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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#include <stdio.h>
#include <atomic>
#include <list>
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/function_view.h"
#include "modules/audio_processing/aec3/echo_canceller3.h"
#include "modules/audio_processing/agc/agc_manager_direct.h"
#include "modules/audio_processing/agc/gain_control.h"
#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
#include "modules/audio_processing/echo_control_mobile_impl.h"
#include "modules/audio_processing/gain_control_impl.h"
#include "modules/audio_processing/gain_controller2.h"
#include "modules/audio_processing/high_pass_filter.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/ns/noise_suppressor.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"
#include "modules/audio_processing/render_queue_item_verifier.h"
#include "modules/audio_processing/rms_level.h"
#include "modules/audio_processing/transient/transient_suppressor.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/swap_queue.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class ApmDataDumper;
class AudioConverter;
constexpr int RuntimeSettingQueueSize() {
return 100;
}
class AudioProcessingImpl : public AudioProcessing {
public:
// Methods forcing APM to run in a single-threaded manner.
// Acquires both the render and capture locks.
AudioProcessingImpl();
AudioProcessingImpl(const AudioProcessing::Config& config,
std::unique_ptr<CustomProcessing> capture_post_processor,
std::unique_ptr<CustomProcessing> render_pre_processor,
std::unique_ptr<EchoControlFactory> echo_control_factory,
rtc::scoped_refptr<EchoDetector> echo_detector,
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer);
~AudioProcessingImpl() override;
int Initialize() override;
int Initialize(const ProcessingConfig& processing_config) override;
void ApplyConfig(const AudioProcessing::Config& config) override;
bool CreateAndAttachAecDump(absl::string_view file_name,
int64_t max_log_size_bytes,
rtc::TaskQueue* worker_queue) override;
bool CreateAndAttachAecDump(FILE* handle,
int64_t max_log_size_bytes,
rtc::TaskQueue* worker_queue) override;
// TODO(webrtc:5298) Deprecated variant.
void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
void DetachAecDump() override;
void SetRuntimeSetting(RuntimeSetting setting) override;
bool PostRuntimeSetting(RuntimeSetting setting) override;
// Capture-side exclusive methods possibly running APM in a
// multi-threaded manner. Acquire the capture lock.
int ProcessStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) override;
int ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) override;
bool GetLinearAecOutput(
rtc::ArrayView<std::array<float, 160>> linear_output) const override;
void set_output_will_be_muted(bool muted) override;
void HandleCaptureOutputUsedSetting(bool capture_output_used)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
int set_stream_delay_ms(int delay) override;
void set_stream_key_pressed(bool key_pressed) override;
void set_stream_analog_level(int level) override;
int recommended_stream_analog_level() const
RTC_LOCKS_EXCLUDED(mutex_capture_) override;
// Render-side exclusive methods possibly running APM in a
// multi-threaded manner. Acquire the render lock.
int ProcessReverseStream(const int16_t* const src,
const StreamConfig& input_config,
const StreamConfig& output_config,
int16_t* const dest) override;
int AnalyzeReverseStream(const float* const* data,
const StreamConfig& reverse_config) override;
int ProcessReverseStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) override;
// Methods only accessed from APM submodules or
// from AudioProcessing tests in a single-threaded manner.
// Hence there is no need for locks in these.
int proc_sample_rate_hz() const override;
int proc_split_sample_rate_hz() const override;
size_t num_input_channels() const override;
size_t num_proc_channels() const override;
size_t num_output_channels() const override;
size_t num_reverse_channels() const override;
int stream_delay_ms() const override;
AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
return GetStatistics();
}
AudioProcessingStats GetStatistics() override {
return stats_reporter_.GetStatistics();
}
AudioProcessing::Config GetConfig() const override;
protected:
// Overridden in a mock.
virtual void InitializeLocked()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
void AssertLockedForTest()
RTC_ASSERT_EXCLUSIVE_LOCK(mutex_render_, mutex_capture_) {
mutex_render_.AssertHeld();
mutex_capture_.AssertHeld();
}
private:
// TODO(peah): These friend classes should be removed as soon as the new
// parameter setting scheme allows.
