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APM: Add field trial parameters and rename

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Add AGC2 digital adaptive config parameters in the field trial
"WebRTC-Audio-InputVolumeControllerExperiment". Rename it as
"WebRTC-Audio-GainController2" to reflect that the override now adjusts
the parameters for both input volume controller and adaptive digital
controller.

Bug: webrtc:7494
Change-Id: Ifbc1b8be76cf23b0b6b74b22b5167a45972cab38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286880
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38855}
This commit is contained in:
Hanna Silen 2022-12-08 17:40:01 +01:00 committed by WebRTC LUCI CQ
parent 6ebf5e3379
commit ca6535593f
3 changed files with 160 additions and 66 deletions
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148
modules/audio_processing/audio_processing_impl.cc
View File

@ -325,83 +325,124 @@ int HandleUnsupportedAudioFormats(const float* const* src,
return error_code;
}
const absl::optional<InputVolumeController::Config>
GetInputVolumeControllerConfigOverride() {
constexpr char kInputVolumeControllerFieldTrial[] =
"WebRTC-Audio-InputVolumeControllerExperiment";
const absl::optional<AudioProcessingImpl::GainController2ConfigOverride>
GetGainController2ConfigOverride() {
constexpr char kFieldTrialName[] = "WebRTC-Audio-GainController2";
if (!field_trial::IsEnabled(kInputVolumeControllerFieldTrial)) {
if (!field_trial::IsEnabled(kFieldTrialName)) {
return absl::nullopt;
}
constexpr InputVolumeController::Config kDefaultConfig;
constexpr InputVolumeController::Config kDefaultInputVolumeControllerConfig;
FieldTrialFlag enabled("Enabled", false);
FieldTrialConstrained<int> clipped_level_min(
"clipped_level_min", kDefaultConfig.clipped_level_min, 0, 255);
"clipped_level_min",
kDefaultInputVolumeControllerConfig.clipped_level_min, 0, 255);
FieldTrialConstrained<int> clipped_level_step(
"clipped_level_step", kDefaultConfig.clipped_level_step, 0, 255);
"clipped_level_step",
kDefaultInputVolumeControllerConfig.clipped_level_step, 0, 255);
FieldTrialConstrained<double> clipped_ratio_threshold(
"clipped_ratio_threshold", kDefaultConfig.clipped_ratio_threshold, 0, 1);
"clipped_ratio_threshold",
kDefaultInputVolumeControllerConfig.clipped_ratio_threshold, 0, 1);
FieldTrialConstrained<int> clipped_wait_frames(
"clipped_wait_frames", kDefaultConfig.clipped_wait_frames, 0,
"clipped_wait_frames",
kDefaultInputVolumeControllerConfig.clipped_wait_frames, 0,
absl::nullopt);
FieldTrialParameter<bool> enable_clipping_predictor(
"enable_clipping_predictor", kDefaultConfig.enable_clipping_predictor);
"enable_clipping_predictor",
kDefaultInputVolumeControllerConfig.enable_clipping_predictor);
FieldTrialConstrained<int> target_range_max_dbfs(
"target_range_max_dbfs", kDefaultConfig.target_range_max_dbfs, -90, 30);
"target_range_max_dbfs",
kDefaultInputVolumeControllerConfig.target_range_max_dbfs, -90, 30);
FieldTrialConstrained<int> target_range_min_dbfs(
"target_range_min_dbfs", kDefaultConfig.target_range_min_dbfs, -90, 30);
"target_range_min_dbfs",
kDefaultInputVolumeControllerConfig.target_range_min_dbfs, -90, 30);
FieldTrialConstrained<int> update_input_volume_wait_frames(
"update_input_volume_wait_frames",
kDefaultConfig.update_input_volume_wait_frames, 0, absl::nullopt);
kDefaultInputVolumeControllerConfig.update_input_volume_wait_frames, 0,
absl::nullopt);
FieldTrialConstrained<double> speech_probability_threshold(
"speech_probability_threshold",
kDefaultConfig.speech_probability_threshold, 0, 1);
kDefaultInputVolumeControllerConfig.speech_probability_threshold, 0, 1);
FieldTrialConstrained<double> speech_ratio_threshold(
"speech_ratio_threshold", kDefaultConfig.speech_ratio_threshold, 0, 1);
"speech_ratio_threshold",
kDefaultInputVolumeControllerConfig.speech_ratio_threshold, 0, 1);
// Field-trial based override for the input volume controller config.
constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
kDefaultAdaptiveDigitalConfig;
FieldTrialConstrained<double> headroom_db(
