
This CL adds functionality that allows adjusting the audio levels internally in APM. The main purpose of the functionality is to allow APM to optionally be moved to an integration that does not provide an analog gain to control, and the implementation of this has been tailored specifically to meet the requirements for that. More specifically, this CL does -Add a new variant of the pre-amplifier gain that is intended to replace the pre-amplifier gain (but at the moment can coexist with that). The main differences with the pre-amplifier gain is that an attenuating gain is allowed, the gain is applied jointly with any emulated analog gain, and that its packaging fits better with the post gain. -Add an emulation of an analog microphone gain. The emulation is designed to match the analog mic gain functionality in Chrome OS (which is digital) but should be usable also on other platforms. -Add a post-gain which is applied after all processing has been applied. The purpose of this gain is for it to work well with the integration in ChromeOS, and be used to compensate for the offset that there is applied on some USB audio devices. Bug: b/177830918 Change-Id: I0f312996e4088c9bd242a713a703eaaeb17f188a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209707 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33466}
767 lines
32 KiB
C++
767 lines
32 KiB
C++
/*
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* Copyright (c) 2014 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 "modules/audio_processing/audio_processing_impl.h"
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#include <array>
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#include <memory>
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#include "api/scoped_refptr.h"
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#include "modules/audio_processing/common.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/optionally_built_submodule_creators.h"
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#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
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#include "modules/audio_processing/test/echo_canceller_test_tools.h"
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#include "modules/audio_processing/test/echo_control_mock.h"
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#include "modules/audio_processing/test/test_utils.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/random.h"
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#include "rtc_base/ref_counted_object.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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using ::testing::Invoke;
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using ::testing::NotNull;
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class MockInitialize : public AudioProcessingImpl {
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public:
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explicit MockInitialize(const webrtc::Config& config)
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: AudioProcessingImpl(config) {}
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MOCK_METHOD(void, InitializeLocked, (), (override));
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void RealInitializeLocked() RTC_NO_THREAD_SAFETY_ANALYSIS {
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AudioProcessingImpl::InitializeLocked();
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}
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MOCK_METHOD(void, AddRef, (), (const, override));
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MOCK_METHOD(rtc::RefCountReleaseStatus, Release, (), (const, override));
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};
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// Creates MockEchoControl instances and provides a raw pointer access to
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// the next created one. The raw pointer is meant to be used with gmock.
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// Returning a pointer of the next created MockEchoControl instance is necessary
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// for the following reasons: (i) gmock expectations must be set before any call
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// occurs, (ii) APM is initialized the first time that
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// AudioProcessingImpl::ProcessStream() is called and the initialization leads
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// to the creation of a new EchoControl object.
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class MockEchoControlFactory : public EchoControlFactory {
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public:
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MockEchoControlFactory() : next_mock_(std::make_unique<MockEchoControl>()) {}
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// Returns a pointer to the next MockEchoControl that this factory creates.
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MockEchoControl* GetNext() const { return next_mock_.get(); }
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std::unique_ptr<EchoControl> Create(int sample_rate_hz,
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int num_render_channels,
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int num_capture_channels) override {
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std::unique_ptr<EchoControl> mock = std::move(next_mock_);
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next_mock_ = std::make_unique<MockEchoControl>();
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return mock;
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}
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private:
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std::unique_ptr<MockEchoControl> next_mock_;
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};
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// Mocks EchoDetector and records the first samples of the last analyzed render
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// stream frame. Used to check what data is read by an EchoDetector
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// implementation injected into an APM.
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class TestEchoDetector : public EchoDetector {
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public:
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TestEchoDetector()
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: analyze_render_audio_called_(false),
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last_render_audio_first_sample_(0.f) {}
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~TestEchoDetector() override = default;
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void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override {
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last_render_audio_first_sample_ = render_audio[0];
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analyze_render_audio_called_ = true;
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}
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void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
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}
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void Initialize(int capture_sample_rate_hz,
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int num_capture_channels,
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int render_sample_rate_hz,
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int num_render_channels) override {}
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EchoDetector::Metrics GetMetrics() const override { return {}; }
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// Returns true if AnalyzeRenderAudio() has been called at least once.
