Add tests directory for AudioFlinger
Native tests for audio resampler through gtest framework.
Change-Id: Ia340ca51f9b9b78d67e03e3541e4462b98578c95
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
new file mode 100644
index 0000000..8f9c270
--- /dev/null
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -0,0 +1,471 @@
+/*
+ * 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]);
+ }
+}