Make LevelEstimation not a ProcessingComponent.
BUG=webrtc:5355 Review URL: https://codereview.webrtc.org/1523483002 Cr-Commit-Position: refs/heads/master@{#11033}
This commit is contained in:
@ -150,14 +150,13 @@ struct AudioProcessingImpl::ApmPublicSubmodules {
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: echo_cancellation(nullptr),
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: echo_cancellation(nullptr),
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echo_control_mobile(nullptr),
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echo_control_mobile(nullptr),
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gain_control(nullptr),
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gain_control(nullptr),
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level_estimator(nullptr),
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voice_detection(nullptr) {}
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voice_detection(nullptr) {}
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// Accessed externally of APM without any lock acquired.
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// Accessed externally of APM without any lock acquired.
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EchoCancellationImpl* echo_cancellation;
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EchoCancellationImpl* echo_cancellation;
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EchoControlMobileImpl* echo_control_mobile;
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EchoControlMobileImpl* echo_control_mobile;
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GainControlImpl* gain_control;
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GainControlImpl* gain_control;
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rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
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rtc::scoped_ptr<HighPassFilterImpl> high_pass_filter;
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LevelEstimatorImpl* level_estimator;
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rtc::scoped_ptr<LevelEstimatorImpl> level_estimator;
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rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
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rtc::scoped_ptr<NoiseSuppressionImpl> noise_suppression;
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VoiceDetectionImpl* voice_detection;
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VoiceDetectionImpl* voice_detection;
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rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
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rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc;
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@ -243,8 +242,8 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
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new GainControlImpl(this, &crit_capture_, &crit_capture_);
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new GainControlImpl(this, &crit_capture_, &crit_capture_);
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public_submodules_->high_pass_filter.reset(
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public_submodules_->high_pass_filter.reset(
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new HighPassFilterImpl(&crit_capture_));
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new HighPassFilterImpl(&crit_capture_));
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public_submodules_->level_estimator =
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public_submodules_->level_estimator.reset(
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new LevelEstimatorImpl(this, &crit_capture_);
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new LevelEstimatorImpl(&crit_capture_));
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public_submodules_->noise_suppression.reset(
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public_submodules_->noise_suppression.reset(
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new NoiseSuppressionImpl(&crit_capture_));
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new NoiseSuppressionImpl(&crit_capture_));
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public_submodules_->voice_detection =
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public_submodules_->voice_detection =
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@ -258,8 +257,6 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
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public_submodules_->echo_control_mobile);
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public_submodules_->echo_control_mobile);
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private_submodules_->component_list.push_back(
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private_submodules_->component_list.push_back(
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public_submodules_->gain_control);
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public_submodules_->gain_control);
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private_submodules_->component_list.push_back(
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public_submodules_->level_estimator);
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private_submodules_->component_list.push_back(
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private_submodules_->component_list.push_back(
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public_submodules_->voice_detection);
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public_submodules_->voice_detection);
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}
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}
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@ -398,6 +395,7 @@ int AudioProcessingImpl::InitializeLocked() {
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InitializeIntelligibility();
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InitializeIntelligibility();
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InitializeHighPassFilter();
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InitializeHighPassFilter();
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InitializeNoiseSuppression();
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InitializeNoiseSuppression();
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InitializeLevelEstimator();
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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if (debug_dump_.debug_file->Open()) {
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if (debug_dump_.debug_file->Open()) {
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@ -804,7 +802,7 @@ int AudioProcessingImpl::ProcessStreamLocked() {
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}
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}
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// The level estimator operates on the recombined data.
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// The level estimator operates on the recombined data.
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RETURN_ON_ERR(public_submodules_->level_estimator->ProcessStream(ca));
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public_submodules_->level_estimator->ProcessStream(ca);
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capture_.was_stream_delay_set = false;
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capture_.was_stream_delay_set = false;
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return kNoError;
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return kNoError;
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@ -1146,7 +1144,7 @@ HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
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LevelEstimator* AudioProcessingImpl::level_estimator() const {
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LevelEstimator* AudioProcessingImpl::level_estimator() const {
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// Adding a lock here has no effect as it allows any access to the submodule
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// Adding a lock here has no effect as it allows any access to the submodule
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// from the returned pointer.
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// from the returned pointer.