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bool analyze_render_audio_called() const {
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return analyze_render_audio_called_;
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}
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// Returns the first sample of the last analyzed render frame.
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float last_render_audio_first_sample() const {
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return last_render_audio_first_sample_;
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}
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private:
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bool analyze_render_audio_called_;
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float last_render_audio_first_sample_;
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};
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// Mocks CustomProcessing and applies ProcessSample() to all the samples.
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// Meant to be injected into an APM to modify samples in a known and detectable
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// way.
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class TestRenderPreProcessor : public CustomProcessing {
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public:
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TestRenderPreProcessor() = default;
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~TestRenderPreProcessor() = default;
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void Initialize(int sample_rate_hz, int num_channels) override {}
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void Process(AudioBuffer* audio) override {
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for (size_t k = 0; k < audio->num_channels(); ++k) {
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rtc::ArrayView<float> channel_view(audio->channels()[k],
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audio->num_frames());
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std::transform(channel_view.begin(), channel_view.end(),
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channel_view.begin(), ProcessSample);
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}
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}
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std::string ToString() const override { return "TestRenderPreProcessor"; }
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void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
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// Modifies a sample. This member is used in Process() to modify a frame and
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// it is publicly visible to enable tests.
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static constexpr float ProcessSample(float x) { return 2.f * x; }
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};
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} // namespace
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TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
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webrtc::Config webrtc_config;
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MockInitialize mock(webrtc_config);
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ON_CALL(mock, InitializeLocked())
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.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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mock.Initialize();
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constexpr size_t kMaxSampleRateHz = 32000;
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constexpr size_t kMaxNumChannels = 2;
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std::array<int16_t, kMaxNumChannels * kMaxSampleRateHz / 100> frame;
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frame.fill(0);
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StreamConfig config(16000, 1, /*has_keyboard=*/false);
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// Call with the default parameters; there should be an init.
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EXPECT_CALL(mock, InitializeLocked()).Times(0);
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EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
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EXPECT_NOERR(
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mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
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// New sample rate. (Only impacts ProcessStream).
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config = StreamConfig(32000, 1, /*has_keyboard=*/false);
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
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// New number of channels.
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// TODO(peah): Investigate why this causes 2 inits.
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config = StreamConfig(32000, 2, /*has_keyboard=*/false);
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EXPECT_CALL(mock, InitializeLocked()).Times(2);
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EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
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// ProcessStream sets num_channels_ == num_output_channels.
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EXPECT_NOERR(
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mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
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// A new sample rate passed to ProcessReverseStream should cause an init.
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config = StreamConfig(16000, 2, /*has_keyboard=*/false);
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EXPECT_CALL(mock, InitializeLocked()).Times(1);
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EXPECT_NOERR(
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mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
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}
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TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting().Create());
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webrtc::AudioProcessing::Config apm_config;
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apm_config.pre_amplifier.enabled = true;
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apm_config.pre_amplifier.fixed_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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constexpr int kSampleRateHz = 48000;
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constexpr int16_t kAudioLevel = 10000;
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constexpr size_t kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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EXPECT_EQ(frame[100], kAudioLevel)
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<< "With factor 1, frame shouldn't be modified.";
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constexpr float kGainFactor = 2.f;
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
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// Process for two frames to have time to ramp up gain.
