/* * 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 (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