blob: faa8d169e684852ee3c3702ae8726f49cb90a954 [file] [log] [blame]
/*
* Copyright (c) 2017 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 <memory>
#include <string>
#include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h"
#include "webrtc/modules/audio_processing/agc2/gain_controller2.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/rtc_base/array_view.h"
#include "webrtc/test/gtest.h"
namespace webrtc {
namespace test {
namespace {
constexpr size_t kNumFrames = 480u;
constexpr size_t kStereo = 2u;
void SetAudioBufferSamples(float value, AudioBuffer* ab) {
for (size_t k = 0; k < ab->num_channels(); ++k) {
auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
for (auto& sample : channel) { sample = value; }
}
}
template<typename Functor>
bool CheckAudioBufferSamples(Functor validator, AudioBuffer* ab) {
for (size_t k = 0; k < ab->num_channels(); ++k) {
auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
for (auto& sample : channel) { if (!validator(sample)) { return false; } }
}
return true;
}
bool TestDigitalGainApplier(float sample_value, float gain, float expected) {
AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
SetAudioBufferSamples(sample_value, &ab);
DigitalGainApplier gain_applier;
for (size_t k = 0; k < ab.num_channels(); ++k) {
auto channel_view = rtc::ArrayView<float>(
ab.channels_f()[k], ab.num_frames());
gain_applier.Process(gain, channel_view);
}
auto check_expectation = [expected](float sample) {
return sample == expected; };
return CheckAudioBufferSamples(check_expectation, &ab);
}
} // namespace
TEST(GainController2, Instance) {
std::unique_ptr<GainController2> gain_controller2;
gain_controller2.reset(new GainController2(
AudioProcessing::kSampleRate48kHz));
}
TEST(GainController2, ToString) {
AudioProcessing::Config config;
config.gain_controller2.enabled = false;
EXPECT_EQ("{enabled: false}",
GainController2::ToString(config.gain_controller2));
config.gain_controller2.enabled = true;
EXPECT_EQ("{enabled: true}",
GainController2::ToString(config.gain_controller2));
}
TEST(GainController2, DigitalGainApplierProcess) {
EXPECT_TRUE(TestDigitalGainApplier(1000.0f, 0.5, 500.0f));
}
TEST(GainController2, DigitalGainApplierCheckClipping) {
EXPECT_TRUE(TestDigitalGainApplier(30000.0f, 1.5, 32767.0f));
EXPECT_TRUE(TestDigitalGainApplier(-30000.0f, 1.5, -32767.0f));
}
TEST(GainController2, Usage) {
std::unique_ptr<GainController2> gain_controller2;
gain_controller2.reset(new GainController2(
AudioProcessing::kSampleRate48kHz));
AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
SetAudioBufferSamples(1000.0f, &ab);
gain_controller2->Process(&ab);
}
} // namespace test
} // namespace webrtc