Deprecate the adaptive level controller
Level control handled by default-on AGC.
Bug: none
Change-Id: I405daeceece12c896d41156b649fcfd556726f77
Reviewed-on: https://webrtc-review.googlesource.com/59682
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22305}diff --git a/api/audio_options.h b/api/audio_options.h
index d62e1f8..8d2880b 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -43,7 +43,6 @@
SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
SetFrom(&experimental_ns, change.experimental_ns);
SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
- SetFrom(&level_control, change.level_control);
SetFrom(&residual_echo_detector, change.residual_echo_detector);
SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
SetFrom(&tx_agc_digital_compression_gain,
@@ -52,8 +51,6 @@
SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
- SetFrom(&level_control_initial_peak_level_dbfs,
- change.level_control_initial_peak_level_dbfs);
}
bool operator==(const AudioOptions& o) const {
@@ -76,7 +73,6 @@
delay_agnostic_aec == o.delay_agnostic_aec &&
experimental_ns == o.experimental_ns &&
intelligibility_enhancer == o.intelligibility_enhancer &&
- level_control == o.level_control &&
residual_echo_detector == o.residual_echo_detector &&
tx_agc_target_dbov == o.tx_agc_target_dbov &&
tx_agc_digital_compression_gain ==
@@ -84,9 +80,7 @@
tx_agc_limiter == o.tx_agc_limiter &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
audio_network_adaptor == o.audio_network_adaptor &&
- audio_network_adaptor_config == o.audio_network_adaptor_config &&
- level_control_initial_peak_level_dbfs ==
- o.level_control_initial_peak_level_dbfs;
+ audio_network_adaptor_config == o.audio_network_adaptor_config;
}
bool operator!=(const AudioOptions& o) const { return !(*this == o); }
@@ -113,9 +107,6 @@
ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
ost << ToStringIfSet("experimental_ns", experimental_ns);
ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
- ost << ToStringIfSet("level_control", level_control);
- ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
- level_control_initial_peak_level_dbfs);
ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
ost << ToStringIfSet("tx_agc_digital_compression_gain",
@@ -161,9 +152,6 @@
rtc::Optional<bool> delay_agnostic_aec;
rtc::Optional<bool> experimental_ns;
rtc::Optional<bool> intelligibility_enhancer;
- rtc::Optional<bool> level_control;
- // Specifies an optional initialization value for the level controller.
- rtc::Optional<float> level_control_initial_peak_level_dbfs;
// Note that tx_agc_* only applies to non-experimental AGC.
rtc::Optional<bool> residual_echo_detector;
rtc::Optional<uint16_t> tx_agc_target_dbov;
diff --git a/api/mediaconstraintsinterface.cc b/api/mediaconstraintsinterface.cc
index 5e6b218..8358644 100644
--- a/api/mediaconstraintsinterface.cc
+++ b/api/mediaconstraintsinterface.cc
@@ -107,9 +107,6 @@
"googNoiseSuppression2";
const char MediaConstraintsInterface::kIntelligibilityEnhancer[] =
"intelligibilityEnhancer";
-const char MediaConstraintsInterface::kLevelControl[] = "levelControl";
-const char MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS[] =
- "levelControlInitialPeakLevelDBFS";
const char MediaConstraintsInterface::kHighpassFilter[] =
"googHighpassFilter";
const char MediaConstraintsInterface::kTypingNoiseDetection[] =
@@ -248,9 +245,6 @@
constraints, MediaConstraintsInterface::kIntelligibilityEnhancer,
&options->intelligibility_enhancer);
ConstraintToOptional<bool>(constraints,
- MediaConstraintsInterface::kLevelControl,
- &options->level_control);
- ConstraintToOptional<bool>(constraints,
MediaConstraintsInterface::kHighpassFilter,
&options->highpass_filter);
ConstraintToOptional<bool>(constraints,
@@ -259,9 +253,6 @@
ConstraintToOptional<bool>(constraints,
MediaConstraintsInterface::kAudioMirroring,
&options->stereo_swapping);
- ConstraintToOptional<float>(
- constraints, MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS,
- &options->level_control_initial_peak_level_dbfs);
ConstraintToOptional<std::string>(
constraints, MediaConstraintsInterface::kAudioNetworkAdaptorConfig,
&options->audio_network_adaptor_config);
diff --git a/api/mediaconstraintsinterface.h b/api/mediaconstraintsinterface.h
index 73e4619..90661b8 100644
--- a/api/mediaconstraintsinterface.h
+++ b/api/mediaconstraintsinterface.h
@@ -74,9 +74,6 @@
static const char kNoiseSuppression[]; // googNoiseSuppression
static const char kExperimentalNoiseSuppression[]; // googNoiseSuppression2
static const char kIntelligibilityEnhancer[]; // intelligibilityEnhancer
- static const char kLevelControl[]; // levelControl
- static const char
- kLevelControlInitialPeakLevelDBFS[]; // levelControlInitialPeakLevelDBFS
static const char kHighpassFilter[]; // googHighpassFilter
static const char kTypingNoiseDetection[]; // googTypingNoiseDetection
static const char kAudioMirroring[]; // googAudioMirroring
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 7d889f3..6cd8805 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -295,7 +295,6 @@
options.delay_agnostic_aec = false;
options.experimental_ns = false;
options.intelligibility_enhancer = false;
- options.level_control = false;
options.residual_echo_detector = true;
bool error = ApplyOptions(options);
RTC_DCHECK(error);
@@ -564,22 +563,8 @@
new webrtc::Intelligibility(*intelligibility_enhancer_));
}
- if (options.level_control) {
- level_control_ = options.level_control;
- }
-
webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();
- RTC_LOG(LS_INFO) << "Level control: "
- << (!!level_control_ ? *level_control_ : -1);
- if (level_control_) {
- apm_config.level_controller.enabled = *level_control_;
- if (options.level_control_initial_peak_level_dbfs) {
- apm_config.level_controller.initial_peak_level_dbfs =
- *options.level_control_initial_peak_level_dbfs;
- }
- }
-
if (options.highpass_filter) {
apm_config.high_pass_filter.enabled = *options.highpass_filter;
}
diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index 0c7baf5..fbf7953 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -120,7 +120,7 @@
webrtc::AgcConfig default_agc_config_;
// Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns
- // level controller, and intelligibility_enhancer values, and apply them
+ // and intelligibility_enhancer values, and apply them
// in case they are missing in the audio options. We need to do this because
// SetExtraOptions() will revert to defaults for options which are not
// provided.
@@ -128,7 +128,6 @@
rtc::Optional<bool> delay_agnostic_aec_;
rtc::Optional<bool> experimental_ns_;
rtc::Optional<bool> intelligibility_enhancer_;
- rtc::Optional<bool> level_control_;
// Jitter buffer settings for new streams.
size_t audio_jitter_buffer_max_packets_ = 50;
bool audio_jitter_buffer_fast_accelerate_ = false;
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 3dcea89..93d3ec6 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -79,27 +79,6 @@
"include/audio_processing.h",
"include/config.cc",
"include/config.h",
- "level_controller/biquad_filter.cc",
- "level_controller/biquad_filter.h",
- "level_controller/down_sampler.cc",
- "level_controller/down_sampler.h",
- "level_controller/gain_applier.cc",
- "level_controller/gain_applier.h",
- "level_controller/gain_selector.cc",
- "level_controller/gain_selector.h",
- "level_controller/level_controller.cc",
- "level_controller/level_controller.h",
- "level_controller/level_controller_constants.h",
- "level_controller/noise_level_estimator.cc",
- "level_controller/noise_level_estimator.h",
- "level_controller/noise_spectrum_estimator.cc",
- "level_controller/noise_spectrum_estimator.h",
- "level_controller/peak_level_estimator.cc",
- "level_controller/peak_level_estimator.h",
- "level_controller/saturating_gain_estimator.cc",
- "level_controller/saturating_gain_estimator.h",
- "level_controller/signal_classifier.cc",
- "level_controller/signal_classifier.h",
"level_estimator_impl.cc",
"level_estimator_impl.h",
"low_cut_filter.cc",
@@ -610,7 +589,6 @@
"echo_detector/moving_max_unittest.cc",
"echo_detector/normalized_covariance_estimator_unittest.cc",
"gain_control_unittest.cc",
- "level_controller/level_controller_unittest.cc",
"level_estimator_unittest.cc",
"low_cut_filter_unittest.cc",
"noise_suppression_unittest.cc",
@@ -638,7 +616,6 @@
sources = [
"audio_processing_performance_unittest.cc",
- "level_controller/level_controller_complexity_unittest.cc",
]
deps = [
":audio_processing",
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index f4b8dee..0caa142 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -37,7 +37,6 @@
#if WEBRTC_INTELLIGIBILITY_ENHANCER
#include "modules/audio_processing/intelligibility/intelligibility_enhancer.h"
#endif
-#include "modules/audio_processing/level_controller/level_controller.h"
#include "modules/audio_processing/level_estimator_impl.h"
#include "modules/audio_processing/low_cut_filter.h"
#include "modules/audio_processing/noise_suppression_impl.h"
@@ -188,7 +187,6 @@
bool beamformer_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
- bool level_controller_enabled,
bool echo_controller_enabled,
bool voice_activity_detector_enabled,
bool level_estimator_enabled,
@@ -208,7 +206,6 @@
(adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
changed |=
(gain_controller2_enabled != gain_controller2_enabled_);
- changed |= (level_controller_enabled != level_controller_enabled_);
changed |= (echo_controller_enabled != echo_controller_enabled_);
changed |= (level_estimator_enabled != level_estimator_enabled_);
changed |=
@@ -224,7 +221,6 @@
beamformer_enabled_ = beamformer_enabled;
adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
gain_controller2_enabled_ = gain_controller2_enabled;
- level_controller_enabled_ = level_controller_enabled;
echo_controller_enabled_ = echo_controller_enabled;
level_estimator_enabled_ = level_estimator_enabled;
voice_activity_detector_enabled_ = voice_activity_detector_enabled;
@@ -256,8 +252,7 @@
bool AudioProcessingImpl::ApmSubmoduleStates::CaptureFullBandProcessingActive()
const {
- return level_controller_enabled_ || gain_controller2_enabled_ ||
- capture_post_processor_enabled_;
+ return gain_controller2_enabled_ || capture_post_processor_enabled_;
}
bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()
@@ -314,7 +309,6 @@
std::unique_ptr<AgcManagerDirect> agc_manager;
std::unique_ptr<GainController2> gain_controller2;
std::unique_ptr<LowCutFilter> low_cut_filter;
- std::unique_ptr<LevelController> level_controller;
std::unique_ptr<EchoDetector> echo_detector;
std::unique_ptr<EchoControl> echo_controller;
std::unique_ptr<CustomProcessing> capture_post_processor;
@@ -440,10 +434,6 @@
private_submodules_->echo_detector.reset(new ResidualEchoDetector());
}
- // TODO(peah): Move this creation to happen only when the level controller
- // is enabled.
