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Reland "Deprecate the adaptive level controller"

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This is a reland of 6f37ed78d99daa36e964ff0a65b205f0916d9949

CQ dry run OK except for missing iOS swarming bots.
NOTRY=True

Original change's description:
> 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}

Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
This commit is contained in:
Sam Zackrisson
2018-03-05 15:59:06 +01:00
committed by Commit Bot
parent e6826d2461
commit ab1aee0be4
39 changed files with 11 additions and 2288 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
api/audio_options.h
View File

@ -43,7 +43,6 @@ struct AudioOptions {
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 @@ struct AudioOptions {
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 @@ struct AudioOptions {
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 @@ struct AudioOptions {
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 @@ struct AudioOptions {
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 @@ struct AudioOptions {
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;

9
api/mediaconstraintsinterface.cc
View File

@ -107,9 +107,6 @@ const char MediaConstraintsInterface::kExperimentalNoiseSuppression[] =
"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[] =
@ -247,9 +244,6 @@ void CopyConstraintsIntoAudioOptions(
ConstraintToOptional<bool>(
constraints, MediaConstraintsInterface::kIntelligibilityEnhancer,
&options->intelligibility_enhancer);
ConstraintToOptional<bool>(constraints,
MediaConstraintsInterface::kLevelControl,
&options->level_control);
ConstraintToOptional<bool>(constraints,
MediaConstraintsInterface::kHighpassFilter,
&options->highpass_filter);
@ -259,9 +253,6 @@ void CopyConstraintsIntoAudioOptions(
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);

3
api/mediaconstraintsinterface.h
View File

@ -74,9 +74,6 @@ class MediaConstraintsInterface {
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

15
media/engine/webrtcvoiceengine.cc
View File

@ -295,7 +295,6 @@ void WebRtcVoiceEngine::Init() {
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 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
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;
}

3
media/engine/webrtcvoiceengine.h
View File

@ -120,7 +120,7 @@ class WebRtcVoiceEngine final {
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 @@ class WebRtcVoiceEngine final {
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;

23
modules/audio_processing/BUILD.gn
View File

@ -79,27 +79,6 @@ rtc_static_library("audio_processing") {
"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",
@ -612,7 +591,6 @@ if (rtc_include_tests) {
"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",
@ -640,7 +618,6 @@ if (rtc_include_tests) {
sources = [
"audio_processing_performance_unittest.cc",
"level_controller/level_controller_complexity_unittest.cc",
]
deps = [
":audio_processing",

63
modules/audio_processing/audio_processing_impl.cc
View File

@ -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 AudioProcessingImpl::ApmSubmoduleStates::Update(
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 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update(
(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 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update(
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::CaptureMultiBandProcessingActive()
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 @@ struct AudioProcessingImpl::ApmPrivateSubmodules {
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 @@ AudioProcessingImpl::AudioProcessingImpl(
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 @@ int AudioProcessingImpl::InitializeLocked() {
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 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
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 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
#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 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
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 @@ bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
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::InitializeGainController2() {
}
}
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 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
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;";
}

4
modules/audio_processing/audio_processing_impl.h
View File

@ -169,7 +169,6 @@ class AudioProcessingImpl : public AudioProcessing {
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 @@ class AudioProcessingImpl : public AudioProcessing {
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 @@ class AudioProcessingImpl : public AudioProcessing {
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 @@ class AudioProcessingImpl : public AudioProcessing {
int stream_delay_ms;
bool beamformer_enabled;
bool intelligibility_enabled;
bool level_controller_enabled = false;
bool echo_controller_enabled = false;
} capture_nonlocked_;

94
modules/audio_processing/audio_processing_unittest.cc
View File

@ -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 @@ INSTANTIATE_TEST_CASE_P(
} // 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 @@ std::unique_ptr<AudioProcessing> CreateApm(bool use_AEC2) {
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);

10
modules/audio_processing/include/audio_processing.h
View File

@ -211,8 +211,8 @@ struct Intelligibility {
// 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 @@ class AudioProcessing : public rtc::RefCountInterface {
// 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;

2
modules/audio_processing/include/config.h
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@ -35,7 +35,7 @@ enum class ConfigOptionID {
kIntelligibility,
kEchoCanceller3, // Deprecated
kAecRefinedAdaptiveFilter,
kLevelControl
kLevelControl // Deprecated
};
// Class Config is designed to ease passing a set of options across webrtc code.

35
modules/audio_processing/level_controller/biquad_filter.cc
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@ -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

58
modules/audio_processing/level_controller/biquad_filter.h
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@ -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_

100
modules/audio_processing/level_controller/down_sampler.cc
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@ -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

40
modules/audio_processing/level_controller/down_sampler.h
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@ -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_

160
modules/audio_processing/level_controller/gain_applier.cc
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@ -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

42
modules/audio_processing/level_controller/gain_applier.h
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@ -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_

87
modules/audio_processing/level_controller/gain_selector.cc
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@ -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

40
modules/audio_processing/level_controller/gain_selector.h
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@ -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_

295
modules/audio_processing/level_controller/level_controller.cc
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@ -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

95
modules/audio_processing/level_controller/level_controller.h
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/*
* 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_

240
modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
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/*
* 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

23
modules/audio_processing/level_controller/level_controller_constants.h
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/*
* 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_

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modules/audio_processing/level_controller/level_controller_unittest.cc
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/*
* 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

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modules/audio_processing/level_controller/noise_level_estimator.cc
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/*
* 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

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modules/audio_processing/level_controller/noise_level_estimator.h
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/*
* 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_

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modules/audio_processing/level_controller/noise_spectrum_estimator.cc
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/*
* 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

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modules/audio_processing/level_controller/noise_spectrum_estimator.h
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/*
* 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_

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modules/audio_processing/level_controller/peak_level_estimator.cc
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/*
* 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

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modules/audio_processing/level_controller/peak_level_estimator.h
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/*
* 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_

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modules/audio_processing/level_controller/saturating_gain_estimator.cc
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/*
* 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

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modules/audio_processing/level_controller/saturating_gain_estimator.h
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/*
* 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_

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modules/audio_processing/level_controller/signal_classifier.cc
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/*
* 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

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modules/audio_processing/level_controller/signal_classifier.h
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/*
* 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_

4
modules/audio_processing/test/aec_dump_based_simulator.cc
View File

@ -473,10 +473,6 @@ void AecDumpBasedSimulator::HandleMessage(
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;
}

3
modules/audio_processing/test/audio_processing_simulator.cc
View File

@ -551,9 +551,6 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
}
echo_control_factory.reset(new EchoCanceller3Factory(cfg));
}
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;
}

1
modules/audio_processing/test/audio_processing_simulator.h
View File

@ -66,7 +66,6 @@ struct SimulationSettings {
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;

4
modules/audio_processing/test/audioproc_float.cc
View File

@ -121,9 +121,6 @@ DEFINE_int(drift_compensation,
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");
@ -264,7 +261,6 @@ SimulationSettings CreateSettings() {
&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,

25
modules/audio_processing/test/debug_dump_test.cc
View File

@ -484,31 +484,6 @@ TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
}
}
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.