summaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_processing/audio_processing_impl.h
diff options
context:
space:
mode:
Diffstat (limited to 'webrtc/modules/audio_processing/audio_processing_impl.h')
-rw-r--r--webrtc/modules/audio_processing/audio_processing_impl.h654
1 files changed, 478 insertions, 176 deletions
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 542886e..d0eec0e 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -8,212 +8,514 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+
+#include <stdio.h>
#include <list>
+#include <memory>
#include <string>
#include <vector>
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/audio_processing/include/audio_processing.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/audio_buffer.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/level_estimator.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/residual_echo_detector.h"
+#include "modules/audio_processing/rms_level.h"
+#include "modules/audio_processing/transient/transient_suppressor.h"
+#include "modules/audio_processing/voice_detection.h"
+#include "rtc_base/gtest_prod_util.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/swap_queue.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
namespace webrtc {
-class AgcManagerDirect;
-class AudioBuffer;
+class ApmDataDumper;
class AudioConverter;
-template<typename T>
-class Beamformer;
-
-class CriticalSectionWrapper;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class FileWrapper;
-class GainControlImpl;
-class GainControlForNewAgc;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class TransientSuppressor;
-class VoiceDetectionImpl;
-class IntelligibilityEnhancer;
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-namespace audioproc {
-
-class Event;
-
-} // namespace audioproc
-#endif
-
class AudioProcessingImpl : public AudioProcessing {
public:
- explicit AudioProcessingImpl(const Config& config);
-
- // AudioProcessingImpl takes ownership of beamformer.
- AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
- virtual ~AudioProcessingImpl();
-
- // AudioProcessing methods.
+ // Methods forcing APM to run in a single-threaded manner.
+ // Acquires both the render and capture locks.
+ explicit AudioProcessingImpl(const webrtc::Config& config);
+ // AudioProcessingImpl takes ownership of capture post processor.
+ AudioProcessingImpl(const webrtc::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(int input_sample_rate_hz,
- int output_sample_rate_hz,
- int reverse_sample_rate_hz,
- ChannelLayout input_layout,
- ChannelLayout output_layout,
- ChannelLayout reverse_layout) override;
+ int Initialize(int capture_input_sample_rate_hz,
+ int capture_output_sample_rate_hz,
+ int render_sample_rate_hz,
+ ChannelLayout capture_input_layout,
+ ChannelLayout capture_output_layout,
+ ChannelLayout render_input_layout) override;
int Initialize(const ProcessingConfig& processing_config) override;
- void SetExtraOptions(const Config& config) override;
- int proc_sample_rate_hz() const override;
- int proc_split_sample_rate_hz() const override;
- int num_input_channels() const override;
- int num_output_channels() const override;
- int num_reverse_channels() const override;
- void set_output_will_be_muted(bool muted) override;
- int ProcessStream(AudioFrame* frame) override;
- int ProcessStream(const float* const* src,
- size_t samples_per_channel,
- int input_sample_rate_hz,
- ChannelLayout input_layout,
- int output_sample_rate_hz,
- ChannelLayout output_layout,
- float* const* dest) override;
+ void ApplyConfig(const AudioProcessing::Config& config) override;
+ bool CreateAndAttachAecDump(const std::string& 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;
+
+ // 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;
- int AnalyzeReverseStream(AudioFrame* frame) override;
- int ProcessReverseStream(AudioFrame* frame) override;
+ bool GetLinearAecOutput(
+ rtc::ArrayView<std::array<float, 160>> linear_output) const override;
+ void set_output_will_be_muted(bool muted) override;
+ 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,
- size_t samples_per_channel,
- int sample_rate_hz,
- ChannelLayout layout) override;
+ const StreamConfig& reverse_config) override;
int ProcessReverseStream(const float* const* src,
- const StreamConfig& reverse_input_config,
- const StreamConfig& reverse_output_config,
+ const StreamConfig& input_config,
+ const StreamConfig& output_config,
float* const* dest) override;
- int set_stream_delay_ms(int delay) 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;
- bool was_stream_delay_set() const override;
- void set_delay_offset_ms(int offset) override;
- int delay_offset_ms() const override;
- void set_stream_key_pressed(bool key_pressed) override;
- int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
- int StartDebugRecording(FILE* handle) override;
- int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
- int StopDebugRecording() override;
- void UpdateHistogramsOnCallEnd() override;
- EchoCancellation* echo_cancellation() const override;
- EchoControlMobile* echo_control_mobile() const override;
- GainControl* gain_control() const override;
- HighPassFilter* high_pass_filter() const override;
- LevelEstimator* level_estimator() const override;
- NoiseSuppression* noise_suppression() const override;
- VoiceDetection* voice_detection() const override;
+
+ AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
+ return GetStatistics();
+ }
+ AudioProcessingStats GetStatistics() override {
+ return stats_reporter_.GetStatistics();
+ }
+
+ // TODO(peah): Remove MutateConfig once the new API allows that.
+ void MutateConfig(rtc::FunctionView<void(AudioProcessing::Config*)> mutator);
+ AudioProcessing::Config GetConfig() const override;
protected:
// Overridden in a mock.
- virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+ virtual void InitializeLocked()
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_, mutex_capture_);
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);
+
+ int recommended_stream_analog_level_locked() const
+ 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();
+ void Enqueue(RuntimeSetting setting);
+
+ private:
+ SwapQueue<RuntimeSetting>& runtime_settings_;
+ };
+
+ std::unique_ptr<ApmDataDumper> data_dumper_;
+ static 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.