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for (int i = 0; i < 2; ++i) {
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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}
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EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
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<< "Frame should be amplified.";
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}
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TEST(AudioProcessingImplTest,
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LevelAdjustmentUpdateCapturePreGainRuntimeSetting) {
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting().Create());
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webrtc::AudioProcessing::Config apm_config;
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apm_config.capture_level_adjustment.enabled = true;
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apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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constexpr int kSampleRateHz = 48000;
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constexpr int16_t kAudioLevel = 10000;
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constexpr size_t kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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EXPECT_EQ(frame[100], kAudioLevel)
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<< "With factor 1, frame shouldn't be modified.";
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constexpr float kGainFactor = 2.f;
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
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// Process for two frames to have time to ramp up gain.
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for (int i = 0; i < 2; ++i) {
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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}
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EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
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<< "Frame should be amplified.";
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}
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TEST(AudioProcessingImplTest,
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LevelAdjustmentUpdateCapturePostGainRuntimeSetting) {
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting().Create());
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webrtc::AudioProcessing::Config apm_config;
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apm_config.capture_level_adjustment.enabled = true;
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apm_config.capture_level_adjustment.post_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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constexpr int kSampleRateHz = 48000;
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constexpr int16_t kAudioLevel = 10000;
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constexpr size_t kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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EXPECT_EQ(frame[100], kAudioLevel)
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<< "With factor 1, frame shouldn't be modified.";
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constexpr float kGainFactor = 2.f;
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePostGain(kGainFactor));
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// Process for two frames to have time to ramp up gain.
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for (int i = 0; i < 2; ++i) {
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frame.fill(kAudioLevel);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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}
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EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
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<< "Frame should be amplified.";
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}
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TEST(AudioProcessingImplTest, EchoControllerObservesSetCaptureUsageChange) {
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// Tests that the echo controller observes that the capture usage has been
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// updated.
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auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
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const MockEchoControlFactory* echo_control_factory_ptr =
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echo_control_factory.get();
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting()
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.SetEchoControlFactory(std::move(echo_control_factory))
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.Create());
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constexpr int16_t kAudioLevel = 10000;
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constexpr int kSampleRateHz = 48000;
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constexpr int kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
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MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
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// Ensure that SetCaptureOutputUsage is not called when no runtime settings
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// are passed.
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EXPECT_CALL(*echo_control_mock, SetCaptureOutputUsage(testing::_)).Times(0);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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// Ensure that SetCaptureOutputUsage is called with the right information when
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// a runtime setting is passed.
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EXPECT_CALL(*echo_control_mock,
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SetCaptureOutputUsage(/*capture_output_used=*/false))
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.Times(1);
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EXPECT_TRUE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/false)));
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apm->ProcessStream(frame.data(), config, config, frame.data());
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EXPECT_CALL(*echo_control_mock,
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SetCaptureOutputUsage(/*capture_output_used=*/true))
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.Times(1);
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EXPECT_TRUE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/true)));
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apm->ProcessStream(frame.data(), config, config, frame.data());
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// The number of positions to place items in the queue is equal to the queue
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// size minus 1.
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constexpr int kNumSlotsInQueue = RuntimeSettingQueueSize();
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// Ensure that SetCaptureOutputUsage is called with the right information when
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// many runtime settings are passed.
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for (int k = 0; k < kNumSlotsInQueue - 1; ++k) {
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EXPECT_TRUE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/false)));
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}
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EXPECT_CALL(*echo_control_mock,
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SetCaptureOutputUsage(/*capture_output_used=*/false))
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.Times(kNumSlotsInQueue - 1);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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// Ensure that SetCaptureOutputUsage is properly called with the fallback
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// value when the runtime settings queue becomes full.
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for (int k = 0; k < kNumSlotsInQueue; ++k) {
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EXPECT_TRUE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/false)));
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}
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EXPECT_FALSE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/false)));
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EXPECT_FALSE(apm->PostRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
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/*capture_output_used=*/false)));
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EXPECT_CALL(*echo_control_mock,
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SetCaptureOutputUsage(/*capture_output_used=*/false))
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.Times(kNumSlotsInQueue);
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EXPECT_CALL(*echo_control_mock,
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SetCaptureOutputUsage(/*capture_output_used=*/true))
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.Times(1);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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}
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TEST(AudioProcessingImplTest,
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EchoControllerObservesPreAmplifierEchoPathGainChange) {
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// Tests that the echo controller observes an echo path gain change when the
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// pre-amplifier submodule changes the gain.