- private_submodules_->level_controller.reset(new LevelController());
-
// TODO(alessiob): Move the injected gain controller once injection is
// implemented.
private_submodules_->gain_controller2.reset(new GainController2());
@@ -602,7 +592,6 @@
proc_sample_rate_hz());
public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
public_submodules_->level_estimator->Initialize();
- InitializeLevelController();
InitializeResidualEchoDetector();
InitializeEchoController();
InitializeGainController2();
@@ -706,40 +695,16 @@
void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
config_ = config;
- bool config_ok = LevelController::Validate(config_.level_controller);
- if (!config_ok) {
- RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
- "level_controller: "
- << LevelController::ToString(config_.level_controller)
- << "\nReverting to default parameter set";
- config_.level_controller = AudioProcessing::Config::LevelController();
- }
-
// Run in a single-threaded manner when applying the settings.
rtc::CritScope cs_render(&crit_render_);
rtc::CritScope cs_capture(&crit_capture_);
- // TODO(peah): Replace the use of capture_nonlocked_.level_controller_enabled
- // with the value in config_ everywhere in the code.
- if (capture_nonlocked_.level_controller_enabled !=
- config_.level_controller.enabled) {
- capture_nonlocked_.level_controller_enabled =
- config_.level_controller.enabled;
- // TODO(peah): Remove the conditional initialization to always initialize
- // the level controller regardless of whether it is enabled or not.
- InitializeLevelController();
- }
- RTC_LOG(LS_INFO) << "Level controller activated: "
- << capture_nonlocked_.level_controller_enabled;
-
- private_submodules_->level_controller->ApplyConfig(config_.level_controller);
-
InitializeLowCutFilter();
RTC_LOG(LS_INFO) << "Highpass filter activated: "
<< config_.high_pass_filter.enabled;
- config_ok = GainController2::Validate(config_.gain_controller2);
+ const bool config_ok = GainController2::Validate(config_.gain_controller2);
if (!config_ok) {
RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
"Gain Controller 2: "
@@ -1259,13 +1224,11 @@
#if WEBRTC_INTELLIGIBILITY_ENHANCER
if (capture_nonlocked_.intelligibility_enabled) {
RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
- int gain_db = public_submodules_->gain_control->is_enabled() ?
- public_submodules_->gain_control->compression_gain_db() :
- 0;
- float gain = DbToRatio(gain_db);
- gain *= capture_nonlocked_.level_controller_enabled ?
- private_submodules_->level_controller->GetLastGain() :
- 1.f;
+ const int gain_db =
+ public_submodules_->gain_control->is_enabled()
+ ? public_submodules_->gain_control->compression_gain_db()
+ : 0;
+ const float gain = DbToRatio(gain_db);
public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
public_submodules_->noise_suppression->NoiseEstimate(), gain);
}
@@ -1335,10 +1298,6 @@
private_submodules_->gain_controller2->Process(capture_buffer);
}
- if (capture_nonlocked_.level_controller_enabled) {
- private_submodules_->level_controller->Process(capture_buffer);
- }
-
if (private_submodules_->capture_post_processor) {
private_submodules_->capture_post_processor->Process(capture_buffer);
}
@@ -1766,7 +1725,6 @@
capture_nonlocked_.beamformer_enabled,
public_submodules_->gain_control->is_enabled(),
config_.gain_controller2.enabled,
- capture_nonlocked_.level_controller_enabled,
capture_nonlocked_.echo_controller_enabled,
public_submodules_->voice_detection->is_enabled(),
public_submodules_->level_estimator->is_enabled(),
@@ -1832,10 +1790,6 @@
}
}
-void AudioProcessingImpl::InitializeLevelController() {
- private_submodules_->level_controller->Initialize(proc_sample_rate_hz());
-}
-
void AudioProcessingImpl::InitializeResidualEchoDetector() {
RTC_DCHECK(private_submodules_->echo_detector);
private_submodules_->echo_detector->Initialize(proc_sample_rate_hz(),
@@ -1938,9 +1892,6 @@
public_submodules_->echo_cancellation->GetExperimentsDescription();
// TODO(peah): Add semicolon-separated concatenations of experiment
// descriptions for other submodules.
- if (capture_nonlocked_.level_controller_enabled) {
- experiments_description += "LevelController;";
- }
if (constants_.agc_clipped_level_min != kClippedLevelMin) {
experiments_description += "AgcClippingLevelExperiment;";
}
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index e7c6621..55c47ac 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -169,7 +169,6 @@
bool beamformer_enabled,
bool adaptive_gain_controller_enabled,
bool gain_controller2_enabled,
- bool level_controller_enabled,
bool echo_controller_enabled,
bool voice_activity_detector_enabled,
bool level_estimator_enabled,
@@ -193,7 +192,6 @@
bool beamformer_enabled_ = false;
bool adaptive_gain_controller_enabled_ = false;
bool gain_controller2_enabled_ = false;
- bool level_controller_enabled_ = false;
bool echo_controller_enabled_ = false;
bool level_estimator_enabled_ = false;
bool voice_activity_detector_enabled_ = false;
@@ -233,7 +231,6 @@
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
int InitializeLocked(const ProcessingConfig& config)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
- void InitializeLevelController() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
void InitializeResidualEchoDetector()
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
void InitializeLowCutFilter() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
@@ -386,7 +383,6 @@
int stream_delay_ms;
bool beamformer_enabled;
bool intelligibility_enabled;
- bool level_controller_enabled = false;
bool echo_controller_enabled = false;
} capture_nonlocked_;
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index ecaeed3..89d6cb9 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -25,7 +25,6 @@
#include "modules/audio_processing/common.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
-#include "modules/audio_processing/level_controller/level_controller_constants.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/test_utils.h"
#include "modules/include/module_common_types.h"
@@ -2821,98 +2820,6 @@
} // namespace
-TEST(ApmConfiguration, DefaultBehavior) {
- // Verify that the level controller is default off, it can be activated using
- // the config, and that the default initial level is maintained after the
- // config has been applied.
- std::unique_ptr<AudioProcessingImpl> apm(
- new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
- AudioProcessing::Config config;
- EXPECT_FALSE(apm->config_.level_controller.enabled);
- // TODO(peah): Add test for the existence of the level controller object once
- // that is created only when that is specified in the config.
- // TODO(peah): Remove the testing for
- // apm->capture_nonlocked_.level_controller_enabled once the value in config_
- // is instead used to activate the level controller.
- EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
- EXPECT_NEAR(kTargetLcPeakLeveldBFS,
- apm->config_.level_controller.initial_peak_level_dbfs,
- std::numeric_limits<float>::epsilon());
- config.level_controller.enabled = true;
- apm->ApplyConfig(config);
- EXPECT_TRUE(apm->config_.level_controller.enabled);
- // TODO(peah): Add test for the existence of the level controller object once
- // that is created only when the that is specified in the config.
- // TODO(peah): Remove the testing for
- // apm->capture_nonlocked_.level_controller_enabled once the value in config_
- // is instead used to activate the level controller.
- EXPECT_TRUE(apm->capture_nonlocked_.level_controller_enabled);
- EXPECT_NEAR(kTargetLcPeakLeveldBFS,
- apm->config_.level_controller.initial_peak_level_dbfs,
- std::numeric_limits<float>::epsilon());
-}
-
-TEST(ApmConfiguration, ValidConfigBehavior) {
- // Verify that the initial level can be specified and is retained after the
- // config has been applied.
- std::unique_ptr<AudioProcessingImpl> apm(
- new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
- AudioProcessing::Config config;
- config.level_controller.initial_peak_level_dbfs = -50.f;
- apm->ApplyConfig(config);
- EXPECT_FALSE(apm->config_.level_controller.enabled);
- // TODO(peah): Add test for the existence of the level controller object once
- // that is created only when the that is specified in the config.
- // TODO(peah): Remove the testing for
- // apm->capture_nonlocked_.level_controller_enabled once the value in config_
- // is instead used to activate the level controller.
- EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
- EXPECT_NEAR(-50.f, apm->config_.level_controller.initial_peak_level_dbfs,
- std::numeric_limits<float>::epsilon());
-}
-
-TEST(ApmConfiguration, InValidConfigBehavior) {
- // Verify that the config is properly reset when nonproper values are applied
- // for the initial level.
-
- // Verify that the config is properly reset when the specified initial peak
- // level is too low.
- std::unique_ptr<AudioProcessingImpl> apm(
- new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
- AudioProcessing::Config config;
- config.level_controller.enabled = true;
- config.level_controller.initial_peak_level_dbfs = -101.f;
- apm->ApplyConfig(config);
- EXPECT_FALSE(apm->config_.level_controller.enabled);
- // TODO(peah): Add test for the existence of the level controller object once
- // that is created only when the that is specified in the config.
- // TODO(peah): Remove the testing for
- // apm->capture_nonlocked_.level_controller_enabled once the value in config_
- // is instead used to activate the level controller.
- EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
- EXPECT_NEAR(kTargetLcPeakLeveldBFS,
- apm->config_.level_controller.initial_peak_level_dbfs,
- std::numeric_limits<float>::epsilon());
-
- // Verify that the config is properly reset when the specified initial peak
- // level is too high.
- apm.reset(new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
- config = AudioProcessing::Config();
- config.level_controller.enabled = true;
- config.level_controller.initial_peak_level_dbfs = 1.f;
- apm->ApplyConfig(config);
- EXPECT_FALSE(apm->config_.level_controller.enabled);
- // TODO(peah): Add test for the existence of the level controller object once
- // that is created only when that is specified in the config.
- // TODO(peah): Remove the testing for
- // apm->capture_nonlocked_.level_controller_enabled once the value in config_
- // is instead used to activate the level controller.
- EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
- EXPECT_NEAR(kTargetLcPeakLeveldBFS,
- apm->config_.level_controller.initial_peak_level_dbfs,
- std::numeric_limits<float>::epsilon());
-}
-
TEST(ApmConfiguration, EnablePostProcessing) {
// Verify that apm uses a capture post processing module if one is provided.
webrtc::Config webrtc_config;
@@ -3007,7 +2914,6 @@
config.residual_echo_detector.enabled = true;
config.high_pass_filter.enabled = false;
config.gain_controller2.enabled = false;
- config.level_controller.enabled = false;
apm->ApplyConfig(config);
EXPECT_EQ(apm->gain_control()->Enable(false), 0);
EXPECT_EQ(apm->level_estimator()->Enable(false), 0);
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 7057f28..33ecf89 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -211,8 +211,8 @@
// AudioProcessing* apm = AudioProcessingBuilder().Create();
//
// AudioProcessing::Config config;
-// config.level_controller.enabled = true;
// config.high_pass_filter.enabled = true;
+// config.gain_controller2.enabled = true;
// apm->ApplyConfig(config)
//
// apm->echo_cancellation()->enable_drift_compensation(false);
@@ -262,14 +262,6 @@
// by changing the default values in the AudioProcessing::Config struct.
