inputFreq);
     resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);

     // set up the reference run
     std::vector<size_t> refIncr;
     refIncr.push_back(outputFrames);
     void* reference = malloc(outputSize);
     resample(reference, outputFrames, refIncr, &provider, resampler);

     int32_t *out = reinterpret_cast<int32_t *>(reference);

     // check signal energy in passband
     const unsigned passbandFrame = passband * outputFreq / 1000.;
     const unsigned stopbandFrame = stopband * outputFreq / 1000.;

     // check each channel separately
     for (size_t i = 0; i < channels; ++i) {
         double passbandEnergy = signalEnergy(out, out + passbandFrame * channels, channels);
         double stopbandEnergy = signalEnergy(out + stopbandFrame * channels,
                 out + outputFrames * channels, channels);
         double dbAtten = -10. * log10(stopbandEnergy / passbandEnergy);
         ASSERT_GT(dbAtten, 60.);

 #if 0
         // internal verification
         printf("if:%d  of:%d  pbf:%d  sbf:%d  sbe: %f  pbe: %f  db: %.2f\n",
                 provider.getNumFrames(), outputFrames,
                 passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten);
         for (size_t i = 0; i < 10; ++i) {
             printf("%d\n", out[i+passbandFrame*channels]);
         }
         for (size_t i = 0; i < 10; ++i) {
             printf("%d\n", out[i+stopbandFrame*channels]);
         }
 #endif
     }

     free(reference);
     delete resampler;
 }

 /* Buffer increment test
  *
  * We compare a reference output, where we consume and process the entire
  * buffer at a time, and a test output, where we provide small chunks of input
  * data and process small chunks of output (which may not be equivalent in size).
  *
  * Two subtests - fixed phase (3:2 down) and interpolated phase (147:320 up)
  */
 TEST(audioflinger_resampler, bufferincrement_fixedphase) {
     // all of these work
     static const enum android::AudioResampler::src_quality kQualityArray[] = {
             android::AudioResampler::LOW_QUALITY,
             android::AudioResampler::MED_QUALITY,
             android::AudioResampler::HIGH_QUALITY,
             android::AudioResampler::VERY_HIGH_QUALITY,
             android::AudioResampler::DYN_LOW_QUALITY,
             android::AudioResampler::DYN_MED_QUALITY,
             android::AudioResampler::DYN_HIGH_QUALITY,
     };

     for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
         testBufferIncrement(2, 48000, 32000, kQualityArray[i]);
     }
 }

 TEST(audioflinger_resampler, bufferincrement_interpolatedphase) {
     // all of these work except low quality
     static const enum android::AudioResampler::src_quality kQualityArray[] = {
 //           android::AudioResampler::LOW_QUALITY,
             android::AudioResampler::MED_QUALITY,
             android::AudioResampler::HIGH_QUALITY,
             android::AudioResampler::VERY_HIGH_QUALITY,
             android::AudioResampler::DYN_LOW_QUALITY,
             android::AudioResampler::DYN_MED_QUALITY,
             android::AudioResampler::DYN_HIGH_QUALITY,
     };

     for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
         testBufferIncrement(2, 22050, 48000, kQualityArray[i]);
     }
 }

 /* Simple aliasing test
  *
  * This checks stopband response of the chirp signal to make sure frequencies
  * are properly suppressed.  It uses downsampling because the stopband can be
  * clearly isolated by input frequencies exceeding the output sample rate (nyquist).
  */
 TEST(audioflinger_resampler, stopbandresponse) {
     // not all of these may work (old resamplers fail on downsampling)
     static const enum android::AudioResampler::src_quality kQualityArray[] = {
             //android::AudioResampler::LOW_QUALITY,
             //android::AudioResampler::MED_QUALITY,
             //android::AudioResampler::HIGH_QUALITY,
             //android::AudioResampler::VERY_HIGH_QUALITY,
             android::AudioResampler::DYN_LOW_QUALITY,
             android::AudioResampler::DYN_MED_QUALITY,
             android::AudioResampler::DYN_HIGH_QUALITY,
     };