+ 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 residual_echo_detector_enabled,
+ bool noise_suppressor_enabled,
+ bool adaptive_gain_controller_enabled,
+ bool gain_controller2_enabled,
+ bool pre_amplifier_enabled,
+ bool echo_controller_enabled,
+ bool voice_detector_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 residual_echo_detector_enabled_ = false;
+ bool noise_suppressor_enabled_ = false;
+ bool adaptive_gain_controller_enabled_ = false;
+ bool gain_controller2_enabled_ = false;
+ bool pre_amplifier_enabled_ = false;
+ bool echo_controller_enabled_ = false;
+ bool voice_detector_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.
+ int MaybeInitializeRender(const ProcessingConfig& processing_config)
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_render_);
+ // Called by capture: Holds the capture lock when reading the format struct
+ // and acquires both locks if reinitialization is needed.
+ int 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.
int InitializeLocked(const ProcessingConfig& config)
- EXCLUSIVE_LOCKS_REQUIRED(crit_);
- int MaybeInitializeLocked(const ProcessingConfig& config)
- EXCLUSIVE_LOCKS_REQUIRED(crit_);
+ 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 submodules, requiring the capture lock
+ // already acquired.
+ void InitializeHighPassFilter(bool forced_reset)
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializeVoiceDetector() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializeGainController1() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializeTransientSuppressor()
+ RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializeNoiseSuppressor() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
+ void InitializePreAmplifier() 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.
// TODO(ekm): Remove once all clients updated to new interface.
- int AnalyzeReverseStream(const float* const* src,
- const StreamConfig& input_config,
- const StreamConfig& output_config);
- int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
- bool is_data_processed() const;
- bool output_copy_needed(bool is_data_processed) const;
- bool synthesis_needed(bool is_data_processed) const;
- bool analysis_needed(bool is_data_processed) const;
- bool is_rev_processed() const;
- bool rev_conversion_needed() const;
- void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
- void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-
- EchoCancellationImpl* echo_cancellation_;
- EchoControlMobileImpl* echo_control_mobile_;
- GainControlImpl* gain_control_;
- HighPassFilterImpl* high_pass_filter_;
- LevelEstimatorImpl* level_estimator_;
- NoiseSuppressionImpl* noise_suppression_;
- VoiceDetectionImpl* voice_detection_;
- rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
-
- std::list<ProcessingComponent*> component_list_;
- CriticalSectionWrapper* crit_;
- rtc::scoped_ptr<AudioBuffer> render_audio_;
- rtc::scoped_ptr<AudioBuffer> capture_audio_;
- rtc::scoped_ptr<AudioConverter> render_converter_;
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- // TODO(andrew): make this more graceful. Ideally we would split this stuff
- // out into a separate class with an "enabled" and "disabled" implementation.
- int WriteMessageToDebugFile();
- int WriteInitMessage();
-
- // Writes Config message. 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.
- int WriteConfigMessage(bool forced);
-
- rtc::scoped_ptr<FileWrapper> debug_file_;
- rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
- std::string event_str_; // Memory for protobuf serialization.
-
- // Serialized string of last saved APM configuration.
- std::string last_serialized_config_;
-#endif
-
- // Format of processing streams at input/output call sites.
- ProcessingConfig api_format_;
-
- // Only the rate and samples fields of fwd_proc_format_ are used because the
- // forward processing number of channels is mutable and is tracked by the
- // capture_audio_.
- StreamConfig fwd_proc_format_;
- StreamConfig rev_proc_format_;
- int split_rate_;
-
- int stream_delay_ms_;
- int delay_offset_ms_;
- bool was_stream_delay_set_;
- int last_stream_delay_ms_;
- int last_aec_system_delay_ms_;
- int stream_delay_jumps_;
- int aec_system_delay_jumps_;
-
- bool output_will_be_muted_ GUARDED_BY(crit_);
-
- bool key_pressed_;
-
- // Only set through the constructor's Config parameter.
- const bool use_new_agc_;
- rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
- int agc_startup_min_volume_;
-
- bool transient_suppressor_enabled_;
- rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
- const bool beamformer_enabled_;
- rtc::scoped_ptr<Beamformer<float>> beamformer_;
- const std::vector<Point> array_geometry_;
- const SphericalPointf target_direction_;
-
- bool intelligibility_enabled_;
- rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
+ 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_);
+
+ // 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.
+ 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;
+ rtc::scoped_refptr<EchoDetector> echo_detector;
+ 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<CustomProcessing> capture_post_processor;
+ std::unique_ptr<CustomProcessing> render_pre_processor;
+ std::unique_ptr<GainApplier> pre_amplifier;
+ std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
+ std::unique_ptr<LevelEstimator> output_level_estimator;
+ std::unique_ptr<VoiceDetection> voice_detector;
+ } 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, false},
+ {kSampleRate16kHz, 1, false},
+ {kSampleRate16kHz, 1, false},
+ {kSampleRate16kHz, 1, false}}}),
+ 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)
+ : multi_channel_render_support(multi_channel_render_support),
+ multi_channel_capture_support(multi_channel_capture_support),
+ enforce_split_band_hpf(enforce_split_band_hpf) {}
+ bool multi_channel_render_support;
+ bool multi_channel_capture_support;
+ bool enforce_split_band_hpf;
+ } constants_;
+
+ struct ApmCaptureState {
+ ApmCaptureState();
+ ~ApmCaptureState();
+ bool was_stream_delay_set;
+ bool output_will_be_muted;
+ 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;
+ int prev_analog_mic_level;
+ float prev_pre_amp_gain;
+ int playout_volume;
+ int prev_playout_volume;
+ AudioProcessingStats stats;
+ struct KeyboardInfo {
+ void Extract(const float* const* data, const StreamConfig& stream_config);
+ size_t num_keyboard_frames = 0;
+ const float* keyboard_data = nullptr;
+ } keyboard_info;
+ int cached_stream_analog_level_ = 0;
+ } 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;
+
+ // 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 // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
+#endif // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_