AudioProcessingImpl: Add a VAD submodule
Add a VoiceActivityDetectorWrapper submodule in AudioProcessingImpl and enable injecting speech probability into GainController2. Bug: webrtc:13663 Change-Id: I05e13b737d085b45ac8ce76660191867c56834c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265166 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37275}
This commit is contained in:
Hanna Silen
WebRTC LUCI CQ
parent
ff45105b42
commit
0c1ad2992b
@ -401,6 +401,7 @@ if (rtc_include_tests) {
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"../../rtc_base/system:file_wrapper",
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"../../system_wrappers",
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"../../system_wrappers:denormal_disabler",
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"../../test:field_trial",
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"../../test:fileutils",
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"../../test:rtc_expect_death",
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"../../test:test_support",
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@ -162,6 +162,7 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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bool noise_suppressor_enabled,
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bool adaptive_gain_controller_enabled,
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bool gain_controller2_enabled,
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bool voice_activity_detector_enabled,
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bool gain_adjustment_enabled,
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bool echo_controller_enabled,
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bool transient_suppressor_enabled) {
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@ -173,6 +174,8 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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changed |=
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(adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
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changed |= (gain_controller2_enabled != gain_controller2_enabled_);
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changed |=
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(voice_activity_detector_enabled != voice_activity_detector_enabled_);
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changed |= (gain_adjustment_enabled != gain_adjustment_enabled_);
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changed |= (echo_controller_enabled != echo_controller_enabled_);
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changed |= (transient_suppressor_enabled != transient_suppressor_enabled_);
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@ -182,6 +185,7 @@ bool AudioProcessingImpl::SubmoduleStates::Update(
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noise_suppressor_enabled_ = noise_suppressor_enabled;
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adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
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gain_controller2_enabled_ = gain_controller2_enabled;
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voice_activity_detector_enabled_ = voice_activity_detector_enabled;
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gain_adjustment_enabled_ = gain_adjustment_enabled;
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echo_controller_enabled_ = echo_controller_enabled;
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transient_suppressor_enabled_ = transient_suppressor_enabled;
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@ -395,6 +399,7 @@ void AudioProcessingImpl::InitializeLocked() {
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InitializeResidualEchoDetector();
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InitializeEchoController();
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InitializeGainController2(/*config_has_changed=*/true);
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InitializeVoiceActivityDetector(/*config_has_changed=*/true);
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InitializeNoiseSuppressor();
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InitializeAnalyzer();
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InitializePostProcessor();
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@ -569,6 +574,7 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
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}
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InitializeGainController2(agc2_config_changed);
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InitializeVoiceActivityDetector(agc2_config_changed);
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if (pre_amplifier_config_changed || gain_adjustment_config_changed) {
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InitializeCaptureLevelsAdjuster();
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@ -1297,10 +1303,19 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
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submodules_.capture_analyzer->Analyze(capture_buffer);
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}
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absl::optional<float> voice_activity_probability = absl::nullopt;
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if (submodules_.gain_controller2) {
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submodules_.gain_controller2->NotifyAnalogLevel(
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recommended_stream_analog_level_locked());
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submodules_.gain_controller2->Process(capture_buffer);
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if (submodules_.voice_activity_detector) {
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voice_activity_probability =
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submodules_.voice_activity_detector->Analyze(
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AudioFrameView<const float>(capture_buffer->channels(),
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capture_buffer->num_channels(),
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capture_buffer->num_frames()));
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}
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submodules_.gain_controller2->Process(voice_activity_probability,
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capture_buffer);
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}
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if (submodules_.capture_post_processor) {
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@ -1692,7 +1707,7 @@ bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
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return submodule_states_.Update(
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config_.high_pass_filter.enabled, !!submodules_.echo_control_mobile,
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!!submodules_.noise_suppressor, !!submodules_.gain_control,
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!!submodules_.gain_controller2,
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!!submodules_.gain_controller2, !!submodules_.voice_activity_detector,
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config_.pre_amplifier.enabled || config_.capture_level_adjustment.enabled,
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capture_nonlocked_.echo_controller_enabled,
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!!submodules_.transient_suppressor);
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@ -1900,9 +1915,35 @@ void AudioProcessingImpl::InitializeGainController2(bool config_has_changed) {
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return;
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}
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if (!submodules_.gain_controller2 || config_has_changed) {
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const bool use_internal_vad =
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transient_suppressor_vad_mode_ != TransientSuppressor::VadMode::kRnnVad;
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submodules_.gain_controller2 = std::make_unique<GainController2>(
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config_.gain_controller2, proc_fullband_sample_rate_hz(),
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num_input_channels());
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num_input_channels(), use_internal_vad);
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}
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}
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void AudioProcessingImpl::InitializeVoiceActivityDetector(
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bool config_has_changed) {
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if (!config_has_changed) {
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return;
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}
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const bool use_vad =
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transient_suppressor_vad_mode_ == TransientSuppressor::VadMode::kRnnVad &&
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config_.gain_controller2.enabled &&
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config_.gain_controller2.adaptive_digital.enabled;
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if (!use_vad) {
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submodules_.voice_activity_detector.reset();
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return;
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}
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if (!submodules_.voice_activity_detector || config_has_changed) {
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RTC_DCHECK(!!submodules_.gain_controller2);
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// TODO(bugs.webrtc.org/13663): Cache CPU features in APM and use here.
