/*
* 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 <errno.h>
#include <fcntl.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/mman.h>
#include <sys/stat.h>
#include <time.h>
#include <unistd.h>
#include <iostream>
#include <memory>
#include <utility>
#include <vector>
#include <gtest/gtest.h>
#include <log/log.h>
#include <media/AudioBufferProvider.h>
#include <media/AudioResampler.h>
#include "../AudioResamplerDyn.h"
#include "../AudioResamplerFirGen.h"
#include "test_utils.h"
template <typename T>
static void printData(T *data, size_t size) {
const size_t stride = 8;
for (size_t i = 0; i < size; ) {
for (size_t j = 0; j < stride && i < size; ++j) {
std::cout << data[i++] << ' '; // extra space before newline
}
std::cout << '\n'; // or endl
}
}
void resample(int channels, 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;
}
size_t framesResampled = resampler->resample(
(int32_t*) output + channels*i, thisFrames, provider);
// we should have enough buffer space, so there is no short count.
ASSERT_EQ(thisFrames, framesResampled);
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 %zu", i);
ASSERT_EQ(check, 0); /* fails */
}
}
}
void testBufferIncrement(size_t channels, bool useFloat,
unsigned inputFreq, unsigned outputFreq,
enum android::AudioResampler::src_quality quality)
{
const audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
// create the provider
std::vector<int> inputIncr;
SignalProvider provider;
if (useFloat) {
provider.setChirp<float>(channels,
0., outputFreq/2., outputFreq, outputFreq/2000.);
} else {
provider.setChirp<int16_t>(channels,
0., outputFreq/2., outputFreq, outputFreq/2000.);
}
provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
size_t outputFrameSize = (channels == 1 ? 2 : channels) * (useFloat ? sizeof(float) : sizeof(int32_t));
size_t outputSize = outputFrameSize * outputFrames;
outputSize &= ~7;
// create the resampler
android::AudioResampler* resampler;
resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
android::AudioResampler::UNITY_GAIN_FLOAT);
// set up the reference run
std::vector<size_t> refIncr;
refIncr.push_back(outputFrames);
void* reference = calloc(outputFrames, outputFrameSize);
resample(channels, 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(format, channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
android::AudioResampler::UNITY_GAIN_FLOAT);
#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 = calloc(outputFrames, outputFrameSize);
inputIncr.push_back(1);
inputIncr.push_back(3);
provider.setIncr(inputIncr);
resample(channels, 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;
}
// TI = resampler input type, int16_t or float
// TO = resampler output type, int32_t or float
template <typename TI, typename TO>
void testStopbandDownconversion(size_t channels,
unsigned inputFreq, unsigned outputFreq,
unsigned passband, unsigned stopband,
enum android::AudioResampler::src_quality quality)
{
// create the provider
std::vector<int> inputIncr;
SignalProvider provider;
provider.setChirp<TI>(channels,
0., inputFreq/2., inputFreq, inputFreq/2000.);
provider.setIncr(inputIncr);
// calculate the output size
size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
size_t outputFrameSize = (channels == 1 ? 2 : channels) * sizeof(TO);
size_t outputSize = outputFrameSize * outputFrames;
outputSize &= ~7;
// create the resampler
android::AudioResampler* resampler;
resampler = android::AudioResampler::create(
is_same<TI, int16_t>::value ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_FLOAT,
channels, outputFreq, quality);
resampler->setSampleRate(inputFreq);
resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
android::AudioResampler::UNITY_GAIN_FLOAT);
// set up the reference run
std::vector<size_t> refIncr;
refIncr.push_back(outputFrames);
void* reference = calloc(outputFrames, outputFrameSize);
resample(channels, reference, outputFrames, refIncr, &provider, resampler);
TO *out = reinterpret_cast<TO *>(reference);
// check signal energy in passband
const unsigned passbandFrame = passband * outputFreq / 1000.;
const unsigned stopbandFrame = stopband * outputFreq / 1000.;
// check each channel separately
if (channels == 1) channels = 2; // workaround (mono duplicates output channel)
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) {
std::cout << out[i+passbandFrame*channels] << std::endl;
}
for (size_t i = 0; i < 10; ++i) {
