AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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112
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -16,13 +16,17 @@
#include <atomic>
#include <list>
#include <memory>
#include <optional>
#include <string>
#include <vector>
#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.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"
@ -35,15 +39,10 @@
#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"
@ -72,12 +71,14 @@ class AudioProcessingImpl : public AudioProcessing {
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;
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;
@ -154,24 +155,12 @@ class AudioProcessingImpl : public AudioProcessing {
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 {
@ -191,49 +180,6 @@ class AudioProcessingImpl : public AudioProcessing {
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_;
@ -254,10 +200,8 @@ class AudioProcessingImpl : public AudioProcessing {
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 echo_controller_enabled);
bool CaptureMultiBandSubModulesActive() const;
bool CaptureMultiBandProcessingPresent() const;
bool CaptureMultiBandProcessingActive(bool ec_processing_active) const;
@ -276,11 +220,9 @@ class AudioProcessingImpl : public AudioProcessing {
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;
};
@ -317,18 +259,9 @@ class AudioProcessingImpl : public AudioProcessing {
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_);
@ -423,10 +356,6 @@ class AudioProcessingImpl : public AudioProcessing {
// 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_;
@ -448,12 +377,10 @@ class AudioProcessingImpl : public AudioProcessing {
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_;
@ -479,19 +406,16 @@ class AudioProcessingImpl : public AudioProcessing {
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)
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),
transient_suppressor_forced_off(transient_suppressor_forced_off) {}
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;
bool transient_suppressor_forced_off;
} constants_;
struct ApmCaptureState {
@ -516,12 +440,12 @@ class AudioProcessingImpl : public AudioProcessing {
AudioProcessingStats stats;
// Input volume applied on the audio input device when the audio is
// acquired. Unspecified when unknown.
absl::optional<int> applied_input_volume;
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.
absl::optional<int> recommended_input_volume;
std::optional<int> recommended_input_volume;
} capture_ RTC_GUARDED_BY(mutex_capture_);
struct ApmCaptureNonLockedState {