     // in this test we assume a maximum transition band between 12kHz and 20kHz.
     // there must be at least 60dB relative attenuation between stopband and passband.
     for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
         testStopbandDownconversion(2, 48000, 32000, 12000, 20000, kQualityArray[i]);
     }

     // in this test we assume a maximum transition band between 7kHz and 15kHz.
     // there must be at least 60dB relative attenuation between stopband and passband.
     // (the weird ratio triggers interpolative resampling)
     for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
         testStopbandDownconversion(2, 48000, 22101, 7000, 15000, kQualityArray[i]);
     }
 }
	/*
	* Copyright (C) 2014 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_NDEBUG 0
	#define LOG_TAG "audioflinger_resampler_tests"

	#include <unistd.h>
	#include <stdio.h>
	#include <stdlib.h>
	#include <fcntl.h>
	#include <string.h>
	#include <sys/mman.h>
	#include <sys/stat.h>
	#include <errno.h>
	#include <time.h>
	#include <math.h>
	#include <vector>
	#include <utility>
	#include <cutils/log.h>
	#include <gtest/gtest.h>
	#include <media/AudioBufferProvider.h>
	#include "AudioResampler.h"

	#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))

	template<typename T, typename U>
	struct is_same
	{
	static const bool value = false;
	};

	template<typename T>
	struct is_same<T, T> // partial specialization
	{
	static const bool value = true;
	};

	template<typename T>
	static inline T convertValue(double val)
	{
	if (is_same<T, int16_t>::value) {
	return floor(val * 32767.0 + 0.5);
	} else if (is_same<T, int32_t>::value) {
	return floor(val * (1UL<<31) + 0.5);
	}
	return val; // assume float or double
	}

	/* Creates a type-independent audio buffer provider from
	* a buffer base address, size, framesize, and input increment array.
	*
	* No allocation or deallocation of the provided buffer is done.
	*/
	class TestProvider : public android::AudioBufferProvider {
	public:
	TestProvider(const void* addr, size_t frames, size_t frameSize,
	const std::vector<size_t>& inputIncr)
	: mAddr(addr),
	mNumFrames(frames),
	mFrameSize(frameSize),
	mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
	{
	}

	virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS )
	{
	size_t requestedFrames = buffer->frameCount;
	if (requestedFrames > mNumFrames - mNextFrame) {
	buffer->frameCount = mNumFrames - mNextFrame;
	}
	if (!mInputIncr.empty()) {
	size_t provided = mInputIncr[mNextIdx++];
	ALOGV("getNextBuffer() mValue[%d]=%u not %u",
	mNextIdx-1, provided, buffer->frameCount);
	if (provided < buffer->frameCount) {
	buffer->frameCount = provided;
	}
	if (mNextIdx >= mInputIncr.size()) {
	mNextIdx = 0;
	}
	}
	ALOGV("getNextBuffer() requested %u frames out of %u frames available"
	" and returned %u frames\n",
	requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
	mUnrel = buffer->frameCount;
	if (buffer->frameCount > 0) {
	buffer->raw = (char )mAddr + mFrameSize mNextFrame;
	return android::NO_ERROR;
	} else {
	buffer->raw = NULL;
	return android::NOT_ENOUGH_DATA;
	}
	}

	virtual void releaseBuffer(Buffer* buffer)
	{
	if (buffer->frameCount > mUnrel) {
	ALOGE("releaseBuffer() released %u frames but only %u available "
	"to release\n", buffer->frameCount, mUnrel);
	mNextFrame += mUnrel;
	mUnrel = 0;
	} else {