std::cout << out[i+stopbandFrame*channels] << std::endl;
}
#endif
}
free(reference);
delete resampler;
}
void testFilterResponse(
size_t channels, unsigned inputFreq, unsigned outputFreq)
{
// create resampler
using ResamplerType = android::AudioResamplerDyn<float, float, float>;
std::unique_ptr<ResamplerType> rdyn(
static_cast<ResamplerType *>(
android::AudioResampler::create(
AUDIO_FORMAT_PCM_FLOAT,
channels,
outputFreq,
android::AudioResampler::DYN_HIGH_QUALITY)));
rdyn->setSampleRate(inputFreq);
// get design parameters
const int phases = rdyn->getPhases();
const int halfLength = rdyn->getHalfLength();
const float *coefs = rdyn->getFilterCoefs();
const double fcr = rdyn->getNormalizedCutoffFrequency();
const double tbw = rdyn->getNormalizedTransitionBandwidth();
const double attenuation = rdyn->getFilterAttenuation();
const double stopbandDb = rdyn->getStopbandAttenuationDb();
const double passbandDb = rdyn->getPassbandRippleDb();
const double fp = fcr - tbw / 2;
const double fs = fcr + tbw / 2;
printf("inputFreq:%d outputFreq:%d design"
" phases:%d halfLength:%d"
" fcr:%lf fp:%lf fs:%lf tbw:%lf"
" attenuation:%lf stopRipple:%.lf passRipple:%lf"
"\n",
inputFreq, outputFreq,
phases, halfLength,
fcr, fp, fs, tbw,
attenuation, stopbandDb, passbandDb);
// verify design parameters
constexpr int32_t passSteps = 1000;
double passMin, passMax, passRipple, stopMax, stopRipple;
android::testFir(coefs, phases, halfLength, fp / phases, fs / phases,
passSteps, phases * passSteps /* stopSteps */,
passMin, passMax, passRipple,
stopMax, stopRipple);
printf("inputFreq:%d outputFreq:%d verify"
" passMin:%lf passMax:%lf passRipple:%lf stopMax:%lf stopRipple:%lf"
"\n",
inputFreq, outputFreq,
passMin, passMax, passRipple, stopMax, stopRipple);
ASSERT_GT(stopRipple, 60.); // enough stopband attenuation
ASSERT_LT(passRipple, 0.2); // small passband ripple
ASSERT_GT(passMin, 0.99); // we do not attenuate the signal (ideally 1.)
}
/* 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, false, 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, false, 22050, 48000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, bufferincrement_fixedphase_multi) {
// only dynamic quality
static const enum android::AudioResampler::src_quality kQualityArray[] = {
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(4, false, 48000, 32000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, bufferincrement_interpolatedphase_multi_float) {
// only dynamic quality
static const enum android::AudioResampler::src_quality kQualityArray[] = {
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(8, true, 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_integer) {
// 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<int16_t, int32_t>(
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<int16_t, int32_t>(
2, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, stopbandresponse_integer_mono) {
// 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<int16_t, int32_t>(
1, 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<int16_t, int32_t>(
1, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, stopbandresponse_integer_multichannel) {
// 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<int16_t, int32_t>(
8, 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<int16_t, int32_t>(
8, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, stopbandresponse_float) {
// 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<float, float>(
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<float, float>(
2, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, stopbandresponse_float_mono) {
// 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<float, float>(
1, 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<float, float>(
1, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, stopbandresponse_float_multichannel) {
// 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<float, float>(
8, 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<float, float>(
8, 48000, 22101, 7000, 15000, kQualityArray[i]);
}
}
TEST(audioflinger_resampler, filterresponse) {
std::vector<int> inSampleRates{
8000,
11025,
12000,
16000,
22050,
24000,
32000,
44100,
48000,
88200,
96000,
176400,
192000,
};
std::vector<int> outSampleRates{
48000,
96000,
};
for (int outSampleRate : outSampleRates) {
for (int inSampleRate : inSampleRates) {
testFilterResponse(2 /* channels */, inSampleRate, outSampleRate);
}
}
}