// The config is applied by passing the struct to the ApplyConfig method.
struct Config {
- struct LevelController {
- bool enabled = false;
-
- // Sets the initial peak level to use inside the level controller in order
- // to compute the signal gain. The unit for the peak level is dBFS and
- // the allowed range is [-100, 0].
- float initial_peak_level_dbfs = -6.0206f;
- } level_controller;
struct ResidualEchoDetector {
bool enabled = true;
} residual_echo_detector;
diff --git a/modules/audio_processing/include/config.h b/modules/audio_processing/include/config.h
index 7c34de8..7615f62 100644
--- a/modules/audio_processing/include/config.h
+++ b/modules/audio_processing/include/config.h
@@ -35,7 +35,7 @@
kIntelligibility,
kEchoCanceller3, // Deprecated
kAecRefinedAdaptiveFilter,
- kLevelControl
+ kLevelControl // Deprecated
};
// Class Config is designed to ease passing a set of options across webrtc code.
diff --git a/modules/audio_processing/level_controller/biquad_filter.cc b/modules/audio_processing/level_controller/biquad_filter.cc
deleted file mode 100644
index 5a4ddc8..0000000
--- a/modules/audio_processing/level_controller/biquad_filter.cc
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/biquad_filter.h"
-
-namespace webrtc {
-
-// This method applies a biquad filter to an input signal x to produce an
-// output signal y. The biquad coefficients are specified at the construction
-// of the object.
-void BiQuadFilter::Process(rtc::ArrayView<const float> x,
- rtc::ArrayView<float> y) {
- for (size_t k = 0; k < x.size(); ++k) {
- // Use temporary variable for x[k] to allow in-place function call
- // (that x and y refer to the same array).
- const float tmp = x[k];
- y[k] = coefficients_.b[0] * tmp + coefficients_.b[1] * biquad_state_.b[0] +
- coefficients_.b[2] * biquad_state_.b[1] -
- coefficients_.a[0] * biquad_state_.a[0] -
- coefficients_.a[1] * biquad_state_.a[1];
- biquad_state_.b[1] = biquad_state_.b[0];
- biquad_state_.b[0] = tmp;
- biquad_state_.a[1] = biquad_state_.a[0];
- biquad_state_.a[0] = y[k];
- }
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/biquad_filter.h b/modules/audio_processing/level_controller/biquad_filter.h
deleted file mode 100644
index dad104d..0000000
--- a/modules/audio_processing/level_controller/biquad_filter.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
-
-#include <vector>
-
-#include "api/array_view.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class BiQuadFilter {
- public:
- struct BiQuadCoefficients {
- float b[3];
- float a[2];
- };
-
- BiQuadFilter() = default;
-
- void Initialize(const BiQuadCoefficients& coefficients) {
- coefficients_ = coefficients;
- }
-
- // Produces a filtered output y of the input x. Both x and y need to
- // have the same length.
- void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
-
- private:
- struct BiQuadState {
- BiQuadState() {
- std::fill(b, b + arraysize(b), 0.f);
- std::fill(a, a + arraysize(a), 0.f);
- }
-
- float b[2];
- float a[2];
- };
-
- BiQuadState biquad_state_;
- BiQuadCoefficients coefficients_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(BiQuadFilter);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
diff --git a/modules/audio_processing/level_controller/down_sampler.cc b/modules/audio_processing/level_controller/down_sampler.cc
deleted file mode 100644
index a1702f4..0000000
--- a/modules/audio_processing/level_controller/down_sampler.cc
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/down_sampler.h"
-
-#include <string.h>
-#include <algorithm>
-#include <vector>
-
-#include "modules/audio_processing/include/audio_processing.h"
-#include "modules/audio_processing/level_controller/biquad_filter.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-namespace {
-
-// Bandlimiter coefficients computed based on that only
-// the first 40 bins of the spectrum for the downsampled
-// signal are used.
-// [B,A] = butter(2,(41/64*4000)/8000)
-const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
- {0.1455f, 0.2911f, 0.1455f},
- {-0.6698f, 0.2520f}};
-
-// [B,A] = butter(2,(41/64*4000)/16000)
-const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
- {0.0462f, 0.0924f, 0.0462f},
- {-1.3066f, 0.4915f}};
-
-// [B,A] = butter(2,(41/64*4000)/24000)
-const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
- {0.0226f, 0.0452f, 0.0226f},
- {-1.5320f, 0.6224f}};
-
-} // namespace
-
-DownSampler::DownSampler(ApmDataDumper* data_dumper)
- : data_dumper_(data_dumper) {
- Initialize(48000);
-}
-void DownSampler::Initialize(int sample_rate_hz) {
- RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate48kHz);
-
- sample_rate_hz_ = sample_rate_hz;
- down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
-
- /// Note that the down sampling filter is not used if the sample rate is 8
- /// kHz.
- if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) {
- low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
- } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
- low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
- } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
- low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
- }
-}
-
-void DownSampler::DownSample(rtc::ArrayView<const float> in,
- rtc::ArrayView<float> out) {
- data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
- RTC_DCHECK_EQ(sample_rate_hz_ * AudioProcessing::kChunkSizeMs / 1000,
- in.size());
- RTC_DCHECK_EQ(
- AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
- out.size());
- const size_t kMaxNumFrames =
- AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
- float x[kMaxNumFrames];
-
- // Band-limit the signal to 4 kHz.
- if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) {
- low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
-
- // Downsample the signal.
- size_t k = 0;
- for (size_t j = 0; j < out.size(); ++j) {
- RTC_DCHECK_GT(kMaxNumFrames, k);
- out[j] = x[k];
- k += down_sampling_factor_;
- }
- } else {
- std::copy(in.data(), in.data() + in.size(), out.data());
- }
-
- data_dumper_->DumpWav("lc_down_sampler_output", out,
- AudioProcessing::kSampleRate8kHz, 1);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/down_sampler.h b/modules/audio_processing/level_controller/down_sampler.h
deleted file mode 100644
index d650242..0000000
--- a/modules/audio_processing/level_controller/down_sampler.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
-
-#include "api/array_view.h"
-#include "modules/audio_processing/level_controller/biquad_filter.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-
-class DownSampler {
- public:
- explicit DownSampler(ApmDataDumper* data_dumper);
- void Initialize(int sample_rate_hz);
-
- void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
-
- private:
- ApmDataDumper* data_dumper_;
- int sample_rate_hz_;
- int down_sampling_factor_;
- BiQuadFilter low_pass_filter_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
diff --git a/modules/audio_processing/level_controller/gain_applier.cc b/modules/audio_processing/level_controller/gain_applier.cc
deleted file mode 100644
index 018f809..0000000
--- a/modules/audio_processing/level_controller/gain_applier.cc
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/gain_applier.h"
-
-#include <algorithm>
-
-#include "api/array_view.h"
-#include "rtc_base/checks.h"
-
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-
-namespace webrtc {
-namespace {
-
-const float kMaxSampleValue = 32767.f;
-const float kMinSampleValue = -32767.f;
-
-int CountSaturations(rtc::ArrayView<const float> in) {
- return std::count_if(in.begin(), in.end(), [](const float& v) {
- return v >= kMaxSampleValue || v <= kMinSampleValue;
- });
-}
-
-int CountSaturations(const AudioBuffer& audio) {
- int num_saturations = 0;
- for (size_t k = 0; k < audio.num_channels(); ++k) {
- num_saturations += CountSaturations(rtc::ArrayView<const float>(
- audio.channels_const_f()[k], audio.num_frames()));
- }
- return num_saturations;
-}
-
-void LimitToAllowedRange(rtc::ArrayView<float> x) {
- for (auto& v : x) {
- v = std::max(kMinSampleValue, v);
- v = std::min(kMaxSampleValue, v);
- }
-}
-
-void LimitToAllowedRange(AudioBuffer* audio) {
- for (size_t k = 0; k < audio->num_channels(); ++k) {
- LimitToAllowedRange(
- rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
- }
-}
-
-float ApplyIncreasingGain(float new_gain,
- float old_gain,
- float step_size,
- rtc::ArrayView<float> x) {
- RTC_DCHECK_LT(0.f, step_size);
- float gain = old_gain;
- for (auto& v : x) {
- gain = std::min(new_gain, gain + step_size);
- v *= gain;
- }
- return gain;
-}
-
-float ApplyDecreasingGain(float new_gain,
- float old_gain,
- float step_size,
- rtc::ArrayView<float> x) {
- RTC_DCHECK_GT(0.f, step_size);
- float gain = old_gain;
- for (auto& v : x) {
- gain = std::max(new_gain, gain + step_size);
- v *= gain;
- }
- return gain;
-}
-
-float ApplyConstantGain(float gain, rtc::ArrayView<float> x) {
- for (auto& v : x) {
- v *= gain;
- }
-
- return gain;
-}
-
-float ApplyGain(float new_gain,
- float old_gain,
- float increase_step_size,
- float decrease_step_size,
- rtc::ArrayView<float> x) {
- RTC_DCHECK_LT(0.f, increase_step_size);
- RTC_DCHECK_GT(0.f, decrease_step_size);
- if (new_gain == old_gain) {
- return ApplyConstantGain(new_gain, x);
- } else if (new_gain > old_gain) {
- return ApplyIncreasingGain(new_gain, old_gain, increase_step_size, x);
- } else {
- return ApplyDecreasingGain(new_gain, old_gain, decrease_step_size, x);
- }
-}
-
-} // namespace
-
-GainApplier::GainApplier(ApmDataDumper* data_dumper)
- : data_dumper_(data_dumper) {}
-
-void GainApplier::Initialize(int sample_rate_hz) {
- RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate48kHz);
- const float kGainIncreaseStepSize48kHz = 0.0001f;
- const float kGainDecreaseStepSize48kHz = -0.01f;
- const float kGainSaturatedDecreaseStepSize48kHz = -0.05f;
-
- last_frame_was_saturated_ = false;
- old_gain_ = 1.f;
- gain_increase_step_size_ =
- kGainIncreaseStepSize48kHz *
- (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
- gain_normal_decrease_step_size_ =
- kGainDecreaseStepSize48kHz *
- (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
- gain_saturated_decrease_step_size_ =
- kGainSaturatedDecreaseStepSize48kHz *
- (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
-}
-
-int GainApplier::Process(float new_gain, AudioBuffer* audio) {
- RTC_CHECK_NE(0.f, gain_increase_step_size_);
- RTC_CHECK_NE(0.f, gain_normal_decrease_step_size_);
- RTC_CHECK_NE(0.f, gain_saturated_decrease_step_size_);
- int num_saturations = 0;
- if (new_gain != 1.f) {
- float last_applied_gain = 1.f;
- float gain_decrease_step_size = last_frame_was_saturated_
- ? gain_saturated_decrease_step_size_
- : gain_normal_decrease_step_size_;
- for (size_t k = 0; k < audio->num_channels(); ++k) {
- last_applied_gain = ApplyGain(
- new_gain, old_gain_, gain_increase_step_size_,
- gain_decrease_step_size,
- rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
- }
-
- num_saturations = CountSaturations(*audio);
- LimitToAllowedRange(audio);
- old_gain_ = last_applied_gain;
- }
-
- data_dumper_->DumpRaw("lc_last_applied_gain", 1, &old_gain_);
-
- return num_saturations;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/gain_applier.h b/modules/audio_processing/level_controller/gain_applier.h
deleted file mode 100644
index 5669f45..0000000
--- a/modules/audio_processing/level_controller/gain_applier.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_GAIN_APPLIER_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
-
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-class AudioBuffer;
-
-class GainApplier {
- public:
- explicit GainApplier(ApmDataDumper* data_dumper);
- void Initialize(int sample_rate_hz);
-
- // Applies the specified gain to the audio frame and returns the resulting
- // number of saturated sample values.