	ALOGV("releaseBuffer() released %u frames out of %u frames available "
	"to release\n", buffer->frameCount, mUnrel);
	mNextFrame += buffer->frameCount;
	mUnrel -= buffer->frameCount;
	}
	buffer->frameCount = 0;
	buffer->raw = NULL;
	}

	void reset()
	{
	mNextFrame = 0;
	}

	size_t getNumFrames()
	{
	return mNumFrames;
	}

	void setIncr(const std::vector<size_t> inputIncr)
	{
	mNextIdx = 0;
	mInputIncr = inputIncr;
	}

	protected:
	const void* mAddr; // base address
	size_t mNumFrames; // total frames
	int mFrameSize; // frame size (# channels * bytes per sample)
	size_t mNextFrame; // index of next frame to provide
	size_t mUnrel; // number of frames not yet released
	std::vector<size_t> mInputIncr; // number of frames provided per call
	size_t mNextIdx; // index of next entry in mInputIncr to use
	};

	/* Creates a buffer filled with a sine wave.
	*
	* Returns a pair consisting of the sine signal buffer and the number of frames.
	* The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
	*/
	template<typename T>
	static std::pair<T*, size_t> createSine(size_t channels,
	double freq, double samplingRate, double time)
	{
	double tscale = 1. / samplingRate;
	size_t frames = static_cast<size_t>(samplingRate * time);
	T* buffer = new T[frames * channels];
	for (size_t i = 0; i < frames; ++i) {
	double t = i * tscale;
	double y = sin(2. * M_PI * freq * t);
	T yt = convertValue<T>(y);

	for (size_t j = 0; j < channels; ++j) {
	buffer[i*channels + j] = yt / (j + 1);
	}
	}
	return std::make_pair(buffer, frames);
	}

	/* Creates a buffer filled with a chirp signal (a sine wave sweep).
	*
	* Returns a pair consisting of the chirp signal buffer and the number of frames.
	* The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
	*
	* When creating the Chirp, note that the frequency is the true sinusoidal
	* frequency not the sampling rate.
	*
	* http://en.wikipedia.org/wiki/Chirp
	*/
	template<typename T>
	static std::pair<T*, size_t> createChirp(size_t channels,
	double minfreq, double maxfreq, double samplingRate, double time)
	{
	double tscale = 1. / samplingRate;
	size_t frames = static_cast<size_t>(samplingRate * time);
	T buffer = new T[frames channels];
	// note the chirp constant k has a divide-by-two.
	double k = (maxfreq - minfreq) / (2. * time);
	for (size_t i = 0; i < frames; ++i) {
	double t = i * tscale;
	double y = sin(2. * M_PI * (k * t + minfreq) * t);
	T yt = convertValue<T>(y);

	for (size_t j = 0; j < channels; ++j) {
	buffer[i*channels + j] = yt / (j + 1);
	}
	}
	return std::make_pair(buffer, frames);
	}

	/* This derived class creates a buffer provider of datatype T,
	* consisting of an input signal, e.g. from createChirp().
	* The number of frames can be obtained from the base class
	* TestProvider::getNumFrames().
	*/
	template <typename T>
	class SignalProvider : public TestProvider {
	public:
	SignalProvider(const std::pair<T*, size_t>& bufferInfo, size_t channels,
	const std::vector<size_t>& values)
	: TestProvider(bufferInfo.first, bufferInfo.second, channels * sizeof(T), values),
	mManagedPtr(bufferInfo.first)
	{
	}

	virtual ~SignalProvider()
	{
	delete[] mManagedPtr;
	}

	protected:
	T* mManagedPtr;
	};

	void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr,
	android::AudioBufferProvider provider, android::AudioResampler resampler)
	{
	for (size_t i = 0, j = 0; i < outputFrames; ) {
	size_t thisFrames = outputIncr[j++];
	if (j >= outputIncr.size()) {
	j = 0;
	}
	if (thisFrames == 0 \|\| thisFrames > outputFrames - i) {
	thisFrames = outputFrames - i;
	}
	resampler->resample((int32_t) output + 2i, thisFrames, provider);
	i += thisFrames;
	}
	}