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auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
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const auto* echo_control_factory_ptr = echo_control_factory.get();
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting()
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.SetEchoControlFactory(std::move(echo_control_factory))
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.Create());
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// Disable AGC.
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webrtc::AudioProcessing::Config apm_config;
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apm_config.gain_controller1.enabled = false;
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apm_config.gain_controller2.enabled = false;
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apm_config.pre_amplifier.enabled = true;
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apm_config.pre_amplifier.fixed_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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constexpr int16_t kAudioLevel = 10000;
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constexpr size_t kSampleRateHz = 48000;
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constexpr size_t kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
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MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
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EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
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EXPECT_CALL(*echo_control_mock,
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ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
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.Times(1);
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apm->ProcessStream(frame.data(), config, config, frame.data());
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EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
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EXPECT_CALL(*echo_control_mock,
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ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
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.Times(1);
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apm->SetRuntimeSetting(
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AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
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apm->ProcessStream(frame.data(), config, config, frame.data());
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}
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TEST(AudioProcessingImplTest,
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EchoControllerObservesLevelAdjustmentPreGainEchoPathGainChange) {
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// Tests that the echo controller observes an echo path gain change when the
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// pre-amplifier submodule changes the gain.
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auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
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const auto* echo_control_factory_ptr = echo_control_factory.get();
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std::unique_ptr<AudioProcessing> apm(
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AudioProcessingBuilderForTesting()
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.SetEchoControlFactory(std::move(echo_control_factory))
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.Create());
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// Disable AGC.
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webrtc::AudioProcessing::Config apm_config;
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apm_config.gain_controller1.enabled = false;
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apm_config.gain_controller2.enabled = false;
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apm_config.capture_level_adjustment.enabled = true;
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apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
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apm->ApplyConfig(apm_config);
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constexpr int16_t kAudioLevel = 10000;
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constexpr size_t kSampleRateHz = 48000;
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constexpr size_t kNumChannels = 2;
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std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
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StreamConfig config(kSampleRateHz, kNumChannels, /*has_keyboard=*/false);
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frame.fill(kAudioLevel);
|
|
|
|
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
|
|
.Times(1);
|
|
apm->ProcessStream(frame.data(), config, config, frame.data());
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
|
|
.Times(1);
|
|
apm->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
|
|
apm->ProcessStream(frame.data(), config, config, frame.data());
|
|
}
|
|
|
|
TEST(AudioProcessingImplTest,
|
|
EchoControllerObservesAnalogAgc1EchoPathGainChange) {
|
|
// Tests that the echo controller observes an echo path gain change when the
|
|
// AGC1 analog adaptive submodule changes the analog gain.
|
|
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
|
|
const auto* echo_control_factory_ptr = echo_control_factory.get();
|
|
|
|
std::unique_ptr<AudioProcessing> apm(
|
|
AudioProcessingBuilderForTesting()
|
|
.SetEchoControlFactory(std::move(echo_control_factory))
|
|
.Create());
|
|
webrtc::AudioProcessing::Config apm_config;
|
|
// Enable AGC1.
|
|
apm_config.gain_controller1.enabled = true;
|
|
apm_config.gain_controller1.mode =
|
|
AudioProcessing::Config::GainController1::kAdaptiveAnalog;
|
|
apm_config.gain_controller2.enabled = false;
|
|
apm_config.pre_amplifier.enabled = false;
|
|
apm->ApplyConfig(apm_config);
|
|
|
|
constexpr int16_t kAudioLevel = 1000;
|
|
constexpr size_t kSampleRateHz = 48000;
|
|
constexpr size_t kNumChannels = 2;
|
|
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
|
|
StreamConfig stream_config(kSampleRateHz, kNumChannels,
|
|
/*has_keyboard=*/false);
|
|
frame.fill(kAudioLevel);
|
|
|
|
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
|
|
|
|
const int initial_analog_gain = apm->recommended_stream_analog_level();
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, false))
|
|
.Times(1);
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
|
|
// Force an analog gain change if it did not happen.
