3ec96df907
split domain and only implements floating point operations (to avoid spectral leakage issues and unnecessary complexity). The goal of this CL is adding the new sub-module into APM without providing an implementation that could replace the existing gain control modules. The focus is in fact on initialization, reset, and configuration of AGC2. The module itself only applies a hard-coded gain value. This behavior will change in the coming CLs. BUG=webrtc:7494 Review-Url: https://codereview.webrtc.org/2848593002 Cr-Commit-Position: refs/heads/master@{#18222}
100 lines
3.1 KiB
C++
100 lines
3.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include <string>
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#include "webrtc/base/array_view.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/agc2/gain_controller2.h"
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#include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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namespace test {
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namespace {
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constexpr size_t kNumFrames = 480u;
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constexpr size_t kStereo = 2u;
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void SetAudioBufferSamples(float value, AudioBuffer* ab) {
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for (size_t k = 0; k < ab->num_channels(); ++k) {
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auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
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for (auto& sample : channel) { sample = value; }
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}
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}
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template<typename Functor>
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bool CheckAudioBufferSamples(Functor validator, AudioBuffer* ab) {
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for (size_t k = 0; k < ab->num_channels(); ++k) {
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auto channel = rtc::ArrayView<float>(ab->channels_f()[k], ab->num_frames());
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for (auto& sample : channel) { if (!validator(sample)) { return false; } }
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}
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return true;
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}
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bool TestDigitalGainApplier(float sample_value, float gain, float expected) {
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AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
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SetAudioBufferSamples(sample_value, &ab);
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DigitalGainApplier gain_applier;
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for (size_t k = 0; k < ab.num_channels(); ++k) {
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auto channel_view = rtc::ArrayView<float>(
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ab.channels_f()[k], ab.num_frames());
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gain_applier.Process(gain, channel_view);
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}
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auto check_expectation = [expected](float sample) {
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return sample == expected; };
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return CheckAudioBufferSamples(check_expectation, &ab);
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}
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} // namespace
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TEST(GainController2, Instance) {
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std::unique_ptr<GainController2> gain_controller2;
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gain_controller2.reset(new GainController2(
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AudioProcessing::kSampleRate48kHz));
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}
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TEST(GainController2, ToString) {
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AudioProcessing::Config config;
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config.gain_controller2.enabled = false;
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EXPECT_EQ("{enabled: false}",
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GainController2::ToString(config.gain_controller2));
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config.gain_controller2.enabled = true;
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EXPECT_EQ("{enabled: true}",
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GainController2::ToString(config.gain_controller2));
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}
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TEST(GainController2, DigitalGainApplierProcess) {
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EXPECT_TRUE(TestDigitalGainApplier(1000.0f, 0.5, 500.0f));
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}
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TEST(GainController2, DigitalGainApplierCheckClipping) {
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EXPECT_TRUE(TestDigitalGainApplier(30000.0f, 1.5, 32767.0f));
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EXPECT_TRUE(TestDigitalGainApplier(-30000.0f, 1.5, -32767.0f));
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}
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TEST(GainController2, Usage) {
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std::unique_ptr<GainController2> gain_controller2;
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gain_controller2.reset(new GainController2(
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AudioProcessing::kSampleRate48kHz));
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AudioBuffer ab(kNumFrames, kStereo, kNumFrames, kStereo, kNumFrames);
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SetAudioBufferSamples(1000.0f, &ab);
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gain_controller2->Process(&ab);
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}
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} // namespace test
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} // namespace webrtc
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