	void buffercmp(const void reference, const void test,
	size_t outputFrameSize, size_t outputFrames)
	{
	for (size_t i = 0; i < outputFrames; ++i) {
	int check = memcmp((const char)reference + i outputFrameSize,
	(const char)test + i outputFrameSize, outputFrameSize);
	if (check) {
	ALOGE("Failure at frame %d", i);
	ASSERT_EQ(check, 0); /* fails */
	}
	}
	}

	void testBufferIncrement(size_t channels, unsigned inputFreq, unsigned outputFreq,
	enum android::AudioResampler::src_quality quality)
	{
	// create the provider
	std::vector<size_t> inputIncr;
	SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
	0., outputFreq/2., outputFreq, outputFreq/2000.),
	channels, inputIncr);

	// calculate the output size
	size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
	size_t outputFrameSize = 2 * sizeof(int32_t);
	size_t outputSize = outputFrameSize * outputFrames;
	outputSize &= ~7;

	// create the resampler
	const int volumePrecision = 12; /* typical unity gain */
	android::AudioResampler* resampler;

	resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
	resampler->setSampleRate(inputFreq);
	resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);

	// set up the reference run
	std::vector<size_t> refIncr;
	refIncr.push_back(outputFrames);
	void* reference = malloc(outputSize);
	resample(reference, outputFrames, refIncr, &provider, resampler);

	provider.reset();

	#if 0
	/* this test will fail - API interface issue: reset() does not clear internal buffers */
	resampler->reset();
	#else
	delete resampler;
	resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
	resampler->setSampleRate(inputFreq);
	resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);
	#endif

	// set up the test run
	std::vector<size_t> outIncr;
	outIncr.push_back(1);
	outIncr.push_back(2);
	outIncr.push_back(3);
	void* test = malloc(outputSize);
	resample(test, outputFrames, outIncr, &provider, resampler);

	// check
	buffercmp(reference, test, outputFrameSize, outputFrames);

	free(reference);
	free(test);
	delete resampler;
	}

	template <typename T>
	inline double sqr(T v)
	{
	double dv = static_cast<double>(v);
	return dv * dv;
	}

	template <typename T>
	double signalEnergy(T start, T end, unsigned stride)
	{
	double accum = 0;

	for (T *p = start; p < end; p += stride) {
	accum += sqr(*p);
	}
	unsigned count = (end - start + stride - 1) / stride;
	return accum / count;
	}

	void testStopbandDownconversion(size_t channels,
	unsigned inputFreq, unsigned outputFreq,
	unsigned passband, unsigned stopband,
	enum android::AudioResampler::src_quality quality)
	{
	// create the provider
	std::vector<size_t> inputIncr;
	SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
	0., inputFreq/2., inputFreq, inputFreq/2000.),
	channels, inputIncr);

	// calculate the output size
	size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
	size_t outputFrameSize = 2 * sizeof(int32_t);
	size_t outputSize = outputFrameSize * outputFrames;
	outputSize &= ~7;

	// create the resampler
	const int volumePrecision = 12; /* typical unity gain */
	android::AudioResampler* resampler;

	resampler = android::AudioResampler::create(16, channels, outputFreq, quality);
	resampler->setSampleRate(inputFreq);
	resampler->setVolume(1 << volumePrecision, 1 << volumePrecision);

	// set up the reference run
	std::vector<size_t> refIncr;
	refIncr.push_back(outputFrames);
	void* reference = malloc(outputSize);
	resample(reference, outputFrames, refIncr, &provider, resampler);

	int32_t out = reinterpret_cast<int32_t >(reference);

	// check signal energy in passband
	const unsigned passbandFrame = passband * outputFreq / 1000.;
	const unsigned stopbandFrame = stopband * outputFreq / 1000.;

	// check each channel separately
	for (size_t i = 0; i < channels; ++i) {
	double passbandEnergy = signalEnergy(out, out + passbandFrame * channels, channels);
	double stopbandEnergy = signalEnergy(out + stopbandFrame * channels,
	out + outputFrames * channels, channels);
	double dbAtten = -10. * log10(stopbandEnergy / passbandEnergy);
	ASSERT_GT(dbAtten, 60.);