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submodules_.voice_activity_detector =
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std::make_unique<VoiceActivityDetectorWrapper>(
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config_.gain_controller2.adaptive_digital.vad_reset_period_ms,
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submodules_.gain_controller2->GetCpuFeatures(),
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proc_fullband_sample_rate_hz());
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}
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}
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@ -207,6 +207,7 @@ class AudioProcessingImpl : public AudioProcessing {
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bool noise_suppressor_enabled,
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bool adaptive_gain_controller_enabled,
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bool gain_controller2_enabled,
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bool voice_activity_detector_enabled,
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bool gain_adjustment_enabled,
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bool echo_controller_enabled,
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bool transient_suppressor_enabled);
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@ -228,6 +229,7 @@ class AudioProcessingImpl : public AudioProcessing {
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bool mobile_echo_controller_enabled_ = false;
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bool noise_suppressor_enabled_ = false;
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bool adaptive_gain_controller_enabled_ = false;
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bool voice_activity_detector_enabled_ = false;
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bool gain_controller2_enabled_ = false;
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bool gain_adjustment_enabled_ = false;
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bool echo_controller_enabled_ = false;
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@ -273,6 +275,11 @@ class AudioProcessingImpl : public AudioProcessing {
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// and `config_has_changed` is true, recreates the sub-module.
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void InitializeGainController2(bool config_has_changed)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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// Initializes the `VoiceActivityDetectorWrapper` sub-module. If the
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// sub-module is enabled and `config_has_changed` is true, recreates the
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// sub-module.
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void InitializeVoiceActivityDetector(bool config_has_changed)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeNoiseSuppressor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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void InitializeCaptureLevelsAdjuster()
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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@ -393,6 +400,7 @@ class AudioProcessingImpl : public AudioProcessing {
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std::unique_ptr<AgcManagerDirect> agc_manager;
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std::unique_ptr<GainControlImpl> gain_control;
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std::unique_ptr<GainController2> gain_controller2;
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std::unique_ptr<VoiceActivityDetectorWrapper> voice_activity_detector;
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std::unique_ptr<HighPassFilter> high_pass_filter;
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std::unique_ptr<EchoControl> echo_controller;
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std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
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@ -23,6 +23,7 @@
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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 "test/field_trial.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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@ -481,6 +482,78 @@ TEST(AudioProcessingImplTest,
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apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
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}
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TEST(AudioProcessingImplTest,
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EchoControllerObservesNoDigitalAgc2EchoPathGainChange) {
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// Tests that the echo controller doesn't observe an echo path gain change
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// when the AGC2 digital submodule changes the digital gain without analog
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// gain changes.
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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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rtc::scoped_refptr<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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webrtc::AudioProcessing::Config apm_config;
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// Disable AGC1 analog.
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apm_config.gain_controller1.enabled = false;
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// Enable AGC2 digital.