|
|
if (initial_analog_gain == apm->recommended_stream_analog_level()) {
|
|
apm->set_stream_analog_level(initial_analog_gain + 1);
|
|
}
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock, ProcessCapture(NotNull(), testing::_, true))
|
|
.Times(1);
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
}
|
|
|
|
TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) {
|
|
// Tests that the echo controller observes an echo path gain change when a
|
|
// playout volume change is reported.
|
|
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
|
|
const auto* echo_control_factory_ptr = echo_control_factory.get();
|
|
|
|
std::unique_ptr<AudioProcessing> apm(
|
|
AudioProcessingBuilderForTesting()
|
|
.SetEchoControlFactory(std::move(echo_control_factory))
|
|
.Create());
|
|
// Disable AGC.
|
|
webrtc::AudioProcessing::Config apm_config;
|
|
apm_config.gain_controller1.enabled = false;
|
|
apm_config.gain_controller2.enabled = false;
|
|
apm->ApplyConfig(apm_config);
|
|
|
|
constexpr int16_t kAudioLevel = 10000;
|
|
constexpr size_t kSampleRateHz = 48000;
|
|
constexpr size_t kNumChannels = 2;
|
|
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
|
|
StreamConfig stream_config(kSampleRateHz, kNumChannels,
|
|
/*has_keyboard=*/false);
|
|
frame.fill(kAudioLevel);
|
|
|
|
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
|
|
.Times(1);
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
|
|
.Times(1);
|
|
apm->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
|
|
.Times(1);
|
|
apm->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
|
|
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
|
|
EXPECT_CALL(*echo_control_mock,
|
|
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
|
|
.Times(1);
|
|
apm->SetRuntimeSetting(
|
|
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100));
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
|
}
|
|
|
|
TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
|
|
// Make sure that signal changes caused by a render pre-processing sub-module
|
|
// take place before any echo detector analysis.
|
|
rtc::scoped_refptr<TestEchoDetector> test_echo_detector(
|
|
new rtc::RefCountedObject<TestEchoDetector>());
|
|
std::unique_ptr<CustomProcessing> test_render_pre_processor(
|
|
new TestRenderPreProcessor());
|
|
// Create APM injecting the test echo detector and render pre-processor.
|
|
std::unique_ptr<AudioProcessing> apm(
|
|
AudioProcessingBuilderForTesting()
|
|
.SetEchoDetector(test_echo_detector)
|
|
.SetRenderPreProcessing(std::move(test_render_pre_processor))
|
|
.Create());
|
|
webrtc::AudioProcessing::Config apm_config;
|
|
apm_config.pre_amplifier.enabled = true;
|
|
apm_config.residual_echo_detector.enabled = true;
|
|
apm->ApplyConfig(apm_config);
|
|
|
|
constexpr int16_t kAudioLevel = 1000;
|
|
constexpr int kSampleRateHz = 16000;
|
|
constexpr size_t kNumChannels = 1;
|
|
// Explicitly initialize APM to ensure no render frames are discarded.
|
|
const ProcessingConfig processing_config = {{
|
|
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
|
|
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
|
|
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
|
|
{kSampleRateHz, kNumChannels, /*has_keyboard=*/false},
|
|
}};
|
|
apm->Initialize(processing_config);
|
|
|
|
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
|
|
StreamConfig stream_config(kSampleRateHz, kNumChannels,
|
|
/*has_keyboard=*/false);
|
|
|
|
constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel);
|
|
constexpr float kExpectedPreprocessedAudioLevel =
|
|
TestRenderPreProcessor::ProcessSample(kAudioLevelFloat);
|
|
ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel);
|
|
|
|
// Analyze a render stream frame.