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
ToggleTransientSuppressor);
FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
ReinitializeTransientSuppressor);
FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
BitexactWithDisabledModules);
FRIEND_TEST_ALL_PREFIXES(
AudioProcessingImplGainController2FieldTrialParametrizedTest,
ConfigAdjustedWhenExperimentEnabled);
void set_stream_analog_level_locked(int level)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void UpdateRecommendedInputVolumeLocked()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void OverrideSubmoduleCreationForTesting(
const ApmSubmoduleCreationOverrides& overrides);
// Class providing thread-safe message pipe functionality for
// `runtime_settings_`.
class RuntimeSettingEnqueuer {
public:
explicit RuntimeSettingEnqueuer(
SwapQueue<RuntimeSetting>* runtime_settings);
~RuntimeSettingEnqueuer();
// Enqueue setting and return whether the setting was successfully enqueued.
bool Enqueue(RuntimeSetting setting);
private:
SwapQueue<RuntimeSetting>& runtime_settings_;
};
const std::unique_ptr<ApmDataDumper> data_dumper_;
static std::atomic<int> instance_count_;
const bool use_setup_specific_default_aec3_config_;
// Parameters for the "GainController2" experiment which determines whether
// the following APM sub-modules are created and, if so, their configurations:
// AGC2 (`gain_controller2`), AGC1 (`gain_control`, `agc_manager`) and TS
// (`transient_suppressor`).
// TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
// field trial is removed.
struct GainController2ExperimentParams {
struct Agc2Config {
InputVolumeController::Config input_volume_controller;
AudioProcessing::Config::GainController2::AdaptiveDigital
adaptive_digital_controller;
};
// When `agc2_config` is specified, all gain control switches to AGC2 and
// the configuration is overridden.
absl::optional<Agc2Config> agc2_config;
// When true, the transient suppressor submodule is never created regardless
// of the APM configuration.
bool disallow_transient_suppressor_usage;
};
// Specified when the "WebRTC-Audio-GainController2" field trial is specified.
// TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
// field trial is removed.
const absl::optional<GainController2ExperimentParams>
gain_controller2_experiment_params_;
// Parses the "WebRTC-Audio-GainController2" field trial. If disabled, returns
// an unspecified value.
static absl::optional<GainController2ExperimentParams>
GetGainController2ExperimentParams();
// When `experiment_params` is specified, returns an APM configuration
// modified according to the experiment parameters. Otherwise returns
// `config`.
static AudioProcessing::Config AdjustConfig(
const AudioProcessing::Config& config,
const absl::optional<GainController2ExperimentParams>& experiment_params);
// Returns true if the APM VAD sub-module should be used.
static bool UseApmVadSubModule(
const AudioProcessing::Config& config,
const absl::optional<GainController2ExperimentParams>& experiment_params);
TransientSuppressor::VadMode transient_suppressor_vad_mode_;
SwapQueue<RuntimeSetting> capture_runtime_settings_;
SwapQueue<RuntimeSetting> render_runtime_settings_;
RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
// EchoControl factory.
const std::unique_ptr<EchoControlFactory> echo_control_factory_;
class SubmoduleStates {
public:
SubmoduleStates(bool capture_post_processor_enabled,
bool render_pre_processor_enabled,
bool capture_analyzer_enabled);
// Updates the submodule state and returns true if it has changed.
