/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "gtest/gtest.h"
#include "common_audio/resampler/include/resampler.h"
// TODO(andrew): this is a work-in-progress. Many more tests are needed.
namespace webrtc {
namespace {
const ResamplerType kTypes[] = {
kResamplerSynchronous,
kResamplerAsynchronous,
kResamplerSynchronousStereo,
kResamplerAsynchronousStereo
// kResamplerInvalid excluded
};
const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
// Rates we must support.
const int kMaxRate = 96000;
const int kRates[] = {
8000,
16000,
32000,
44000,
48000,
kMaxRate
};
const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
const int kMaxChannels = 2;
const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
// TODO(andrew): should we be supporting these combinations?
bool ValidRates(int in_rate, int out_rate) {
// Not the most compact notation, for clarity.
if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
(out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
return false;
}
return true;
}
class ResamplerTest : public testing::Test {
protected:
ResamplerTest();
virtual void SetUp();
virtual void TearDown();
Resampler rs_;
int16_t data_in_[kDataSize];
int16_t data_out_[kDataSize];
};
ResamplerTest::ResamplerTest() {}
void ResamplerTest::SetUp() {
// Initialize input data with anything. The tests are content independent.
memset(data_in_, 1, sizeof(data_in_));
}
void ResamplerTest::TearDown() {}
TEST_F(ResamplerTest, Reset) {
// The only failure mode for the constructor is if Reset() fails. For the
// time being then (until an Init function is added), we rely on Reset()
// to test the constructor.
// Check that all required combinations are supported.
for (size_t i = 0; i < kRatesSize; ++i) {
for (size_t j = 0; j < kRatesSize; ++j) {
for (size_t k = 0; k < kTypesSize; ++k) {
std::ostringstream ss;
ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
<< ", type: " << kTypes[k];
SCOPED_TRACE(ss.str());
if (ValidRates(kRates[i], kRates[j]))
EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
else
EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
}
}
}
}
// TODO(tlegrand): Replace code inside the two tests below with a function
// with number of channels and ResamplerType as input.
TEST_F(ResamplerTest, Synchronous) {
for (size_t i = 0; i < kRatesSize; ++i) {
for (size_t j = 0; j < kRatesSize; ++j) {
std::ostringstream ss;
ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
SCOPED_TRACE(ss.str());
if (ValidRates(kRates[i], kRates[j])) {
int in_length = kRates[i] / 100;
int out_length = 0;
EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
out_length));
EXPECT_EQ(kRates[j] / 100, out_length);
} else {
EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
}
}
}
}
TEST_F(ResamplerTest, SynchronousStereo) {
// Number of channels is 2, stereo mode.
const int kChannels = 2;
for (size_t i = 0; i < kRatesSize; ++i) {
for (size_t j = 0; j < kRatesSize; ++j) {
std::ostringstream ss;
ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
SCOPED_TRACE(ss.str());
if (ValidRates(kRates[i], kRates[j])) {
int in_length = kChannels * kRates[i] / 100;
int out_length = 0;
EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
kResamplerSynchronousStereo));
EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
out_length));
EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
} else {
EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
kResamplerSynchronousStereo));
}
}
}
}
} // namespace
} // namespace webrtc