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return public_submodules_->level_estimator;
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return public_submodules_->level_estimator.get();
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}
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}
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NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
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NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
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@ -1178,6 +1176,9 @@ bool AudioProcessingImpl::is_data_processed() const {
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if (public_submodules_->noise_suppression->is_enabled()) {
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if (public_submodules_->noise_suppression->is_enabled()) {
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enabled_count++;
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enabled_count++;
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}
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}
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if (public_submodules_->level_estimator->is_enabled()) {
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enabled_count++;
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}
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// Data is unchanged if no components are enabled, or if only
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// Data is unchanged if no components are enabled, or if only
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// public_submodules_->level_estimator
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// public_submodules_->level_estimator
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@ -1302,6 +1303,10 @@ void AudioProcessingImpl::InitializeNoiseSuppression() {
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proc_sample_rate_hz());
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proc_sample_rate_hz());
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}
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}
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void AudioProcessingImpl::InitializeLevelEstimator() {
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public_submodules_->level_estimator->Initialize();
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}
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void AudioProcessingImpl::MaybeUpdateHistograms() {
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void AudioProcessingImpl::MaybeUpdateHistograms() {
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static const int kMinDiffDelayMs = 60;
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static const int kMinDiffDelayMs = 60;
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@ -185,6 +185,8 @@ class AudioProcessingImpl : public AudioProcessing {
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EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
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EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
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void InitializeNoiseSuppression()
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void InitializeNoiseSuppression()
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EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
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EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
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void InitializeLevelEstimator()
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EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
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int InitializeLocked(const ProcessingConfig& config)
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int InitializeLocked(const ProcessingConfig& config)
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EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
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EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
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@ -11,83 +11,55 @@
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#include "webrtc/modules/audio_processing/level_estimator_impl.h"
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#include "webrtc/modules/audio_processing/level_estimator_impl.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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namespace webrtc {
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namespace webrtc {
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LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm,
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LevelEstimatorImpl::LevelEstimatorImpl(rtc::CriticalSection* crit)
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rtc::CriticalSection* crit)
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: crit_(crit), rms_(new RMSLevel()) {
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: ProcessingComponent(), crit_(crit) {
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RTC_DCHECK(apm);
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RTC_DCHECK(crit);
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RTC_DCHECK(crit);
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}
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}
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LevelEstimatorImpl::~LevelEstimatorImpl() {}
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LevelEstimatorImpl::~LevelEstimatorImpl() {}
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int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
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void LevelEstimatorImpl::Initialize() {
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rtc::CritScope cs(crit_);
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rtc::CritScope cs(crit_);
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rms_->Reset();
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}
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if (!is_component_enabled()) {
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void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
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return AudioProcessing::kNoError;
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RTC_DCHECK(audio);
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rtc::CritScope cs(crit_);
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if (!enabled_) {
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return;
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}
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}
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RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
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for (int i = 0; i < audio->num_channels(); i++) {
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for (int i = 0; i < audio->num_channels(); ++i) {
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rms_->Process(audio->channels_const()[i], audio->num_frames());
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rms_level->Process(audio->channels_const()[i],
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audio->num_frames());
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}
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}
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return AudioProcessing::kNoError;
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}
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}
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int LevelEstimatorImpl::Enable(bool enable) {
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int LevelEstimatorImpl::Enable(bool enable) {
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rtc::CritScope cs(crit_);
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rtc::CritScope cs(crit_);
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return EnableComponent(enable);
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if (enable && !enabled_) {
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rms_->Reset();
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}
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enabled_ = enable;
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return AudioProcessing::kNoError;
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}
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}
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bool LevelEstimatorImpl::is_enabled() const {
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bool LevelEstimatorImpl::is_enabled() const {
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rtc::CritScope cs(crit_);
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rtc::CritScope cs(crit_);
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return is_component_enabled();
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return enabled_;
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}
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}
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int LevelEstimatorImpl::RMS() {
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int LevelEstimatorImpl::RMS() {
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rtc::CritScope cs(crit_);
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rtc::CritScope cs(crit_);
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if (!is_component_enabled()) {
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if (!enabled_) {
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return AudioProcessing::kNotEnabledError;
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return AudioProcessing::kNotEnabledError;
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}
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}
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RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
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return rms_->RMS();
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return rms_level->RMS();
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}
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}
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// The ProcessingComponent implementation is pretty weird in this class since
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// we have only a single instance of the trivial underlying component.
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void* LevelEstimatorImpl::CreateHandle() const {
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return new RMSLevel;
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}
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void LevelEstimatorImpl::DestroyHandle(void* handle) const {
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delete static_cast<RMSLevel*>(handle);
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}
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int LevelEstimatorImpl::InitializeHandle(void* handle) const {
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rtc::CritScope cs(crit_);
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static_cast<RMSLevel*>(handle)->Reset();
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return AudioProcessing::kNoError;
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}
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int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
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return AudioProcessing::kNoError;
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}
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int LevelEstimatorImpl::num_handles_required() const {
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return 1;
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}
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int LevelEstimatorImpl::GetHandleError(void* /*handle*/) const {
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return AudioProcessing::kUnspecifiedError;
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}
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} // namespace webrtc
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} // namespace webrtc
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@ -11,42 +11,36 @@
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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namespace webrtc {
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namespace webrtc {
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class AudioBuffer;
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class AudioBuffer;
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class RMSLevel;
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class LevelEstimatorImpl : public LevelEstimator,
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class LevelEstimatorImpl : public LevelEstimator {
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public ProcessingComponent {
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public:
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public:
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LevelEstimatorImpl(const AudioProcessing* apm, rtc::CriticalSection* crit);
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explicit LevelEstimatorImpl(rtc::CriticalSection* crit);
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virtual ~LevelEstimatorImpl();
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~LevelEstimatorImpl() override;
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int ProcessStream(AudioBuffer* audio);
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// TODO(peah): Fold into ctor, once public API is removed.
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void Initialize();
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void ProcessStream(AudioBuffer* audio);
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// LevelEstimator implementation.
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bool is_enabled() const override;
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private:
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// LevelEstimator implementation.
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// LevelEstimator implementation.
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int Enable(bool enable) override;
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int Enable(bool enable) override;
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bool is_enabled() const override;
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int RMS() override;
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int RMS() override;
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// ProcessingComponent implementation.
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private:
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void* CreateHandle() const override;
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rtc::CriticalSection* const crit_ = nullptr;
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int InitializeHandle(void* handle) const override;
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bool enabled_ GUARDED_BY(crit_) = false;
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int ConfigureHandle(void* handle) const override;
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rtc::scoped_ptr<RMSLevel> rms_ GUARDED_BY(crit_);
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void DestroyHandle(void* handle) const override;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LevelEstimatorImpl);
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int num_handles_required() const override;
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int GetHandleError(void* handle) const override;
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rtc::CriticalSection* const crit_;
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};
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};
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} // namespace webrtc
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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Block a user