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apm_config.gain_controller2.enabled = true;
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apm_config.gain_controller2.adaptive_digital.enabled = true;
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apm->ApplyConfig(apm_config);
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constexpr int16_t kAudioLevel = 1000;
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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 stream_config(kSampleRateHz, kNumChannels);
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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, ProcessCapture(NotNull(), testing::_,
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/*echo_path_change=*/false))
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.Times(1);
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apm->ProcessStream(frame.data(), stream_config, stream_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, ProcessCapture(NotNull(), testing::_,
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/*echo_path_change=*/false))
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.Times(1);
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apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
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}
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TEST(AudioProcessingImplTest, ProcessWithAgc2InjectedSpeechProbability) {
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// Tests that a stream is successfully processed for the field trial
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// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` using
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// injected speech probability in AGC2 digital.
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webrtc::test::ScopedFieldTrials field_trials(
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"WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/");
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rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
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ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
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webrtc::AudioProcessing::Config apm_config;
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// Disable AGC1 analog.
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apm_config.gain_controller1.enabled = false;
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// Enable AGC2 digital.
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apm_config.gain_controller2.enabled = true;
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apm_config.gain_controller2.adaptive_digital.enabled = true;
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apm->ApplyConfig(apm_config);
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constexpr int kSampleRateHz = 48000;
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constexpr int kNumChannels = 1;
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std::array<float, kSampleRateHz / 100> buffer;
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float* channel_pointers[] = {buffer.data()};
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StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
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/*num_channels=*/kNumChannels);
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Random random_generator(2341U);
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constexpr int kFramesToProcess = 10;
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for (int i = 0; i < kFramesToProcess; ++i) {
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RandomizeSampleVector(&random_generator, buffer);
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ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
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channel_pointers),
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kNoErr);
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}
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}
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TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) {
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// Tests that the echo controller observes an echo path gain change when a
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// playout volume change is reported.
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@ -69,7 +69,8 @@ int GainController2::instance_count_ = 0;
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GainController2::GainController2(const Agc2Config& config,
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int sample_rate_hz,
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int num_channels)
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int num_channels,
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bool use_internal_vad)
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: cpu_features_(GetAllowedCpuFeatures()),
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data_dumper_(rtc::AtomicOps::Increment(&instance_count_)),
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fixed_gain_applier_(
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@ -86,7 +87,7 @@ GainController2::GainController2(const Agc2Config& config,
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RTC_DCHECK(Validate(config));
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data_dumper_.InitiateNewSetOfRecordings();
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const bool use_vad = config.adaptive_digital.enabled;
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if (use_vad) {
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if (use_vad && use_internal_vad) {
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// TODO(bugs.webrtc.org/7494): Move `vad_reset_period_ms` from adaptive
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// digital to gain controller 2 config.
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vad_ = std::make_unique<VoiceActivityDetectorWrapper>(
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@ -125,13 +126,18 @@ void GainController2::SetFixedGainDb(float gain_db) {
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fixed_gain_applier_.SetGainFactor(gain_factor);
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}
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void GainController2::Process(AudioBuffer* audio) {
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void GainController2::Process(absl::optional<float> speech_probability,
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AudioBuffer* audio) {
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data_dumper_.DumpRaw("agc2_notified_analog_level", analog_level_);
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AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
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audio->num_frames());
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absl::optional<float> speech_probability;
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if (vad_) {
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speech_probability = vad_->Analyze(float_frame);
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} else if (speech_probability.has_value()) {
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RTC_DCHECK_GE(speech_probability.value(), 0.0f);
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RTC_DCHECK_LE(speech_probability.value(), 1.0f);
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}
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if (speech_probability.has_value()) {
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data_dumper_.DumpRaw("agc2_speech_probability", speech_probability.value());
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}
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fixed_gain_applier_.ApplyGain(float_frame);
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@ -30,9 +30,12 @@ class AudioBuffer;
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// microphone gain and/or applying digital gain.
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class GainController2 {
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public:
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// Ctor. If `use_internal_vad` is true, an internal voice activity
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// detector is used for digital adaptive gain.
