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The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
98 lines
3.3 KiB
C++
98 lines
3.3 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 <algorithm>
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#include "api/array_view.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/gain_controller2.h"
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#include "rtc_base/checks.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 kFrameSizeMs = 10u;
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constexpr size_t kStereo = 2u;
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void SetAudioBufferSamples(float value, AudioBuffer* ab) {
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// Sets all the samples in |ab| to |value|.
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for (size_t k = 0; k < ab->num_channels(); ++k) {
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std::fill(ab->channels_f()[k], ab->channels_f()[k] + ab->num_frames(),
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value);
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}
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}
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} // namespace
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TEST(GainController2, CreateApplyConfig) {
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// Instances GainController2 and applies different configurations.
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std::unique_ptr<GainController2> gain_controller2(new GainController2());
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// Check that the default config is valid.
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AudioProcessing::Config::GainController2 config;
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EXPECT_TRUE(GainController2::Validate(config));
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gain_controller2->ApplyConfig(config);
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// Check that attenuation is not allowed.
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config.fixed_gain_db = -5.f;
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EXPECT_FALSE(GainController2::Validate(config));
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// Check that valid configurations are applied.
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for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 50.f}) {
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config.fixed_gain_db = fixed_gain_db;
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EXPECT_TRUE(GainController2::Validate(config));
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gain_controller2->ApplyConfig(config);
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}
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}
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TEST(GainController2, ToString) {
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// Tests GainController2::ToString().
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AudioProcessing::Config::GainController2 config;
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config.fixed_gain_db = 5.f;
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config.enabled = false;
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config.enable_limiter = true;
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EXPECT_EQ("{enabled: false, fixed_gain_dB: 5, enable_limiter: true}",
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GainController2::ToString(config));
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config.enabled = true;
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EXPECT_EQ("{enabled: true, fixed_gain_dB: 5, enable_limiter: true}",
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GainController2::ToString(config));
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}
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TEST(GainController2, Usage) {
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// Tests GainController2::Process() on an AudioBuffer instance.
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std::unique_ptr<GainController2> gain_controller2(new GainController2());
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gain_controller2->Initialize(AudioProcessing::kSampleRate48kHz);
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const size_t num_frames = rtc::CheckedDivExact<size_t>(
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kFrameSizeMs * AudioProcessing::kSampleRate48kHz, 1000);
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AudioBuffer ab(num_frames, kStereo, num_frames, kStereo, num_frames);
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constexpr float sample_value = 1000.f;
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SetAudioBufferSamples(sample_value, &ab);
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AudioProcessing::Config::GainController2 config;
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// Check that samples are not modified when the fixed gain is 0 dB.
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config.fixed_gain_db = 0.f;
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gain_controller2->ApplyConfig(config);
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gain_controller2->Process(&ab);
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EXPECT_EQ(ab.channels_f()[0][0], sample_value);
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// Check that samples are amplified when the fixed gain is greater than 0 dB.
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config.fixed_gain_db = 5.f;
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gain_controller2->ApplyConfig(config);
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gain_controller2->Process(&ab);
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EXPECT_LT(sample_value, ab.channels_f()[0][0]);
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}
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} // namespace test
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} // namespace webrtc
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