"headroom_db", kDefaultAdaptiveDigitalConfig.headroom_db, 0,
absl::nullopt);
FieldTrialConstrained<double> max_gain_db(
"max_gain_db", kDefaultAdaptiveDigitalConfig.max_gain_db, 0,
absl::nullopt);
FieldTrialConstrained<double> max_gain_change_db_per_second(
"max_gain_change_db_per_second",
kDefaultAdaptiveDigitalConfig.max_gain_change_db_per_second, 0,
absl::nullopt);
FieldTrialConstrained<double> max_output_noise_level_dbfs(
"max_output_noise_level_dbfs",
kDefaultAdaptiveDigitalConfig.max_output_noise_level_dbfs, absl::nullopt,
0);
// Field-trial based override for the input volume controller and adaptive
// digital configs.
const std::string field_trial_name =
field_trial::FindFullName(kInputVolumeControllerFieldTrial);
field_trial::FindFullName(kFieldTrialName);
ParseFieldTrial({&enabled, &clipped_level_min, &clipped_level_step,
&clipped_ratio_threshold, &clipped_wait_frames,
&enable_clipping_predictor, &target_range_max_dbfs,
&target_range_min_dbfs, &update_input_volume_wait_frames,
&speech_probability_threshold, &speech_ratio_threshold},
&speech_probability_threshold, &speech_ratio_threshold,
&headroom_db, &max_gain_db, &max_gain_change_db_per_second,
&max_output_noise_level_dbfs},
field_trial_name);
// Checked already by `IsEnabled()` before parsing, therefore always true.
RTC_DCHECK(enabled);
return InputVolumeController::Config{
.clipped_level_min = static_cast<int>(clipped_level_min.Get()),
.clipped_level_step = static_cast<int>(clipped_level_step.Get()),
.clipped_ratio_threshold =
static_cast<float>(clipped_ratio_threshold.Get()),
.clipped_wait_frames = static_cast<int>(clipped_wait_frames.Get()),
.enable_clipping_predictor =
static_cast<bool>(enable_clipping_predictor.Get()),
.target_range_max_dbfs = static_cast<int>(target_range_max_dbfs.Get()),
.target_range_min_dbfs = static_cast<int>(target_range_min_dbfs.Get()),
.update_input_volume_wait_frames =
static_cast<int>(update_input_volume_wait_frames.Get()),
.speech_probability_threshold =
static_cast<float>(speech_probability_threshold.Get()),
.speech_ratio_threshold =
static_cast<float>(speech_ratio_threshold.Get()),
return AudioProcessingImpl::GainController2ConfigOverride{
InputVolumeController::Config{
.clipped_level_min = static_cast<int>(clipped_level_min.Get()),
.clipped_level_step = static_cast<int>(clipped_level_step.Get()),
.clipped_ratio_threshold =
static_cast<float>(clipped_ratio_threshold.Get()),
.clipped_wait_frames = static_cast<int>(clipped_wait_frames.Get()),
.enable_clipping_predictor =
static_cast<bool>(enable_clipping_predictor.Get()),
.target_range_max_dbfs =
static_cast<int>(target_range_max_dbfs.Get()),
.target_range_min_dbfs =
static_cast<int>(target_range_min_dbfs.Get()),
.update_input_volume_wait_frames =
static_cast<int>(update_input_volume_wait_frames.Get()),
.speech_probability_threshold =
static_cast<float>(speech_probability_threshold.Get()),
.speech_ratio_threshold =
static_cast<float>(speech_ratio_threshold.Get()),
},
AudioProcessingImpl::GainController2ConfigOverride::AdaptiveDigitalConfig{
.headroom_db = static_cast<float>(headroom_db.Get()),
.max_gain_db = static_cast<float>(max_gain_db.Get()),
.max_gain_change_db_per_second =
static_cast<float>(max_gain_change_db_per_second.Get()),
.max_output_noise_level_dbfs =
static_cast<float>(max_output_noise_level_dbfs.Get()),
},
};
}
// If `disallow_transient_supporessor_usage` is true, disables transient
// suppression. When `input_volume_controller_config_override` is specified,
// suppression. When `gain_controller2_config_override` is specified,
// switches all gain control to AGC2.
AudioProcessing::Config AdjustConfig(
const AudioProcessing::Config& config,
bool disallow_transient_supporessor_usage,
const absl::optional<InputVolumeController::Config>&
input_volume_controller_config_override) {
const absl::optional<AudioProcessingImpl::GainController2ConfigOverride>&
gain_controller2_config_override) {
AudioProcessing::Config adjusted_config = config;
// Override the transient suppressor configuration.
@ -410,15 +451,14 @@ AudioProcessing::Config AdjustConfig(
}