|
|
frame.fill(kAudioLevel);
|
|
ASSERT_EQ(AudioProcessing::Error::kNoError,
|
|
apm->ProcessReverseStream(frame.data(), stream_config,
|
|
stream_config, frame.data()));
|
|
// Trigger a call to in EchoDetector::AnalyzeRenderAudio() via
|
|
// ProcessStream().
|
|
frame.fill(kAudioLevel);
|
|
ASSERT_EQ(AudioProcessing::Error::kNoError,
|
|
apm->ProcessStream(frame.data(), stream_config, stream_config,
|
|
frame.data()));
|
|
// Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is
|
|
// triggered, the line below checks that the call has occurred. If not, the
|
|
// APM implementation may have changed and this test might need to be adapted.
|
|
ASSERT_TRUE(test_echo_detector->analyze_render_audio_called());
|
|
// Check that the data read in EchoDetector::AnalyzeRenderAudio() is that
|
|
// produced by the render pre-processor.
|
|
EXPECT_EQ(kExpectedPreprocessedAudioLevel,
|
|
test_echo_detector->last_render_audio_first_sample());
|
|
}
|
|
|
|
// Disabling build-optional submodules and trying to enable them via the APM
|
|
// config should be bit-exact with running APM with said submodules disabled.
|
|
// This mainly tests that SetCreateOptionalSubmodulesForTesting has an effect.
|
|
TEST(ApmWithSubmodulesExcludedTest, BitexactWithDisabledModules) {
|
|
rtc::scoped_refptr<AudioProcessingImpl> apm =
|
|
new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config());
|
|
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
|
|
|
|
ApmSubmoduleCreationOverrides overrides;
|
|
overrides.transient_suppression = true;
|
|
apm->OverrideSubmoduleCreationForTesting(overrides);
|
|
|
|
AudioProcessing::Config apm_config = apm->GetConfig();
|
|
apm_config.transient_suppression.enabled = true;
|
|
apm->ApplyConfig(apm_config);
|
|
|
|
rtc::scoped_refptr<AudioProcessing> apm_reference =
|
|
AudioProcessingBuilder().Create();
|
|
apm_config = apm_reference->GetConfig();
|
|
apm_config.transient_suppression.enabled = false;
|
|
apm_reference->ApplyConfig(apm_config);
|
|
|
|
constexpr int kSampleRateHz = 16000;
|
|
constexpr int kNumChannels = 1;
|
|
std::array<float, kSampleRateHz / 100> buffer;
|
|
std::array<float, kSampleRateHz / 100> buffer_reference;
|
|
float* channel_pointers[] = {buffer.data()};
|
|
float* channel_pointers_reference[] = {buffer_reference.data()};
|
|
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
|
|
/*num_channels=*/kNumChannels,
|
|
/*has_keyboard=*/false);
|
|
Random random_generator(2341U);
|
|
constexpr int kFramesToProcessPerConfiguration = 10;
|
|
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
std::copy(buffer.begin(), buffer.end(), buffer_reference.begin());
|
|
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
|
channel_pointers),
|
|
kNoErr);
|
|
ASSERT_EQ(
|
|
apm_reference->ProcessStream(channel_pointers_reference, stream_config,
|
|
stream_config, channel_pointers_reference),
|
|
kNoErr);
|
|
for (int j = 0; j < kSampleRateHz / 100; ++j) {
|
|
EXPECT_EQ(buffer[j], buffer_reference[j]);
|
|
}
|
|
}
|
|
}
|
|
|
|
// Disable transient suppressor creation and run APM in ways that should trigger
|
|
// calls to the transient suppressor API.
