Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions

View File

@ -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;
int AnalyzeReverseStream(const float* const* data,
size_t samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) override;
int ProcessReverseStream(const float* const* src,
const StreamConfig& reverse_input_config,
const StreamConfig& reverse_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;
int set_stream_delay_ms(int delay) 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;
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();
}
// 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_);
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_);
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_);
// 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_);
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_;
// Notifies attached AecDump of current configuration and capture data.
void RecordUnprocessedCaptureStream(const float* const* capture_stream)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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();
void RecordUnprocessedCaptureStream(const int16_t* const data,
const StreamConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// 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);
// 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_);
rtc::scoped_ptr<FileWrapper> debug_file_;
rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
std::string event_str_; // Memory for protobuf serialization.
void RecordProcessedCaptureStream(const int16_t* const data,
const StreamConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Serialized string of last saved APM configuration.
std::string last_serialized_config_;
#endif
// Notifies attached AecDump about current state (delay, drift, etc).
void RecordAudioProcessingState()
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
// Format of processing streams at input/output call sites.
ProcessingConfig api_format_;
// 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_;
// 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_;
// 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_);
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_;
// Critical sections.
mutable Mutex mutex_render_ RTC_ACQUIRED_BEFORE(mutex_capture_);
mutable Mutex mutex_capture_;
bool output_will_be_muted_ GUARDED_BY(crit_);
// Struct containing the Config specifying the behavior of APM.
AudioProcessing::Config config_;
bool key_pressed_;
// Overrides for testing the exclusion of some submodules from the build.
ApmSubmoduleCreationOverrides submodule_creation_overrides_
RTC_GUARDED_BY(mutex_capture_);
// 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_;
// Class containing information about what submodules are active.
SubmoduleStates submodule_states_;
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_;
// 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_;
bool intelligibility_enabled_;
rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
// 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_