- int Process(float new_gain, AudioBuffer* audio);
-
- private:
- ApmDataDumper* const data_dumper_;
- float old_gain_ = 1.f;
- float gain_increase_step_size_ = 0.f;
- float gain_normal_decrease_step_size_ = 0.f;
- float gain_saturated_decrease_step_size_ = 0.f;
- bool last_frame_was_saturated_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
diff --git a/modules/audio_processing/level_controller/gain_selector.cc b/modules/audio_processing/level_controller/gain_selector.cc
deleted file mode 100644
index 3ab75b1..0000000
--- a/modules/audio_processing/level_controller/gain_selector.cc
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/gain_selector.h"
-
-#include <math.h>
-#include <algorithm>
-
-#include "modules/audio_processing/include/audio_processing.h"
-#include "modules/audio_processing/level_controller/level_controller_constants.h"
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-
-GainSelector::GainSelector() {
- Initialize(AudioProcessing::kSampleRate48kHz);
-}
-
-void GainSelector::Initialize(int sample_rate_hz) {
- gain_ = 1.f;
- frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100);
- highly_nonstationary_signal_hold_counter_ = 0;
-}
-
-// Chooses the gain to apply by the level controller such that
-// 1) The level of the stationary noise does not exceed
-// a predefined threshold.
-// 2) The gain does not exceed the gain that has been found
-// to saturate the signal.
-// 3) The peak level achieves the target peak level.
-// 4) The gain is not below 1.
-// 4) The gain is 1 if the signal has been classified as stationary
-// for a long time.
-// 5) The gain is not above the maximum gain.
-float GainSelector::GetNewGain(float peak_level,
- float noise_energy,
- float saturating_gain,
- bool gain_jumpstart,
- SignalClassifier::SignalType signal_type) {
- RTC_DCHECK_LT(0.f, peak_level);
-
- if (signal_type == SignalClassifier::SignalType::kHighlyNonStationary ||
- gain_jumpstart) {
- highly_nonstationary_signal_hold_counter_ = 100;
- } else {
- highly_nonstationary_signal_hold_counter_ =
- std::max(0, highly_nonstationary_signal_hold_counter_ - 1);
- }
-
- float desired_gain;
- if (highly_nonstationary_signal_hold_counter_ > 0) {
- // Compute a desired gain that ensures that the peak level is amplified to
- // the target level.
- desired_gain = kTargetLcPeakLevel / peak_level;
-
- // Limit the desired gain so that it does not amplify the noise too much.
- float max_noise_energy = kMaxLcNoisePower * frame_length_;
- if (noise_energy * desired_gain * desired_gain > max_noise_energy) {
- RTC_DCHECK_LE(0.f, noise_energy);
- desired_gain = sqrtf(max_noise_energy / noise_energy);
- }
- } else {
- // If the signal has been stationary for a long while, apply a gain of 1 to
- // avoid amplifying pure noise.
- desired_gain = 1.0f;
- }
-
- // Smootly update the gain towards the desired gain.
- gain_ += 0.2f * (desired_gain - gain_);
-
- // Limit the gain to not exceed the maximum and the saturating gains, and to
- // ensure that the lowest possible gain is 1.
- gain_ = std::min(gain_, saturating_gain);
- gain_ = std::min(gain_, kMaxLcGain);
- gain_ = std::max(gain_, 1.f);
-
- return gain_;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/gain_selector.h b/modules/audio_processing/level_controller/gain_selector.h
deleted file mode 100644
index 7966c43..0000000
--- a/modules/audio_processing/level_controller/gain_selector.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
-
-#include "rtc_base/constructormagic.h"
-
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-
-namespace webrtc {
-
-class GainSelector {
- public:
- GainSelector();
- void Initialize(int sample_rate_hz);
- float GetNewGain(float peak_level,
- float noise_energy,
- float saturating_gain,
- bool gain_jumpstart,
- SignalClassifier::SignalType signal_type);
-
- private:
- float gain_;
- size_t frame_length_;
- int highly_nonstationary_signal_hold_counter_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(GainSelector);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
diff --git a/modules/audio_processing/level_controller/level_controller.cc b/modules/audio_processing/level_controller/level_controller.cc
deleted file mode 100644
index b7854a0..0000000
--- a/modules/audio_processing/level_controller/level_controller.cc
+++ /dev/null
@@ -1,295 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/level_controller.h"
-
-#include <math.h>
-#include <algorithm>
-#include <numeric>
-
-#include "api/array_view.h"
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/level_controller/gain_applier.h"
-#include "modules/audio_processing/level_controller/gain_selector.h"
-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-#include "rtc_base/arraysize.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
-#include "system_wrappers/include/metrics.h"
-
-namespace webrtc {
-namespace {
-
-void UpdateAndRemoveDcLevel(float forgetting_factor,
- float* dc_level,
- rtc::ArrayView<float> x) {
- RTC_DCHECK(!x.empty());
- float mean =
- std::accumulate(x.begin(), x.end(), 0.0f) / static_cast<float>(x.size());
- *dc_level += forgetting_factor * (mean - *dc_level);
-
- for (float& v : x) {
- v -= *dc_level;
- }
-}
-
-float FrameEnergy(const AudioBuffer& audio) {
- float energy = 0.f;
- for (size_t k = 0; k < audio.num_channels(); ++k) {
- float channel_energy =
- std::accumulate(audio.channels_const_f()[k],
- audio.channels_const_f()[k] + audio.num_frames(), 0.f,
- [](float a, float b) -> float { return a + b * b; });
- energy = std::max(channel_energy, energy);
- }
- return energy;
-}
-
-float PeakLevel(const AudioBuffer& audio) {
- float peak_level = 0.f;
- for (size_t k = 0; k < audio.num_channels(); ++k) {
- auto* channel_peak_level = std::max_element(
- audio.channels_const_f()[k],
- audio.channels_const_f()[k] + audio.num_frames(),
- [](float a, float b) { return std::abs(a) < std::abs(b); });
- peak_level = std::max(*channel_peak_level, peak_level);
- }
- return peak_level;
-}
-
-const int kMetricsFrameInterval = 1000;
-
-} // namespace
-
-int LevelController::instance_count_ = 0;
-
-void LevelController::Metrics::Initialize(int sample_rate_hz) {
- RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate48kHz);
-
- Reset();
- frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100);
-}
-
-void LevelController::Metrics::Reset() {
- metrics_frame_counter_ = 0;
- gain_sum_ = 0.f;
- peak_level_sum_ = 0.f;
- noise_energy_sum_ = 0.f;
- max_gain_ = 0.f;
- max_peak_level_ = 0.f;
- max_noise_energy_ = 0.f;
-}
-
-void LevelController::Metrics::Update(float long_term_peak_level,
- float noise_energy,
- float gain,
- float frame_peak_level) {
- const float kdBFSOffset = 90.3090f;
- gain_sum_ += gain;
- peak_level_sum_ += long_term_peak_level;
- noise_energy_sum_ += noise_energy;
- max_gain_ = std::max(max_gain_, gain);
- max_peak_level_ = std::max(max_peak_level_, long_term_peak_level);
- max_noise_energy_ = std::max(max_noise_energy_, noise_energy);
-
- ++metrics_frame_counter_;
- if (metrics_frame_counter_ == kMetricsFrameInterval) {
- RTC_DCHECK_LT(0, frame_length_);
- RTC_DCHECK_LT(0, kMetricsFrameInterval);
-
- const int max_noise_power_dbfs = static_cast<int>(
- 10 * log10(max_noise_energy_ / frame_length_ + 1e-10f) - kdBFSOffset);
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxNoisePower",
- max_noise_power_dbfs, -90, 0, 50);
-
- const int average_noise_power_dbfs = static_cast<int>(
- 10 * log10(noise_energy_sum_ / (frame_length_ * kMetricsFrameInterval) +
- 1e-10f) -
- kdBFSOffset);
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageNoisePower",
- average_noise_power_dbfs, -90, 0, 50);
-
- const int max_peak_level_dbfs = static_cast<int>(
- 10 * log10(max_peak_level_ * max_peak_level_ + 1e-10f) - kdBFSOffset);
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxPeakLevel",
- max_peak_level_dbfs, -90, 0, 50);
-
- const int average_peak_level_dbfs = static_cast<int>(
- 10 * log10(peak_level_sum_ * peak_level_sum_ /
- (kMetricsFrameInterval * kMetricsFrameInterval) +
- 1e-10f) -
- kdBFSOffset);
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AveragePeakLevel",
- average_peak_level_dbfs, -90, 0, 50);
-
- RTC_DCHECK_LE(1.f, max_gain_);
- RTC_DCHECK_LE(1.f, gain_sum_ / kMetricsFrameInterval);
-
- const int max_gain_db = static_cast<int>(10 * log10(max_gain_ * max_gain_));
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxGain", max_gain_db, 0,
- 33, 30);
-
- const int average_gain_db = static_cast<int>(
- 10 * log10(gain_sum_ * gain_sum_ /
- (kMetricsFrameInterval * kMetricsFrameInterval)));
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageGain",
- average_gain_db, 0, 33, 30);
-
- const int long_term_peak_level_dbfs = static_cast<int>(
- 10 * log10(long_term_peak_level * long_term_peak_level + 1e-10f) -
- kdBFSOffset);
-
- const int frame_peak_level_dbfs = static_cast<int>(
- 10 * log10(frame_peak_level * frame_peak_level + 1e-10f) - kdBFSOffset);
-
- RTC_LOG(LS_INFO) << "Level Controller metrics: {Max noise power: "
- << max_noise_power_dbfs
- << " dBFS, Average noise power: "
- << average_noise_power_dbfs
- << " dBFS, Max long term peak level: "
- << max_peak_level_dbfs
- << " dBFS, Average long term peak level: "
- << average_peak_level_dbfs
- << " dBFS, Max gain: "
- << max_gain_db
- << " dB, Average gain: "
- << average_gain_db
- << " dB, Long term peak level: "
- << long_term_peak_level_dbfs
- << " dBFS, Last frame peak level: "
- << frame_peak_level_dbfs
- << " dBFS}";
-
- Reset();
- }
-}
-
-LevelController::LevelController()
- : data_dumper_(new ApmDataDumper(instance_count_)),
- gain_applier_(data_dumper_.get()),
- signal_classifier_(data_dumper_.get()),
- peak_level_estimator_(kTargetLcPeakLeveldBFS) {
- Initialize(AudioProcessing::kSampleRate48kHz);
- ++instance_count_;
-}
-
-LevelController::~LevelController() {}
-
-void LevelController::Initialize(int sample_rate_hz) {
- RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
- sample_rate_hz == AudioProcessing::kSampleRate48kHz);
- data_dumper_->InitiateNewSetOfRecordings();
- gain_selector_.Initialize(sample_rate_hz);
- gain_applier_.Initialize(sample_rate_hz);
- signal_classifier_.Initialize(sample_rate_hz);
- noise_level_estimator_.Initialize(sample_rate_hz);
- peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs);
- saturating_gain_estimator_.Initialize();
- metrics_.Initialize(sample_rate_hz);
-
- last_gain_ = 1.0f;
- sample_rate_hz_ = sample_rate_hz;
- dc_forgetting_factor_ = 0.01f * sample_rate_hz / 48000.f;
- std::fill(dc_level_, dc_level_ + arraysize(dc_level_), 0.f);
-}
-
-void LevelController::Process(AudioBuffer* audio) {
- RTC_DCHECK_LT(0, audio->num_channels());
- RTC_DCHECK_GE(2, audio->num_channels());
- RTC_DCHECK_NE(0.f, dc_forgetting_factor_);
- RTC_DCHECK(sample_rate_hz_);
- data_dumper_->DumpWav("lc_input", audio->num_frames(),
- audio->channels_const_f()[0], *sample_rate_hz_, 1);
-
- // Remove DC level.
