| /* |
| * 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 <optional> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/base/nullability.h" |
| #include "absl/strings/string_view.h" |
| #include "api/array_view.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/audio/audio_processing_statistics.h" |
| #include "api/function_view.h" |
| #include "api/task_queue/task_queue_base.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/ns/noise_suppressor.h" |
| #include "modules/audio_processing/render_queue_item_verifier.h" |
| #include "modules/audio_processing/rms_level.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, |
| absl::Nonnull<TaskQueueBase*> worker_queue) override; |
| bool CreateAndAttachAecDump( |
| FILE* handle, |
| int64_t max_log_size_bytes, |
| absl::Nonnull<TaskQueueBase*> 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); |
| |
| void set_stream_analog_level_locked(int level) |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_); |
| void UpdateRecommendedInputVolumeLocked() |
| RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_); |
| |
| // 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_; |
| |
| 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 gain_adjustment_enabled, |
| bool echo_controller_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 gain_controller2_enabled_ = false; |
| bool gain_adjustment_enabled_ = false; |
| bool echo_controller_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_); |
| // Initializes the `GainController2` sub-module. If the sub-module is enabled, |
| // recreates it. |
| void InitializeGainController2() 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_; |
| |
| // 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<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<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) |
| : 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) {} |
| bool multi_channel_render_support; |
| bool multi_channel_capture_support; |
| bool enforce_split_band_hpf; |
| bool minimize_processing_for_unused_output; |
| } 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. |
| std::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. |
| std::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_ |