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GainController2(const AudioProcessing::Config::GainController2& config,
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int sample_rate_hz,
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int num_channels);
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int num_channels,
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bool use_internal_vad);
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GainController2(const GainController2&) = delete;
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GainController2& operator=(const GainController2&) = delete;
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~GainController2();
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@ -44,13 +47,18 @@ class GainController2 {
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void SetFixedGainDb(float gain_db);
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// Applies fixed and adaptive digital gains to `audio` and runs a limiter.
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void Process(AudioBuffer* audio);
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// If the internal VAD is used, `speech_probability` is ignored. Otherwise
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// `speech_probability` is used for digital adaptive gain if it's available
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// (limited to values [0.0, 1.0]).
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void Process(absl::optional<float> speech_probability, AudioBuffer* audio);
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// Handles analog level changes.
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void NotifyAnalogLevel(int level);
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static bool Validate(const AudioProcessing::Config::GainController2& config);
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AvailableCpuFeatures GetCpuFeatures() const { return cpu_features_; }
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private:
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static int instance_count_;
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const AvailableCpuFeatures cpu_features_;
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@ -47,7 +47,7 @@ float RunAgc2WithConstantInput(GainController2& agc2,
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// Give time to the level estimator to converge.
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for (int i = 0; i < num_frames + 1; ++i) {
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SetAudioBufferSamples(input_level, ab);
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agc2.Process(&ab);
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agc2.Process(/*speech_probability=*/absl::nullopt, &ab);
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}
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// Return the last sample from the last processed frame.
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@ -62,7 +62,8 @@ std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode(
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config.fixed_digital.gain_db = fixed_gain_db;
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EXPECT_TRUE(GainController2::Validate(config));
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return std::make_unique<GainController2>(config, sample_rate_hz,
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/*num_channels=*/1);
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/*num_channels=*/1,
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/*use_internal_vad=*/true);
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}
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} // namespace
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@ -138,7 +139,8 @@ TEST(GainController2, CheckAdaptiveDigitalMaxOutputNoiseLevelConfig) {
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// Checks that the default config is applied.
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TEST(GainController2, ApplyDefaultConfig) {
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auto gain_controller2 = std::make_unique<GainController2>(
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Agc2Config{}, /*sample_rate_hz=*/16000, /*num_channels=*/2);
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Agc2Config{}, /*sample_rate_hz=*/16000, /*num_channels=*/2,
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/*use_internal_vad=*/true);
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EXPECT_TRUE(gain_controller2.get());
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}
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@ -253,7 +255,8 @@ TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) {
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Agc2Config config;
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config.fixed_digital.gain_db = 0.0f;
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config.adaptive_digital.enabled = true;
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GainController2 agc2(config, kSampleRateHz, kStereo);
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GainController2 agc2(config, kSampleRateHz, kStereo,
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/*use_internal_vad=*/true);
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test::InputAudioFile input_file(
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test::GetApmCaptureTestVectorFileName(kSampleRateHz),
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@ -276,16 +279,16 @@ TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) {
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stream_config.num_channels(), &input_file,
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frame);
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// Apply a fixed gain to the input audio.
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for (float& x : frame)
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for (float& x : frame) {
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x *= gain;
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}
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test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
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// Process.
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agc2.Process(&audio_buffer);
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agc2.Process(/*speech_probability=*/absl::nullopt, &audio_buffer);
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}
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// Estimate the applied gain by processing a probing frame.
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SetAudioBufferSamples(/*value=*/1.0f, audio_buffer);
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agc2.Process(&audio_buffer);
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agc2.Process(/*speech_probability=*/absl::nullopt, &audio_buffer);
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const float applied_gain_db =
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20.0f * std::log10(audio_buffer.channels_const()[0][0]);
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@ -294,5 +297,196 @@ TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) {
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EXPECT_NEAR(applied_gain_db, kExpectedGainDb, kToleranceDb);
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}
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// Processes a test audio file and checks that the injected speech probability
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// is ignored when the internal VAD is used.
