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webrtc/modules/audio_processing/audio_processing_impl.h
Alessio Bazzica 40b5bd72d0 APM: fix TS initialization bugs with WebRTC-Audio-GainController2 
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.

This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
   disabled and TS is enabled
2. when the initial APM sample rate is different from the
   capture one and the VAD APM sub-module is not re-initialized

This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.

Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
2023-01-16 20:30:12 +00:00

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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/ignore_wundef.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_
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