bool Update(bool high_pass_filter_enabled,
bool mobile_echo_controller_enabled,
bool noise_suppressor_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
bool voice_activity_detector_enabled,
bool gain_adjustment_enabled,
bool echo_controller_enabled,
bool transient_suppressor_enabled);
bool CaptureMultiBandSubModulesActive() const;
bool CaptureMultiBandProcessingPresent() const;
bool CaptureMultiBandProcessingActive(bool ec_processing_active) const;
bool CaptureFullBandProcessingActive() const;
bool CaptureAnalyzerActive() const;
bool RenderMultiBandSubModulesActive() const;
bool RenderFullBandProcessingActive() const;
bool RenderMultiBandProcessingActive() const;
bool HighPassFilteringRequired() const;
private:
const bool capture_post_processor_enabled_ = false;
const bool render_pre_processor_enabled_ = false;
const bool capture_analyzer_enabled_ = false;
bool high_pass_filter_enabled_ = false;
bool mobile_echo_controller_enabled_ = false;
bool noise_suppressor_enabled_ = false;
bool adaptive_gain_controller_enabled_ = false;
bool voice_activity_detector_enabled_ = false;
bool gain_controller2_enabled_ = false;
bool gain_adjustment_enabled_ = false;
bool echo_controller_enabled_ = false;
bool transient_suppressor_enabled_ = false;
bool first_update_ = true;
};
// Methods for modifying the formats struct that is used by both
// the render and capture threads. The check for whether modifications are
// needed is done while holding a single lock only, thereby avoiding that the
// capture thread blocks the render thread.
// Called by render: Holds the render lock when reading the format struct and
// acquires both locks if reinitialization is required.
void MaybeInitializeRender(const StreamConfig& input_config,
const StreamConfig& output_config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
// Called by capture: Acquires and releases the capture lock to read the
// format struct and acquires both locks if reinitialization is needed.
void MaybeInitializeCapture(const StreamConfig& input_config,
const StreamConfig& output_config);
// Method for updating the state keeping track of the active submodules.
// Returns a bool indicating whether the state has changed.
bool UpdateActiveSubmoduleStates()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Methods requiring APM running in a single-threaded manner, requiring both
// the render and capture lock to be acquired.
void InitializeLocked(const ProcessingConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
void InitializeResidualEchoDetector()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
void InitializeEchoController()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
// Initializations of capture-only sub-modules, requiring the capture lock
// already acquired.
void InitializeHighPassFilter(bool forced_reset)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializeTransientSuppressor()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Initializes the `GainController2` sub-module. If the sub-module is enabled,
// recreates it.
void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Initializes the `VoiceActivityDetectorWrapper` sub-module. If the
// sub-module is enabled, recreates it. Call `InitializeGainController2()`
// first.
// TODO(bugs.webrtc.org/13663): Remove if TS is removed otherwise remove call
// order requirement - i.e., decouple from `InitializeGainController2()`.
void InitializeVoiceActivityDetector()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializeNoiseSuppressor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializeCaptureLevelsAdjuster()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void InitializeAnalyzer() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Initializations of render-only submodules, requiring the render lock
// already acquired.
void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
// Sample rate used for the fullband processing.
int proc_fullband_sample_rate_hz() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Empties and handles the respective RuntimeSetting queues.
void HandleCaptureRuntimeSettings()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void HandleRenderRuntimeSettings()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
void EmptyQueuedRenderAudio() RTC_LOCKS_EXCLUDED(mutex_capture_);
void EmptyQueuedRenderAudioLocked()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void AllocateRenderQueue()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
void QueueBandedRenderAudio(AudioBuffer* audio)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
void QueueNonbandedRenderAudio(AudioBuffer* audio)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
// Capture-side exclusive methods possibly running APM in a multi-threaded
// manner that are called with the render lock already acquired.
int ProcessCaptureStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Render-side exclusive methods possibly running APM in a multi-threaded
// manner that are called with the render lock already acquired.
int AnalyzeReverseStreamLocked(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
int ProcessRenderStreamLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
// Collects configuration settings from public and private
// submodules to be saved as an audioproc::Config message on the
// AecDump if it is attached. If not `forced`, only writes the current
// config if it is different from the last saved one; if `forced`,
// writes the config regardless of the last saved.
void WriteAecDumpConfigMessage(bool forced)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Notifies attached AecDump of current configuration and capture data.
void RecordUnprocessedCaptureStream(const float* const* capture_stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void RecordUnprocessedCaptureStream(const int16_t* const data,
const StreamConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Notifies attached AecDump of current configuration and
// processed capture data and issues a capture stream recording
// request.