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TEST(GainController2,
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CheckInjectedVadProbabilityNotUsedWithAdaptiveDigitalController) {
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constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
|
||||
constexpr int kStereo = 2;
|
||||
|
||||
// Create AGC2 enabling only the adaptive digital controller.
|
||||
Agc2Config config;
|
||||
config.fixed_digital.gain_db = 0.0f;
|
||||
config.adaptive_digital.enabled = true;
|
||||
GainController2 agc2(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/true);
|
||||
GainController2 agc2_reference(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/true);
|
||||
|
||||
test::InputAudioFile input_file(
|
||||
test::GetApmCaptureTestVectorFileName(kSampleRateHz),
|
||||
/*loop_at_end=*/true);
|
||||
const StreamConfig stream_config(kSampleRateHz, kStereo);
|
||||
|
||||
// Init buffers.
|
||||
constexpr int kFrameDurationMs = 10;
|
||||
std::vector<float> frame(kStereo * stream_config.num_frames());
|
||||
AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
|
||||
kSampleRateHz, kStereo);
|
||||
AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz,
|
||||
kStereo, kSampleRateHz, kStereo);
|
||||
|
||||
// Simulate.
|
||||
constexpr float kGainDb = -6.0f;
|
||||
const float gain = std::pow(10.0f, kGainDb / 20.0f);
|
||||
constexpr int kDurationMs = 10000;
|
||||
constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
|
||||
constexpr float kSpeechProbabilities[] = {1.0f, 0.3f};
|
||||
constexpr float kEpsilon = 0.0001f;
|
||||
bool all_samples_zero = true;
|
||||
for (int i = 0, j = 0; i < kNumFramesToProcess; ++i, j = 1 - j) {
|
||||
ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
|
||||
stream_config.num_channels(), &input_file,
|
||||
frame);
|
||||
// Apply a fixed gain to the input audio.
|
||||
for (float& x : frame) {
|
||||
x *= gain;
|
||||
}
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
|
||||
agc2.Process(kSpeechProbabilities[j], &audio_buffer);
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame,
|
||||
&audio_buffer_reference);
|
||||
agc2_reference.Process(absl::nullopt, &audio_buffer_reference);
|
||||
|
||||
// Check the output buffers.
|
||||
for (int i = 0; i < kStereo; ++i) {
|
||||
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
|
||||
all_samples_zero &=
|
||||
fabs(audio_buffer.channels_const()[i][j]) < kEpsilon;
|
||||
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j],
|
||||
audio_buffer_reference.channels_const()[i][j]);
|
||||
}
|
||||
}
|
||||
}
|
||||
EXPECT_FALSE(all_samples_zero);
|
||||
}
|
||||
|
||||
// Processes a test audio file and checks that the injected speech probability
|
||||
// is not ignored when the internal VAD is not used.
|
||||
TEST(GainController2,
|
||||
CheckInjectedVadProbabilityUsedWithAdaptiveDigitalController) {
|
||||
constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
|
||||
constexpr int kStereo = 2;
|
||||
|
||||
// Create AGC2 enabling only the adaptive digital controller.
|
||||
Agc2Config config;
|
||||
config.fixed_digital.gain_db = 0.0f;
|
||||
config.adaptive_digital.enabled = true;
|
||||
GainController2 agc2(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/false);
|
||||
GainController2 agc2_reference(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/true);
|
||||
|
||||
test::InputAudioFile input_file(
|
||||
test::GetApmCaptureTestVectorFileName(kSampleRateHz),
|
||||
/*loop_at_end=*/true);
|
||||
const StreamConfig stream_config(kSampleRateHz, kStereo);
|
||||
|
||||
// Init buffers.
|
||||
constexpr int kFrameDurationMs = 10;
|
||||
std::vector<float> frame(kStereo * stream_config.num_frames());
|
||||
AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
|
||||
kSampleRateHz, kStereo);
|
||||
AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz,
|
||||
kStereo, kSampleRateHz, kStereo);
|
||||
// Simulate.
|
||||
constexpr float kGainDb = -6.0f;
|
||||
const float gain = std::pow(10.0f, kGainDb / 20.0f);
|
||||
constexpr int kDurationMs = 10000;
|
||||
constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
|
||||
constexpr float kSpeechProbabilities[] = {1.0f, 0.3f};
|
||||
constexpr float kEpsilon = 0.0001f;
|
||||
bool all_samples_zero = true;
|
||||
bool all_samples_equal = true;
|
||||
for (int i = 0, j = 0; i < kNumFramesToProcess; ++i, j = 1 - j) {
|
||||
ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
|
||||
stream_config.num_channels(), &input_file,
|
||||
frame);
|
||||
// Apply a fixed gain to the input audio.