- for (size_t k = 0; k < audio->num_channels(); ++k) {
- UpdateAndRemoveDcLevel(
- dc_forgetting_factor_, &dc_level_[k],
- rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
- }
-
- SignalClassifier::SignalType signal_type;
- signal_classifier_.Analyze(*audio, &signal_type);
- int tmp = static_cast<int>(signal_type);
- data_dumper_->DumpRaw("lc_signal_type", 1, &tmp);
-
- // Estimate the noise energy.
- float noise_energy =
- noise_level_estimator_.Analyze(signal_type, FrameEnergy(*audio));
-
- // Estimate the overall signal peak level.
- const float frame_peak_level = PeakLevel(*audio);
- const float long_term_peak_level =
- peak_level_estimator_.Analyze(signal_type, frame_peak_level);
-
- float saturating_gain = saturating_gain_estimator_.GetGain();
-
- // Compute the new gain to apply.
- last_gain_ =
- gain_selector_.GetNewGain(long_term_peak_level, noise_energy,
- saturating_gain, gain_jumpstart_, signal_type);
-
- // Unflag the jumpstart of the gain as it should only happen once.
- gain_jumpstart_ = false;
-
- // Apply the gain to the signal.
- int num_saturations = gain_applier_.Process(last_gain_, audio);
-
- // Estimate the gain that saturates the overall signal.
- saturating_gain_estimator_.Update(last_gain_, num_saturations);
-
- // Update the metrics.
- metrics_.Update(long_term_peak_level, noise_energy, last_gain_,
- frame_peak_level);
-
- data_dumper_->DumpRaw("lc_selected_gain", 1, &last_gain_);
- data_dumper_->DumpRaw("lc_noise_energy", 1, &noise_energy);
- data_dumper_->DumpRaw("lc_peak_level", 1, &long_term_peak_level);
- data_dumper_->DumpRaw("lc_saturating_gain", 1, &saturating_gain);
-
- data_dumper_->DumpWav("lc_output", audio->num_frames(),
- audio->channels_f()[0], *sample_rate_hz_, 1);
-}
-
-void LevelController::ApplyConfig(
- const AudioProcessing::Config::LevelController& config) {
- RTC_DCHECK(Validate(config));
- config_ = config;
- peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs);
- gain_jumpstart_ = true;
-}
-
-std::string LevelController::ToString(
- const AudioProcessing::Config::LevelController& config) {
- std::stringstream ss;
- ss << "{"
- << "enabled: " << (config.enabled ? "true" : "false") << ", "
- << "initial_peak_level_dbfs: " << config.initial_peak_level_dbfs << "}";
- return ss.str();
-}
-
-bool LevelController::Validate(
- const AudioProcessing::Config::LevelController& config) {
- return (config.initial_peak_level_dbfs <
- std::numeric_limits<float>::epsilon() &&
- config.initial_peak_level_dbfs >
- -(100.f + std::numeric_limits<float>::epsilon()));
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/level_controller.h b/modules/audio_processing/level_controller/level_controller.h
deleted file mode 100644
index 224b886..0000000
--- a/modules/audio_processing/level_controller/level_controller.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
-
-#include <memory>
-#include <vector>
-
-#include "api/optional.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "modules/audio_processing/level_controller/gain_applier.h"
-#include "modules/audio_processing/level_controller/gain_selector.h"
-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-class AudioBuffer;
-
-class LevelController {
- public:
- LevelController();
- ~LevelController();
-
- void Initialize(int sample_rate_hz);
- void Process(AudioBuffer* audio);
- float GetLastGain() { return last_gain_; }
-
- // TODO(peah): This method is a temporary solution as the the aim is to
- // instead apply the config inside the constructor. Therefore this is likely
- // to change.
- void ApplyConfig(const AudioProcessing::Config::LevelController& config);
- // Validates a config.
- static bool Validate(const AudioProcessing::Config::LevelController& config);
- // Dumps a config to a string.
- static std::string ToString(
- const AudioProcessing::Config::LevelController& config);
-
- private:
- class Metrics {
- public:
- Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
- void Initialize(int sample_rate_hz);
- void Update(float long_term_peak_level,
- float noise_level,
- float gain,
- float frame_peak_level);
-
- private:
- void Reset();
-
- size_t metrics_frame_counter_;
- float gain_sum_;
- float peak_level_sum_;
- float noise_energy_sum_;
- float max_gain_;
- float max_peak_level_;
- float max_noise_energy_;
- float frame_length_;
- };
-
- std::unique_ptr<ApmDataDumper> data_dumper_;
- GainSelector gain_selector_;
- GainApplier gain_applier_;
- SignalClassifier signal_classifier_;
- NoiseLevelEstimator noise_level_estimator_;
- PeakLevelEstimator peak_level_estimator_;
- SaturatingGainEstimator saturating_gain_estimator_;
- Metrics metrics_;
- rtc::Optional<int> sample_rate_hz_;
- static int instance_count_;
- float dc_level_[2];
- float dc_forgetting_factor_;
- float last_gain_;
- bool gain_jumpstart_ = false;
- AudioProcessing::Config::LevelController config_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
diff --git a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
deleted file mode 100644
index 83f6725..0000000
--- a/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
+++ /dev/null
@@ -1,240 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include <numeric>
-#include <vector>
-
-#include "api/array_view.h"
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "modules/audio_processing/level_controller/level_controller.h"
-#include "modules/audio_processing/test/audio_buffer_tools.h"
-#include "modules/audio_processing/test/bitexactness_tools.h"
-#include "modules/audio_processing/test/performance_timer.h"
-#include "modules/audio_processing/test/simulator_buffers.h"
-#include "rtc_base/random.h"
-#include "system_wrappers/include/clock.h"
-#include "test/gtest.h"
-#include "test/testsupport/perf_test.h"
-
-namespace webrtc {
-namespace {
-
-const size_t kNumFramesToProcess = 300;
-const size_t kNumFramesToProcessAtWarmup = 300;
-const size_t kToTalNumFrames =
- kNumFramesToProcess + kNumFramesToProcessAtWarmup;
-
-void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
- test::SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz,
- sample_rate_hz, num_channels, num_channels,
- num_channels, num_channels);
- test::PerformanceTimer timer(kNumFramesToProcess);
-
- LevelController level_controller;
- level_controller.Initialize(sample_rate_hz);
-
- for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
- buffers.UpdateInputBuffers();
-
- if (frame_no >= kNumFramesToProcessAtWarmup) {
- timer.StartTimer();
- }
- level_controller.Process(buffers.capture_input_buffer.get());
- if (frame_no >= kNumFramesToProcessAtWarmup) {
- timer.StopTimer();
- }
- }
- webrtc::test::PrintResultMeanAndError(
- "level_controller_call_durations",
- "_" + std::to_string(sample_rate_hz) + "Hz_" +
- std::to_string(num_channels) + "_channels",
- "StandaloneLevelControl", timer.GetDurationAverage(),
- timer.GetDurationStandardDeviation(), "us", false);
-}
-
-void RunTogetherWithApm(const std::string& test_description,
- int render_input_sample_rate_hz,
- int render_output_sample_rate_hz,
- int capture_input_sample_rate_hz,
- int capture_output_sample_rate_hz,
- size_t num_channels,
- bool use_mobile_aec,
- bool include_default_apm_processing) {
- test::SimulatorBuffers buffers(
- render_input_sample_rate_hz, capture_input_sample_rate_hz,
- render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels,
- num_channels, num_channels, num_channels);
- test::PerformanceTimer render_timer(kNumFramesToProcess);
- test::PerformanceTimer capture_timer(kNumFramesToProcess);
- test::PerformanceTimer total_timer(kNumFramesToProcess);
-
- webrtc::Config config;
- AudioProcessing::Config apm_config;
- if (include_default_apm_processing) {
- config.Set<DelayAgnostic>(new DelayAgnostic(true));
- config.Set<ExtendedFilter>(new ExtendedFilter(true));
- }
- apm_config.level_controller.enabled = true;
- apm_config.residual_echo_detector.enabled = include_default_apm_processing;
-
- std::unique_ptr<AudioProcessing> apm;
- apm.reset(AudioProcessingBuilder().Create(config));
- ASSERT_TRUE(apm.get());
- apm->ApplyConfig(apm_config);
-
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->gain_control()->Enable(include_default_apm_processing));
- if (use_mobile_aec) {
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->echo_cancellation()->Enable(false));
- ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable(
- include_default_apm_processing));
- } else {
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->echo_cancellation()->Enable(include_default_apm_processing));
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->echo_control_mobile()->Enable(false));
- }
- apm_config.high_pass_filter.enabled = include_default_apm_processing;
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->noise_suppression()->Enable(include_default_apm_processing));
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->voice_detection()->Enable(include_default_apm_processing));
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->level_estimator()->Enable(include_default_apm_processing));
-
- StreamConfig render_input_config(render_input_sample_rate_hz, num_channels,
- false);
- StreamConfig render_output_config(render_output_sample_rate_hz, num_channels,
- false);
- StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels,
- false);
- StreamConfig capture_output_config(capture_output_sample_rate_hz,
- num_channels, false);
-
- for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
- buffers.UpdateInputBuffers();
-
- if (frame_no >= kNumFramesToProcessAtWarmup) {
- total_timer.StartTimer();
- render_timer.StartTimer();
- }
- ASSERT_EQ(AudioProcessing::kNoError,
- apm->ProcessReverseStream(
- &buffers.render_input[0], render_input_config,
- render_output_config, &buffers.render_output[0]));
-
- if (frame_no >= kNumFramesToProcessAtWarmup) {
- render_timer.StopTimer();
-
- capture_timer.StartTimer();
- }
-
- ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
- ASSERT_EQ(
- AudioProcessing::kNoError,
- apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
- capture_output_config, &buffers.capture_output[0]));
-
- if (frame_no >= kNumFramesToProcessAtWarmup) {
- capture_timer.StopTimer();
- total_timer.StopTimer();
- }
- }
-
- webrtc::test::PrintResultMeanAndError(
- "level_controller_call_durations",
- "_" + std::to_string(render_input_sample_rate_hz) + "_" +
- std::to_string(render_output_sample_rate_hz) + "_" +
- std::to_string(capture_input_sample_rate_hz) + "_" +
- std::to_string(capture_output_sample_rate_hz) + "Hz_" +
- std::to_string(num_channels) + "_channels" + "_render",
- test_description, render_timer.GetDurationAverage(),
- render_timer.GetDurationStandardDeviation(), "us", false);
- webrtc::test::PrintResultMeanAndError(
- "level_controller_call_durations",
- "_" + std::to_string(render_input_sample_rate_hz) + "_" +
- std::to_string(render_output_sample_rate_hz) + "_" +
- std::to_string(capture_input_sample_rate_hz) + "_" +
- std::to_string(capture_output_sample_rate_hz) + "Hz_" +
- std::to_string(num_channels) + "_channels" + "_capture",
- test_description, capture_timer.GetDurationAverage(),
- capture_timer.GetDurationStandardDeviation(), "us", false);
- webrtc::test::PrintResultMeanAndError(
- "level_controller_call_durations",
- "_" + std::to_string(render_input_sample_rate_hz) + "_" +
- std::to_string(render_output_sample_rate_hz) + "_" +
- std::to_string(capture_input_sample_rate_hz) + "_" +
- std::to_string(capture_output_sample_rate_hz) + "Hz_" +
- std::to_string(num_channels) + "_channels" + "_total",
- test_description, total_timer.GetDurationAverage(),
- total_timer.GetDurationStandardDeviation(), "us", false);
-}
-
-} // namespace
-
-// TODO(peah): Reactivate once issue 7712 has been resolved.