|
|
TEST(ApmWithSubmodulesExcludedTest, ReinitializeTransientSuppressor) {
|
|
rtc::scoped_refptr<AudioProcessingImpl> apm =
|
|
new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config());
|
|
ASSERT_EQ(apm->Initialize(), kNoErr);
|
|
|
|
ApmSubmoduleCreationOverrides overrides;
|
|
overrides.transient_suppression = true;
|
|
apm->OverrideSubmoduleCreationForTesting(overrides);
|
|
|
|
AudioProcessing::Config config = apm->GetConfig();
|
|
config.transient_suppression.enabled = true;
|
|
apm->ApplyConfig(config);
|
|
// 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
|
|
float buffer[960];
|
|
float* channel_pointers[] = {&buffer[0], &buffer[480]};
|
|
Random random_generator(2341U);
|
|
constexpr int kFramesToProcessPerConfiguration = 3;
|
|
|
|
StreamConfig initial_stream_config(/*sample_rate_hz=*/16000,
|
|
/*num_channels=*/1,
|
|
/*has_keyboard=*/false);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, initial_stream_config,
|
|
initial_stream_config, channel_pointers),
|
|
kNoErr);
|
|
}
|
|
|
|
StreamConfig stereo_stream_config(/*sample_rate_hz=*/16000,
|
|
/*num_channels=*/2,
|
|
/*has_keyboard=*/false);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, stereo_stream_config,
|
|
stereo_stream_config, channel_pointers),
|
|
kNoErr);
|
|
}
|
|
|
|
StreamConfig high_sample_rate_stream_config(/*sample_rate_hz=*/48000,
|
|
/*num_channels=*/1,
|
|
/*has_keyboard=*/false);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(
|
|
apm->ProcessStream(channel_pointers, high_sample_rate_stream_config,
|
|
high_sample_rate_stream_config, channel_pointers),
|
|
kNoErr);
|
|
}
|
|
|
|
StreamConfig keyboard_stream_config(/*sample_rate_hz=*/16000,
|
|
/*num_channels=*/1,
|
|
/*has_keyboard=*/true);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, keyboard_stream_config,
|
|
keyboard_stream_config, channel_pointers),
|
|
kNoErr);
|
|
}
|
|
}
|
|
|
|
// Disable transient suppressor creation and run APM in ways that should trigger
|
|
// calls to the transient suppressor API.
|
|
TEST(ApmWithSubmodulesExcludedTest, ToggleTransientSuppressor) {
|
|
rtc::scoped_refptr<AudioProcessingImpl> apm =
|
|
new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config());
|
|
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
|
|
|
|
ApmSubmoduleCreationOverrides overrides;
|
|
overrides.transient_suppression = true;
|
|
apm->OverrideSubmoduleCreationForTesting(overrides);
|
|
|
|
// 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
|
|
float buffer[960];
|
|
float* channel_pointers[] = {&buffer[0], &buffer[480]};
|
|
Random random_generator(2341U);
|
|
constexpr int kFramesToProcessPerConfiguration = 3;
|
|
StreamConfig stream_config(/*sample_rate_hz=*/16000,
|
|
/*num_channels=*/1,
|
|
/*has_keyboard=*/false);
|
|
|
|
AudioProcessing::Config config = apm->GetConfig();
|
|
config.transient_suppression.enabled = true;
|
|
apm->ApplyConfig(config);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
|
channel_pointers),
|
|
kNoErr);
|
|
}
|
|
|
|
config = apm->GetConfig();
|
|
config.transient_suppression.enabled = false;
|
|
apm->ApplyConfig(config);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
|
channel_pointers),
|
|
kNoErr);
|
|
}
|
|
|
|
config = apm->GetConfig();
|
|
config.transient_suppression.enabled = true;
|
|
apm->ApplyConfig(config);
|
|
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
|
|
RandomizeSampleVector(&random_generator, buffer);
|
|
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
|
channel_pointers),
|
|
kNoErr);
|
|
}
|
|
}
|
|
} // namespace webrtc
|