1 files changed, 411 insertions, 0 deletions
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
new file mode 100644
index 0000000..d6217ba
--- /dev/null
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -0,0 +1,411 @@
+/*
+ * 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 <iostream>
+#include <cutils/log.h>
+#include <gtest/gtest.h>
+#include <media/AudioBufferProvider.h>
+#include "AudioResampler.h"
+#include "test_utils.h"
+
+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;
+        }
+        resampler->resample((int32_t*) output + channels*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 %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 * (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 = malloc(outputSize);
+    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 = malloc(outputSize);
+    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 * 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 = malloc(outputSize);
+    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
+    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;
+}
+
+/* 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_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_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]);
+    }
+}
+