-TEST(LevelControllerPerformanceTest, DISABLED_StandaloneProcessing) {
- int sample_rates_to_test[] = {
- AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz,
- AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz};
- for (auto sample_rate : sample_rates_to_test) {
- for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
- RunStandaloneSubmodule(sample_rate, num_channels);
- }
- }
-}
-
-void TestSomeSampleRatesWithApm(const std::string& test_name,
- bool use_mobile_agc,
- bool include_default_apm_processing) {
- // Test some stereo combinations first.
- size_t num_channels = 2;
- RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate16kHz,
- AudioProcessing::kSampleRate32kHz, num_channels,
- use_mobile_agc, include_default_apm_processing);
- RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
- AudioProcessing::kSampleRate8kHz, num_channels,
- use_mobile_agc, include_default_apm_processing);
- RunTogetherWithApm(test_name, 48000, 48000, 44100, 44100, num_channels,
- use_mobile_agc, include_default_apm_processing);
-
- // Then test mono combinations.
- num_channels = 1;
- RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
- AudioProcessing::kSampleRate48kHz, num_channels,
- use_mobile_agc, include_default_apm_processing);
-}
-
-// TODO(peah): Reactivate once issue 7712 has been resolved.
-#if !defined(WEBRTC_ANDROID)
-TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
-#else
-TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
-#endif
- // Run without default APM processing and desktop AGC.
- TestSomeSampleRatesWithApm("SimpleLevelControlViaApm", false, false);
-}
-
-// TODO(peah): Reactivate once issue 7712 has been resolved.
-#if !defined(WEBRTC_ANDROID)
-TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
-#else
-TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
-#endif
- bool include_default_apm_processing = true;
- TestSomeSampleRatesWithApm("LevelControlAndDefaultDesktopApm", false,
- include_default_apm_processing);
- TestSomeSampleRatesWithApm("LevelControlAndDefaultMobileApm", true,
- include_default_apm_processing);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/level_controller_constants.h b/modules/audio_processing/level_controller/level_controller_constants.h
deleted file mode 100644
index 6cf2cd4..0000000
--- a/modules/audio_processing/level_controller/level_controller_constants.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
-
-namespace webrtc {
-
-const float kMaxLcGain = 10;
-const float kMaxLcNoisePower = 100.f * 100.f;
-const float kTargetLcPeakLevel = 16384.f;
-const float kTargetLcPeakLeveldBFS = -6.0206f;
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
diff --git a/modules/audio_processing/level_controller/level_controller_unittest.cc b/modules/audio_processing/level_controller/level_controller_unittest.cc
deleted file mode 100644
index cb36ae0..0000000
--- a/modules/audio_processing/level_controller/level_controller_unittest.cc
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include <vector>
-
-#include "api/array_view.h"
-#include "api/optional.h"
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "modules/audio_processing/level_controller/level_controller.h"
-#include "modules/audio_processing/test/audio_buffer_tools.h"
-#include "modules/audio_processing/test/bitexactness_tools.h"
-#include "test/gtest.h"
-
-namespace webrtc {
-namespace {
-
-const int kNumFramesToProcess = 1000;
-
-// Processes a specified amount of frames, verifies the results and reports
-// any errors.
-void RunBitexactnessTest(int sample_rate_hz,
- size_t num_channels,
- rtc::Optional<float> initial_peak_level_dbfs,
- rtc::ArrayView<const float> output_reference) {
- LevelController level_controller;
- level_controller.Initialize(sample_rate_hz);
- if (initial_peak_level_dbfs) {
- AudioProcessing::Config::LevelController config;
- config.initial_peak_level_dbfs = *initial_peak_level_dbfs;
- level_controller.ApplyConfig(config);
- }
-
- int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
- const StreamConfig capture_config(sample_rate_hz, num_channels, false);
- AudioBuffer capture_buffer(
- capture_config.num_frames(), capture_config.num_channels(),
- capture_config.num_frames(), capture_config.num_channels(),
- capture_config.num_frames());
- test::InputAudioFile capture_file(
- test::GetApmCaptureTestVectorFileName(sample_rate_hz));
- std::vector<float> capture_input(samples_per_channel * num_channels);
- for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
- ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
- &capture_file, capture_input);
-
- test::CopyVectorToAudioBuffer(capture_config, capture_input,
- &capture_buffer);
-
- level_controller.Process(&capture_buffer);
- }
-
- // Extract test results.
- std::vector<float> capture_output;
- test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
- &capture_output);
-
- // Compare the output with the reference. Only the first values of the output
- // from last frame processed are compared in order not having to specify all
- // preceding frames as testvectors. As the algorithm being tested has a
- // memory, testing only the last frame implicitly also tests the preceeding
- // frames.
- const float kVectorElementErrorBound = 1.0f / 32768.0f;
- EXPECT_TRUE(test::VerifyDeinterleavedArray(
- capture_config.num_frames(), capture_config.num_channels(),
- output_reference, capture_output, kVectorElementErrorBound));
-}
-
-} // namespace
-
-TEST(LevelControllerConfig, ToString) {
- AudioProcessing::Config config;
- config.level_controller.enabled = true;
- config.level_controller.initial_peak_level_dbfs = -6.0206f;
- EXPECT_EQ("{enabled: true, initial_peak_level_dbfs: -6.0206}",
- LevelController::ToString(config.level_controller));
-
- config.level_controller.enabled = false;
- config.level_controller.initial_peak_level_dbfs = -50.f;
- EXPECT_EQ("{enabled: false, initial_peak_level_dbfs: -50}",
- LevelController::ToString(config.level_controller));
-}
-
-TEST(LevelControlBitExactnessTest, Mono8kHz) {
- const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f};
- RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Mono16kHz) {
- const float kOutputReference[] = {-0.013706f, -0.013215f, -0.013018f};
- RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Mono32kHz) {
- const float kOutputReference[] = {-0.014495f, -0.016425f, -0.016085f};
- RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1, rtc::nullopt,
- kOutputReference);
-}
-
-// TODO(peah): Investigate why this particular testcase differ between Android
-// and the rest of the platforms.
-TEST(LevelControlBitExactnessTest, Mono48kHz) {
-#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
- defined(WEBRTC_ANDROID))
- const float kOutputReference[] = {-0.014277f, -0.015180f, -0.017437f};
-#else
- const float kOutputReference[] = {-0.014306f, -0.015209f, -0.017466f};
-#endif
- RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Stereo8kHz) {
- const float kOutputReference[] = {-0.014063f, -0.008450f, -0.012159f,
- -0.051967f, -0.023202f, -0.047858f};
- RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Stereo16kHz) {
- const float kOutputReference[] = {-0.012714f, -0.005896f, -0.012220f,
- -0.053306f, -0.024549f, -0.051527f};
- RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Stereo32kHz) {
- const float kOutputReference[] = {-0.011764f, -0.007044f, -0.013472f,
- -0.053537f, -0.026322f, -0.056253f};
- RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, Stereo48kHz) {
- const float kOutputReference[] = {-0.010643f, -0.006334f, -0.011377f,
- -0.049088f, -0.023600f, -0.050465f};
- RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, rtc::nullopt,
- kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, MonoInitial48kHz) {
- const float kOutputReference[] = {-0.013884f, -0.014761f, -0.016951f};
- RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, -50,
- kOutputReference);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/noise_level_estimator.cc b/modules/audio_processing/level_controller/noise_level_estimator.cc
deleted file mode 100644
index abf4ea2..0000000
--- a/modules/audio_processing/level_controller/noise_level_estimator.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
-
-#include <algorithm>
-
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-
-namespace webrtc {
-
-NoiseLevelEstimator::NoiseLevelEstimator() {
- Initialize(AudioProcessing::kSampleRate48kHz);
-}
-
-NoiseLevelEstimator::~NoiseLevelEstimator() {}
-
-void NoiseLevelEstimator::Initialize(int sample_rate_hz) {
- noise_energy_ = 1.f;
- first_update_ = true;
- min_noise_energy_ = sample_rate_hz * 2.f * 2.f / 100.f;
- noise_energy_hold_counter_ = 0;
-}
-
-float NoiseLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
- float frame_energy) {
- if (frame_energy <= 0.f) {
- return noise_energy_;
- }
-
- if (first_update_) {
- // Initialize the noise energy to the frame energy.