|
||||
for (float& x : frame) {
|
||||
x *= gain;
|
||||
}
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
|
||||
agc2.Process(kSpeechProbabilities[j], &audio_buffer);
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame,
|
||||
&audio_buffer_reference);
|
||||
agc2_reference.Process(absl::nullopt, &audio_buffer_reference);
|
||||
// Check the output buffers.
|
||||
for (int i = 0; i < kStereo; ++i) {
|
||||
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
|
||||
all_samples_zero &=
|
||||
fabs(audio_buffer.channels_const()[i][j]) < kEpsilon;
|
||||
all_samples_equal &=
|
||||
fabs(audio_buffer.channels_const()[i][j] -
|
||||
audio_buffer_reference.channels_const()[i][j]) < kEpsilon;
|
||||
}
|
||||
}
|
||||
}
|
||||
EXPECT_FALSE(all_samples_zero);
|
||||
EXPECT_FALSE(all_samples_equal);
|
||||
}
|
||||
|
||||
// Processes a test audio file and checks that the output is equal when
|
||||
// an injected speech probability from `VoiceActivityDetectorWrapper` and
|
||||
// the speech probability computed by the internal VAD are the same.
|
||||
TEST(GainController2,
|
||||
CheckEqualResultFromInjectedVadProbabilityWithAdaptiveDigitalController) {
|
||||
constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz;
|
||||
constexpr int kStereo = 2;
|
||||
|
||||
// Create AGC2 enabling only the adaptive digital controller.
|
||||
Agc2Config config;
|
||||
config.fixed_digital.gain_db = 0.0f;
|
||||
config.adaptive_digital.enabled = true;
|
||||
GainController2 agc2(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/false);
|
||||
GainController2 agc2_reference(config, kSampleRateHz, kStereo,
|
||||
/*use_internal_vad=*/true);
|
||||
VoiceActivityDetectorWrapper vad(config.adaptive_digital.vad_reset_period_ms,
|
||||
GetAvailableCpuFeatures(), kSampleRateHz);
|
||||
test::InputAudioFile input_file(
|
||||
test::GetApmCaptureTestVectorFileName(kSampleRateHz),
|
||||
/*loop_at_end=*/true);
|
||||
const StreamConfig stream_config(kSampleRateHz, kStereo);
|
||||
|
||||
// Init buffers.
|
||||
constexpr int kFrameDurationMs = 10;
|
||||
std::vector<float> frame(kStereo * stream_config.num_frames());
|
||||
AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo,
|
||||
kSampleRateHz, kStereo);
|
||||
AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz,
|
||||
kStereo, kSampleRateHz, kStereo);
|
||||
|
||||
// Simulate.
|
||||
constexpr float kGainDb = -6.0f;
|
||||
const float gain = std::pow(10.0f, kGainDb / 20.0f);
|
||||
constexpr int kDurationMs = 10000;
|
||||
constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs;
|
||||
for (int i = 0; i < kNumFramesToProcess; ++i) {
|
||||
ReadFloatSamplesFromStereoFile(stream_config.num_frames(),
|
||||
stream_config.num_channels(), &input_file,
|
||||
frame);
|
||||
// Apply a fixed gain to the input audio.
|
||||
for (float& x : frame) {
|
||||
x *= gain;
|
||||
}
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame,
|
||||
&audio_buffer_reference);
|
||||
agc2_reference.Process(absl::nullopt, &audio_buffer_reference);
|
||||
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
|
||||
agc2.Process(vad.Analyze(AudioFrameView<const float>(
|
||||
audio_buffer.channels(), audio_buffer.num_channels(),
|
||||
audio_buffer.num_frames())),
|
||||
&audio_buffer);
|
||||
// Check the output buffer.
|
||||
for (int i = 0; i < kStereo; ++i) {
|
||||
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
|
||||
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j],
|
||||
audio_buffer_reference.channels_const()[i][j]);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
|
||||
Loading…
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Reference in New Issue
Block a user