void RecordProcessedCaptureStream(
const float* const* processed_capture_stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
void RecordProcessedCaptureStream(const int16_t* const data,
const StreamConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Notifies attached AecDump about current state (delay, drift, etc).
void RecordAudioProcessingState()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Ensures that overruns in the capture runtime settings queue is properly
// handled by the code, providing safe-fallbacks to mitigate the implications
// of any settings being missed.
void HandleOverrunInCaptureRuntimeSettingsQueue()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// AecDump instance used for optionally logging APM config, input
// and output to file in the AEC-dump format defined in debug.proto.
std::unique_ptr<AecDump> aec_dump_;
// Hold the last config written with AecDump for avoiding writing
// the same config twice.
InternalAPMConfig apm_config_for_aec_dump_ RTC_GUARDED_BY(mutex_capture_);
// Critical sections.
mutable Mutex mutex_render_ RTC_ACQUIRED_BEFORE(mutex_capture_);
mutable Mutex mutex_capture_;
// Struct containing the Config specifying the behavior of APM.
AudioProcessing::Config config_;
// Overrides for testing the exclusion of some submodules from the build.
ApmSubmoduleCreationOverrides submodule_creation_overrides_
RTC_GUARDED_BY(mutex_capture_);
// Class containing information about what submodules are active.
SubmoduleStates submodule_states_;
// Struct containing the pointers to the submodules.
struct Submodules {
Submodules(std::unique_ptr<CustomProcessing> capture_post_processor,
std::unique_ptr<CustomProcessing> render_pre_processor,
rtc::scoped_refptr<EchoDetector> echo_detector,
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer)
: echo_detector(std::move(echo_detector)),
capture_post_processor(std::move(capture_post_processor)),
render_pre_processor(std::move(render_pre_processor)),
capture_analyzer(std::move(capture_analyzer)) {}
// Accessed internally from capture or during initialization.
const rtc::scoped_refptr<EchoDetector> echo_detector;
const std::unique_ptr<CustomProcessing> capture_post_processor;
const std::unique_ptr<CustomProcessing> render_pre_processor;
const std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
std::unique_ptr<AgcManagerDirect> agc_manager;
std::unique_ptr<GainControlImpl> gain_control;
std::unique_ptr<GainController2> gain_controller2;
std::unique_ptr<VoiceActivityDetectorWrapper> voice_activity_detector;
std::unique_ptr<HighPassFilter> high_pass_filter;
std::unique_ptr<EchoControl> echo_controller;
std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
std::unique_ptr<NoiseSuppressor> noise_suppressor;
std::unique_ptr<TransientSuppressor> transient_suppressor;
std::unique_ptr<CaptureLevelsAdjuster> capture_levels_adjuster;
} submodules_;
// State that is written to while holding both the render and capture locks
// but can be read without any lock being held.
// As this is only accessed internally of APM, and all internal methods in APM
// either are holding the render or capture locks, this construct is safe as
// it is not possible to read the variables while writing them.
struct ApmFormatState {
ApmFormatState()
: // Format of processing streams at input/output call sites.
api_format({{{kSampleRate16kHz, 1},
{kSampleRate16kHz, 1},
{kSampleRate16kHz, 1},
{kSampleRate16kHz, 1}}}),
render_processing_format(kSampleRate16kHz, 1) {}
ProcessingConfig api_format;
StreamConfig render_processing_format;
} formats_;
// APM constants.