- first_update_ = false;
- return noise_energy_ = std::max(frame_energy, min_noise_energy_);
- }
-
- // Update the noise estimate in a minimum statistics-type manner.
- if (signal_type == SignalClassifier::SignalType::kStationary) {
- if (frame_energy > noise_energy_) {
- // Leak the estimate upwards towards the frame energy if no recent
- // downward update.
- noise_energy_hold_counter_ = std::max(noise_energy_hold_counter_ - 1, 0);
-
- if (noise_energy_hold_counter_ == 0) {
- noise_energy_ = std::min(noise_energy_ * 1.01f, frame_energy);
- }
- } else {
- // Update smoothly downwards with a limited maximum update magnitude.
- noise_energy_ =
- std::max(noise_energy_ * 0.9f,
- noise_energy_ + 0.05f * (frame_energy - noise_energy_));
- noise_energy_hold_counter_ = 1000;
- }
- } else {
- // For a non-stationary signal, leak the estimate downwards in order to
- // avoid estimate locking due to incorrect signal classification.
- noise_energy_ = noise_energy_ * 0.99f;
- }
-
- // Ensure a minimum of the estimate.
- return noise_energy_ = std::max(noise_energy_, min_noise_energy_);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/noise_level_estimator.h b/modules/audio_processing/level_controller/noise_level_estimator.h
deleted file mode 100644
index 94ef673..0000000
--- a/modules/audio_processing/level_controller/noise_level_estimator.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
-
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class NoiseLevelEstimator {
- public:
- NoiseLevelEstimator();
- ~NoiseLevelEstimator();
- void Initialize(int sample_rate_hz);
- float Analyze(SignalClassifier::SignalType signal_type, float frame_energy);
-
- private:
- float min_noise_energy_ = 0.f;
- bool first_update_;
- float noise_energy_;
- int noise_energy_hold_counter_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(NoiseLevelEstimator);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
diff --git a/modules/audio_processing/level_controller/noise_spectrum_estimator.cc b/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
deleted file mode 100644
index 6e921c2..0000000
--- a/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
-
-#include <string.h>
-#include <algorithm>
-
-#include "api/array_view.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-#include "rtc_base/arraysize.h"
-
-namespace webrtc {
-namespace {
-constexpr float kMinNoisePower = 100.f;
-} // namespace
-
-NoiseSpectrumEstimator::NoiseSpectrumEstimator(ApmDataDumper* data_dumper)
- : data_dumper_(data_dumper) {
- Initialize();
-}
-
-void NoiseSpectrumEstimator::Initialize() {
- std::fill(noise_spectrum_, noise_spectrum_ + arraysize(noise_spectrum_),
- kMinNoisePower);
-}
-
-void NoiseSpectrumEstimator::Update(rtc::ArrayView<const float> spectrum,
- bool first_update) {
- RTC_DCHECK_EQ(65, spectrum.size());
-
- if (first_update) {
- // Initialize the noise spectral estimate with the signal spectrum.
- std::copy(spectrum.data(), spectrum.data() + spectrum.size(),
- noise_spectrum_);
- } else {
- // Smoothly update the noise spectral estimate towards the signal spectrum
- // such that the magnitude of the updates are limited.
- for (size_t k = 0; k < spectrum.size(); ++k) {
- if (noise_spectrum_[k] < spectrum[k]) {
- noise_spectrum_[k] = std::min(
- 1.01f * noise_spectrum_[k],
- noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k]));
- } else {
- noise_spectrum_[k] = std::max(
- 0.99f * noise_spectrum_[k],
- noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k]));
- }
- }
- }
-
- // Ensure that the noise spectal estimate does not become too low.
- for (auto& v : noise_spectrum_) {
- v = std::max(v, kMinNoisePower);
- }
-
- data_dumper_->DumpRaw("lc_noise_spectrum", 65, noise_spectrum_);
- data_dumper_->DumpRaw("lc_signal_spectrum", spectrum);
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/noise_spectrum_estimator.h b/modules/audio_processing/level_controller/noise_spectrum_estimator.h
deleted file mode 100644
index f10933e..0000000
--- a/modules/audio_processing/level_controller/noise_spectrum_estimator.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
-
-#include "api/array_view.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-
-class NoiseSpectrumEstimator {
- public:
- explicit NoiseSpectrumEstimator(ApmDataDumper* data_dumper);
- void Initialize();
- void Update(rtc::ArrayView<const float> spectrum, bool first_update);
-
- rtc::ArrayView<const float> GetNoiseSpectrum() const {
- return rtc::ArrayView<const float>(noise_spectrum_);
- }
-
- private:
- ApmDataDumper* data_dumper_;
- float noise_spectrum_[65];
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSpectrumEstimator);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
diff --git a/modules/audio_processing/level_controller/peak_level_estimator.cc b/modules/audio_processing/level_controller/peak_level_estimator.cc
deleted file mode 100644
index f602892..0000000
--- a/modules/audio_processing/level_controller/peak_level_estimator.cc
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
-
-#include <algorithm>
-
-#include "common_audio/include/audio_util.h"
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-
-namespace webrtc {
-namespace {
-
-constexpr float kMinLevel = 30.f;
-
-} // namespace
-
-PeakLevelEstimator::PeakLevelEstimator(float initial_peak_level_dbfs) {
- Initialize(initial_peak_level_dbfs);
-}
-
-PeakLevelEstimator::~PeakLevelEstimator() {}
-
-void PeakLevelEstimator::Initialize(float initial_peak_level_dbfs) {
- RTC_DCHECK_LE(-100.f, initial_peak_level_dbfs);
- RTC_DCHECK_GE(0.f, initial_peak_level_dbfs);
-
- peak_level_ = std::max(DbfsToFloatS16(initial_peak_level_dbfs), kMinLevel);
-
- hold_counter_ = 0;
- initialization_phase_ = true;
-}
-
-float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
- float frame_peak_level) {
- if (frame_peak_level == 0) {
- RTC_DCHECK_LE(kMinLevel, peak_level_);
- return peak_level_;
- }
-
- if (peak_level_ < frame_peak_level) {
- // Smoothly update the estimate upwards when the frame peak level is
- // higher than the estimate.
- peak_level_ += 0.1f * (frame_peak_level - peak_level_);
- hold_counter_ = 100;
- initialization_phase_ = false;
- } else {
- hold_counter_ = std::max(0, hold_counter_ - 1);
-
- // When the signal is highly non-stationary, update the estimate slowly
- // downwards if the estimate is lower than the frame peak level.
- if ((signal_type == SignalClassifier::SignalType::kHighlyNonStationary &&
- hold_counter_ == 0) ||
- initialization_phase_) {
- peak_level_ =
- std::max(peak_level_ + 0.01f * (frame_peak_level - peak_level_),
- peak_level_ * 0.995f);
- }
- }
-
- peak_level_ = std::max(peak_level_, kMinLevel);
-
- return peak_level_;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/peak_level_estimator.h b/modules/audio_processing/level_controller/peak_level_estimator.h
deleted file mode 100644
index 0aa55d2..0000000
--- a/modules/audio_processing/level_controller/peak_level_estimator.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
-
-#include "modules/audio_processing/level_controller/level_controller_constants.h"
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class PeakLevelEstimator {
- public:
- explicit PeakLevelEstimator(float initial_peak_level_dbfs);
- ~PeakLevelEstimator();
- void Initialize(float initial_peak_level_dbfs);
- float Analyze(SignalClassifier::SignalType signal_type,
- float frame_peak_level);
- private:
- float peak_level_;
- int hold_counter_;
- bool initialization_phase_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PeakLevelEstimator);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
diff --git a/modules/audio_processing/level_controller/saturating_gain_estimator.cc b/modules/audio_processing/level_controller/saturating_gain_estimator.cc
deleted file mode 100644
index 60110c6..0000000
--- a/modules/audio_processing/level_controller/saturating_gain_estimator.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
-
-#include <math.h>
-#include <algorithm>
-
-#include "modules/audio_processing/level_controller/level_controller_constants.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-
-namespace webrtc {
-
-SaturatingGainEstimator::SaturatingGainEstimator() {
- Initialize();
-}
-
-SaturatingGainEstimator::~SaturatingGainEstimator() {}
-
-void SaturatingGainEstimator::Initialize() {
- saturating_gain_ = kMaxLcGain;
- saturating_gain_hold_counter_ = 0;
-}
-
-void SaturatingGainEstimator::Update(float gain, int num_saturations) {
- bool too_many_saturations = (num_saturations > 2);
-
- if (too_many_saturations) {
- saturating_gain_ = 0.95f * gain;
- saturating_gain_hold_counter_ = 1000;
- } else {
- saturating_gain_hold_counter_ =
- std::max(0, saturating_gain_hold_counter_ - 1);
- if (saturating_gain_hold_counter_ == 0) {
- saturating_gain_ *= 1.001f;
- saturating_gain_ = std::min(kMaxLcGain, saturating_gain_);
- }
- }
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/saturating_gain_estimator.h b/modules/audio_processing/level_controller/saturating_gain_estimator.h
deleted file mode 100644
index 8980f4e..0000000
--- a/modules/audio_processing/level_controller/saturating_gain_estimator.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
-
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-
-class SaturatingGainEstimator {
- public:
- SaturatingGainEstimator();
- ~SaturatingGainEstimator();
- void Initialize();
- void Update(float gain, int num_saturations);
- float GetGain() const { return saturating_gain_; }
-
- private:
- float saturating_gain_;
- int saturating_gain_hold_counter_;
-
- RTC_DISALLOW_COPY_AND_ASSIGN(SaturatingGainEstimator);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
diff --git a/modules/audio_processing/level_controller/signal_classifier.cc b/modules/audio_processing/level_controller/signal_classifier.cc
deleted file mode 100644
index d2d5917..0000000
--- a/modules/audio_processing/level_controller/signal_classifier.cc
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * Copyright (c) 2016 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.
