92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
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 "api/array_view.h"
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#include "modules/audio_processing/agc2/digital_gain_applier.h"
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#include "modules/audio_processing/agc2/gain_controller2.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "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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