const struct ApmConstants {
ApmConstants(bool multi_channel_render_support,
bool multi_channel_capture_support,
bool enforce_split_band_hpf,
bool minimize_processing_for_unused_output,
bool transient_suppressor_forced_off)
: multi_channel_render_support(multi_channel_render_support),
multi_channel_capture_support(multi_channel_capture_support),
enforce_split_band_hpf(enforce_split_band_hpf),
minimize_processing_for_unused_output(
minimize_processing_for_unused_output),
transient_suppressor_forced_off(transient_suppressor_forced_off) {}
bool multi_channel_render_support;
bool multi_channel_capture_support;
bool enforce_split_band_hpf;
bool minimize_processing_for_unused_output;
bool transient_suppressor_forced_off;
} constants_;
struct ApmCaptureState {
ApmCaptureState();
~ApmCaptureState();
bool was_stream_delay_set;
bool capture_output_used;
bool capture_output_used_last_frame;
bool key_pressed;
std::unique_ptr<AudioBuffer> capture_audio;
std::unique_ptr<AudioBuffer> capture_fullband_audio;
std::unique_ptr<AudioBuffer> linear_aec_output;
// Only the rate and samples fields of capture_processing_format_ are used
// because the capture processing number of channels is mutable and is
// tracked by the capture_audio_.
StreamConfig capture_processing_format;
int split_rate;
bool echo_path_gain_change;
float prev_pre_adjustment_gain;
int playout_volume;
int prev_playout_volume;
AudioProcessingStats stats;
// Input volume applied on the audio input device when the audio is
// acquired. Unspecified when unknown.
absl::optional<int> applied_input_volume;
bool applied_input_volume_changed;
// Recommended input volume to apply on the audio input device the next time
// that audio is acquired. Unspecified when no input volume can be
// recommended.
absl::optional<int> recommended_input_volume;
} capture_ RTC_GUARDED_BY(mutex_capture_);
struct ApmCaptureNonLockedState {
ApmCaptureNonLockedState()
: capture_processing_format(kSampleRate16kHz),
split_rate(kSampleRate16kHz),
stream_delay_ms(0) {}
// Only the rate and samples fields of capture_processing_format_ are used
// because the forward processing number of channels is mutable and is
// tracked by the capture_audio_.
StreamConfig capture_processing_format;
int split_rate;
int stream_delay_ms;
bool echo_controller_enabled = false;
} capture_nonlocked_;
struct ApmRenderState {
ApmRenderState();
~ApmRenderState();
std::unique_ptr<AudioConverter> render_converter;
std::unique_ptr<AudioBuffer> render_audio;
} render_ RTC_GUARDED_BY(mutex_render_);
// Class for statistics reporting. The class is thread-safe and no lock is
// needed when accessing it.
class ApmStatsReporter {
public:
ApmStatsReporter();
~ApmStatsReporter();
// Returns the most recently reported statistics.
AudioProcessingStats GetStatistics();
// Update the cached statistics.
void UpdateStatistics(const AudioProcessingStats& new_stats);
private:
Mutex mutex_stats_;
AudioProcessingStats cached_stats_ RTC_GUARDED_BY(mutex_stats_);
SwapQueue<AudioProcessingStats> stats_message_queue_;
} stats_reporter_;
std::vector<int16_t> aecm_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
std::vector<int16_t> aecm_capture_queue_buffer_
RTC_GUARDED_BY(mutex_capture_);
size_t agc_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
RTC_GUARDED_BY(mutex_capture_) = 0;
std::vector<int16_t> agc_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
std::vector<int16_t> agc_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
size_t red_render_queue_element_max_size_ RTC_GUARDED_BY(mutex_render_)
RTC_GUARDED_BY(mutex_capture_) = 0;
std::vector<float> red_render_queue_buffer_ RTC_GUARDED_BY(mutex_render_);
std::vector<float> red_capture_queue_buffer_ RTC_GUARDED_BY(mutex_capture_);
RmsLevel capture_input_rms_ RTC_GUARDED_BY(mutex_capture_);
RmsLevel capture_output_rms_ RTC_GUARDED_BY(mutex_capture_);
int capture_rms_interval_counter_ RTC_GUARDED_BY(mutex_capture_) = 0;
InputVolumeStatsReporter applied_input_volume_stats_reporter_
RTC_GUARDED_BY(mutex_capture_);
InputVolumeStatsReporter recommended_input_volume_stats_reporter_
RTC_GUARDED_BY(mutex_capture_);
// Lock protection not needed.
std::unique_ptr<
SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
aecm_render_signal_queue_;
std::unique_ptr<
SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
agc_render_signal_queue_;
std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
red_render_signal_queue_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_