- */
-
-#include "modules/audio_processing/level_controller/signal_classifier.h"
-
-#include <algorithm>
-#include <numeric>
-#include <vector>
-
-#include "api/array_view.h"
-#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/level_controller/down_sampler.h"
-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
-#include "modules/audio_processing/logging/apm_data_dumper.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-namespace {
-
-void RemoveDcLevel(rtc::ArrayView<float> x) {
- RTC_DCHECK_LT(0, x.size());
- float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f);
- mean /= x.size();
-
- for (float& v : x) {
- v -= mean;
- }
-}
-
-void PowerSpectrum(const OouraFft* ooura_fft,
- rtc::ArrayView<const float> x,
- rtc::ArrayView<float> spectrum) {
- RTC_DCHECK_EQ(65, spectrum.size());
- RTC_DCHECK_EQ(128, x.size());
- float X[128];
- std::copy(x.data(), x.data() + x.size(), X);
- ooura_fft->Fft(X);
-
- float* X_p = X;
- RTC_DCHECK_EQ(X_p, &X[0]);
- spectrum[0] = (*X_p) * (*X_p);
- ++X_p;
- RTC_DCHECK_EQ(X_p, &X[1]);
- spectrum[64] = (*X_p) * (*X_p);
- for (int k = 1; k < 64; ++k) {
- ++X_p;
- RTC_DCHECK_EQ(X_p, &X[2 * k]);
- spectrum[k] = (*X_p) * (*X_p);
- ++X_p;
- RTC_DCHECK_EQ(X_p, &X[2 * k + 1]);
- spectrum[k] += (*X_p) * (*X_p);
- }
-}
-
-webrtc::SignalClassifier::SignalType ClassifySignal(
- rtc::ArrayView<const float> signal_spectrum,
- rtc::ArrayView<const float> noise_spectrum,
- ApmDataDumper* data_dumper) {
- int num_stationary_bands = 0;
- int num_highly_nonstationary_bands = 0;
-
- // Detect stationary and highly nonstationary bands.
- for (size_t k = 1; k < 40; k++) {
- if (signal_spectrum[k] < 3 * noise_spectrum[k] &&
- signal_spectrum[k] * 3 > noise_spectrum[k]) {
- ++num_stationary_bands;
- } else if (signal_spectrum[k] > 9 * noise_spectrum[k]) {
- ++num_highly_nonstationary_bands;
- }
- }
-
- data_dumper->DumpRaw("lc_num_stationary_bands", 1, &num_stationary_bands);
- data_dumper->DumpRaw("lc_num_highly_nonstationary_bands", 1,
- &num_highly_nonstationary_bands);
-
- // Use the detected number of bands to classify the overall signal
- // stationarity.
- if (num_stationary_bands > 15) {
- return SignalClassifier::SignalType::kStationary;
- } else if (num_highly_nonstationary_bands > 15) {
- return SignalClassifier::SignalType::kHighlyNonStationary;
- } else {
- return SignalClassifier::SignalType::kNonStationary;
- }
-}
-
-} // namespace
-
-SignalClassifier::FrameExtender::FrameExtender(size_t frame_size,
- size_t extended_frame_size)
- : x_old_(extended_frame_size - frame_size, 0.f) {}
-
-SignalClassifier::FrameExtender::~FrameExtender() = default;
-
-void SignalClassifier::FrameExtender::ExtendFrame(
- rtc::ArrayView<const float> x,
- rtc::ArrayView<float> x_extended) {
- RTC_DCHECK_EQ(x_old_.size() + x.size(), x_extended.size());
- std::copy(x_old_.data(), x_old_.data() + x_old_.size(), x_extended.data());
- std::copy(x.data(), x.data() + x.size(), x_extended.data() + x_old_.size());
- std::copy(x_extended.data() + x_extended.size() - x_old_.size(),
- x_extended.data() + x_extended.size(), x_old_.data());
-}
-
-SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper)
- : data_dumper_(data_dumper),
- down_sampler_(data_dumper_),
- noise_spectrum_estimator_(data_dumper_) {
- Initialize(AudioProcessing::kSampleRate48kHz);
-}
-SignalClassifier::~SignalClassifier() {}
-
-void SignalClassifier::Initialize(int sample_rate_hz) {
- down_sampler_.Initialize(sample_rate_hz);
- noise_spectrum_estimator_.Initialize();
- frame_extender_.reset(new FrameExtender(80, 128));
- sample_rate_hz_ = sample_rate_hz;
- initialization_frames_left_ = 2;
- consistent_classification_counter_ = 3;
- last_signal_type_ = SignalClassifier::SignalType::kNonStationary;
-}
-
-void SignalClassifier::Analyze(const AudioBuffer& audio,
- SignalType* signal_type) {
- RTC_DCHECK_EQ(audio.num_frames(), sample_rate_hz_ / 100);
-
- // Compute the signal power spectrum.
- float downsampled_frame[80];
- down_sampler_.DownSample(rtc::ArrayView<const float>(
- audio.channels_const_f()[0], audio.num_frames()),
- downsampled_frame);
- float extended_frame[128];
- frame_extender_->ExtendFrame(downsampled_frame, extended_frame);
- RemoveDcLevel(extended_frame);
- float signal_spectrum[65];
- PowerSpectrum(&ooura_fft_, extended_frame, signal_spectrum);
-
- // Classify the signal based on the estimate of the noise spectrum and the
- // signal spectrum estimate.
- *signal_type = ClassifySignal(signal_spectrum,
- noise_spectrum_estimator_.GetNoiseSpectrum(),
- data_dumper_);
-
- // Update the noise spectrum based on the signal spectrum.
- noise_spectrum_estimator_.Update(signal_spectrum,
- initialization_frames_left_ > 0);
-
- // Update the number of frames until a reliable signal spectrum is achieved.
- initialization_frames_left_ = std::max(0, initialization_frames_left_ - 1);
-
- if (last_signal_type_ == *signal_type) {
- consistent_classification_counter_ =
- std::max(0, consistent_classification_counter_ - 1);
- } else {
- last_signal_type_ = *signal_type;
- consistent_classification_counter_ = 3;
- }
-
- if (consistent_classification_counter_ > 0) {
- *signal_type = SignalClassifier::SignalType::kNonStationary;
- }
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/level_controller/signal_classifier.h b/modules/audio_processing/level_controller/signal_classifier.h
deleted file mode 100644
index 2be13fe..0000000
--- a/modules/audio_processing/level_controller/signal_classifier.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2016 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_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
-
-#include <memory>
-#include <vector>
-
-#include "api/array_view.h"
-#include "modules/audio_processing/level_controller/down_sampler.h"
-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
-#include "modules/audio_processing/utility/ooura_fft.h"
-#include "rtc_base/constructormagic.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-class AudioBuffer;
-
-class SignalClassifier {
- public:
- enum class SignalType { kHighlyNonStationary, kNonStationary, kStationary };
-
- explicit SignalClassifier(ApmDataDumper* data_dumper);
- ~SignalClassifier();
-
- void Initialize(int sample_rate_hz);
- void Analyze(const AudioBuffer& audio, SignalType* signal_type);
-
- private:
- class FrameExtender {
- public:
- FrameExtender(size_t frame_size, size_t extended_frame_size);
- ~FrameExtender();
-
- void ExtendFrame(rtc::ArrayView<const float> x,
- rtc::ArrayView<float> x_extended);
-
- private:
- std::vector<float> x_old_;
-
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
- };
-
- ApmDataDumper* const data_dumper_;
- DownSampler down_sampler_;
- std::unique_ptr<FrameExtender> frame_extender_;
- NoiseSpectrumEstimator noise_spectrum_estimator_;
- int sample_rate_hz_;
- int initialization_frames_left_;
- int consistent_classification_counter_;
- SignalType last_signal_type_;
- const OouraFft ooura_fft_;
- RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 6d0b07c..83e8531 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -473,10 +473,6 @@
new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
}
- if (settings_.use_lc) {
- apm_config.level_controller.enabled = *settings_.use_lc;
- }
-
if (settings_.use_ed) {
apm_config.residual_echo_detector.enabled = *settings_.use_ed;
}
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 82bffe4..b4c3525 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -328,9 +328,6 @@
if (settings_.use_aec3 && *settings_.use_aec3) {
echo_control_factory.reset(new EchoCanceller3Factory());
}
- if (settings_.use_lc) {
- apm_config.level_controller.enabled = *settings_.use_lc;
- }
if (settings_.use_hpf) {
apm_config.high_pass_filter.enabled = *settings_.use_hpf;
}
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 41a3f45..a6bdb90 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -66,7 +66,6 @@
rtc::Optional<bool> use_extended_filter;
rtc::Optional<bool> use_drift_compensation;
rtc::Optional<bool> use_aec3;
- rtc::Optional<bool> use_lc;
rtc::Optional<bool> use_experimental_agc;
rtc::Optional<int> aecm_routing_mode;
rtc::Optional<bool> use_aecm_comfort_noise;
diff --git a/modules/audio_processing/test/audioproc_float.cc b/modules/audio_processing/test/audioproc_float.cc
index c5229a4..554d6b4 100644
--- a/modules/audio_processing/test/audioproc_float.cc
+++ b/modules/audio_processing/test/audioproc_float.cc
@@ -121,9 +121,6 @@
DEFINE_int(aec3,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AEC mode AEC3");
-DEFINE_int(lc,
- kParameterNotSpecifiedValue,
- "Activate (1) or deactivate(0) the level control");
DEFINE_int(experimental_agc,
kParameterNotSpecifiedValue,
"Activate (1) or deactivate(0) the experimental AGC");
@@ -261,7 +258,6 @@
&settings.use_refined_adaptive_filter);
SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3);
- SetSettingIfFlagSet(FLAG_lc, &settings.use_lc);
SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc);
SetSettingIfSpecified(FLAG_aecm_routing_mode, &settings.aecm_routing_mode);
SetSettingIfFlagSet(FLAG_aecm_comfort_noise,
diff --git a/modules/audio_processing/test/debug_dump_test.cc b/modules/audio_processing/test/debug_dump_test.cc
index 56f47b0..4d3be48 100644
--- a/modules/audio_processing/test/debug_dump_test.cc
+++ b/modules/audio_processing/test/debug_dump_test.cc
@@ -484,31 +484,6 @@
}
}
-TEST_F(DebugDumpTest, VerifyLevelControllerExperimentalString) {
- Config config;
- AudioProcessing::Config apm_config;
- apm_config.level_controller.enabled = true;
- DebugDumpGenerator generator(config, apm_config);
- generator.StartRecording();
- generator.Process(100);
- generator.StopRecording();
-
- DebugDumpReplayer debug_dump_replayer_;
-
- ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
-
- while (const rtc::Optional<audioproc::Event> event =
- debug_dump_replayer_.GetNextEvent()) {
- debug_dump_replayer_.RunNextEvent();
- if (event->type() == audioproc::Event::CONFIG) {
- const audioproc::Config* msg = &event->config();
- ASSERT_TRUE(msg->has_experiments_description());
- EXPECT_PRED_FORMAT2(testing::IsSubstring, "LevelController",
- msg->experiments_description().c_str());
- }
- }
-}
-
TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) {
Config config;
// Arbitrarily set clipping gain to 17, which will never be the default.