	#if 0
	// internal verification
	printf("if:%d of:%d pbf:%d sbf:%d sbe: %f pbe: %f db: %.2f\n",
	provider.getNumFrames(), outputFrames,
	passbandFrame, stopbandFrame, stopbandEnergy, passbandEnergy, dbAtten);
	for (size_t i = 0; i < 10; ++i) {
	printf("%d\n", out[i+passbandFrame*channels]);
	}
	for (size_t i = 0; i < 10; ++i) {
	printf("%d\n", out[i+stopbandFrame*channels]);
	}
	#endif
	}

	free(reference);
	delete resampler;
	}

	/* Buffer increment test
	*
	* We compare a reference output, where we consume and process the entire
	* buffer at a time, and a test output, where we provide small chunks of input
	* data and process small chunks of output (which may not be equivalent in size).
	*
	* Two subtests - fixed phase (3:2 down) and interpolated phase (147:320 up)
	*/
	TEST(audioflinger_resampler, bufferincrement_fixedphase) {
	// all of these work
	static const enum android::AudioResampler::src_quality kQualityArray[] = {
	android::AudioResampler::LOW_QUALITY,
	android::AudioResampler::MED_QUALITY,
	android::AudioResampler::HIGH_QUALITY,
	android::AudioResampler::VERY_HIGH_QUALITY,
	android::AudioResampler::DYN_LOW_QUALITY,
	android::AudioResampler::DYN_MED_QUALITY,
	android::AudioResampler::DYN_HIGH_QUALITY,
	};

	for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
	testBufferIncrement(2, 48000, 32000, kQualityArray[i]);
	}
	}

	TEST(audioflinger_resampler, bufferincrement_interpolatedphase) {
	// all of these work except low quality
	static const enum android::AudioResampler::src_quality kQualityArray[] = {
	// android::AudioResampler::LOW_QUALITY,
	android::AudioResampler::MED_QUALITY,
	android::AudioResampler::HIGH_QUALITY,
	android::AudioResampler::VERY_HIGH_QUALITY,
	android::AudioResampler::DYN_LOW_QUALITY,
	android::AudioResampler::DYN_MED_QUALITY,
	android::AudioResampler::DYN_HIGH_QUALITY,
	};

	for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
	testBufferIncrement(2, 22050, 48000, kQualityArray[i]);
	}
	}

	/* Simple aliasing test
	*
	* This checks stopband response of the chirp signal to make sure frequencies
	* are properly suppressed. It uses downsampling because the stopband can be
	* clearly isolated by input frequencies exceeding the output sample rate (nyquist).
	*/
	TEST(audioflinger_resampler, stopbandresponse) {
	// not all of these may work (old resamplers fail on downsampling)
	static const enum android::AudioResampler::src_quality kQualityArray[] = {
	//android::AudioResampler::LOW_QUALITY,
	//android::AudioResampler::MED_QUALITY,
	//android::AudioResampler::HIGH_QUALITY,
	//android::AudioResampler::VERY_HIGH_QUALITY,
	android::AudioResampler::DYN_LOW_QUALITY,
	android::AudioResampler::DYN_MED_QUALITY,
	android::AudioResampler::DYN_HIGH_QUALITY,
	};

	// in this test we assume a maximum transition band between 12kHz and 20kHz.
	// there must be at least 60dB relative attenuation between stopband and passband.
	for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
	testStopbandDownconversion(2, 48000, 32000, 12000, 20000, kQualityArray[i]);
	}

	// in this test we assume a maximum transition band between 7kHz and 15kHz.
	// there must be at least 60dB relative attenuation between stopband and passband.
	// (the weird ratio triggers interpolative resampling)
	for (size_t i = 0; i < ARRAY_SIZE(kQualityArray); ++i) {
	testStopbandDownconversion(2, 48000, 22101, 7000, 15000, kQualityArray[i]);
	}
	}