// Override the auto gain control configuration if the AGC1 analog gain
// controller is active and `input_volume_controller_config_override` is
// controller is active and `gain_controller2_config_override` is
// specified.
const bool agc1_analog_enabled =
config.gain_controller1.enabled &&
(config.gain_controller1.mode ==
AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
config.gain_controller1.analog_gain_controller.enabled);
if (agc1_analog_enabled &&
input_volume_controller_config_override.has_value()) {
if (agc1_analog_enabled && gain_controller2_config_override.has_value()) {
// Check that the unadjusted AGC config meets the preconditions.
const bool hybrid_agc_config_detected =
config.gain_controller1.enabled &&
@ -447,9 +487,23 @@ AudioProcessing::Config AdjustConfig(
} else {
adjusted_config.gain_controller1.enabled = false;
adjusted_config.gain_controller1.analog_gain_controller.enabled = false;
adjusted_config.gain_controller2.enabled = true;
adjusted_config.gain_controller2.adaptive_digital.enabled = true;
adjusted_config.gain_controller2.input_volume_controller.enabled = true;
auto& adjusted_adaptive_digital = // Alias.
adjusted_config.gain_controller2.adaptive_digital;
const auto& adaptive_digital_override = // Alias.
gain_controller2_config_override->adaptive_digital_config;
adjusted_adaptive_digital.headroom_db =
adaptive_digital_override.headroom_db;
adjusted_adaptive_digital.max_gain_db =
adaptive_digital_override.max_gain_db;
adjusted_adaptive_digital.max_gain_change_db_per_second =
adaptive_digital_override.max_gain_change_db_per_second;
adjusted_adaptive_digital.max_output_noise_level_dbfs =
adaptive_digital_override.max_output_noise_level_dbfs;
}
}
@ -593,8 +647,7 @@ AudioProcessingImpl::AudioProcessingImpl(
: data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
use_setup_specific_default_aec3_config_(
UseSetupSpecificDefaultAec3Congfig()),
input_volume_controller_config_override_(
GetInputVolumeControllerConfigOverride()),
gain_controller2_config_override_(GetGainController2ConfigOverride()),
use_denormal_disabler_(
!field_trial::IsEnabled("WebRTC-ApmDenormalDisablerKillSwitch")),
disallow_transient_supporessor_usage_(
@ -607,7 +660,7 @@ AudioProcessingImpl::AudioProcessingImpl(
echo_control_factory_(std::move(echo_control_factory)),
config_(AdjustConfig(config,
disallow_transient_supporessor_usage_,
input_volume_controller_config_override_)),
gain_controller2_config_override_)),
submodule_states_(!!capture_post_processor,
!!render_pre_processor,
!!capture_analyzer),
@ -844,7 +897,7 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
const auto adjusted_config =
AdjustConfig(config, disallow_transient_supporessor_usage_,
input_volume_controller_config_override_);
gain_controller2_config_override_);
RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: "
<< adjusted_config.ToString();
@ -2292,8 +2345,9 @@ void AudioProcessingImpl::InitializeGainController2(bool config_has_changed) {
transient_suppressor_vad_mode_ != TransientSuppressor::VadMode::kRnnVad;
submodules_.gain_controller2 = std::make_unique<GainController2>(
config_.gain_controller2,
input_volume_controller_config_override_.value_or(
InputVolumeController::Config{}),
gain_controller2_config_override_.has_value()
? gain_controller2_config_override_->input_volume_controller_config
: InputVolumeController::Config{},
proc_fullband_sample_rate_hz(), num_input_channels(), use_internal_vad);
submodules_.gain_controller2->SetCaptureOutputUsed(
capture_.capture_output_used);

22
modules/audio_processing/audio_processing_impl.h
View File

@ -138,6 +138,18 @@ class AudioProcessingImpl : public AudioProcessing {
AudioProcessing::Config GetConfig() const override;
// TODO(bugs.webrtc.org/7494): Remove when the related field trial is
// removed.
struct GainController2ConfigOverride {
InputVolumeController::Config input_volume_controller_config;
struct AdaptiveDigitalConfig {
float headroom_db;
float max_gain_db;
float max_gain_change_db_per_second;
float max_output_noise_level_dbfs;
} adaptive_digital_config;
};
protected:
// Overridden in a mock.
virtual void InitializeLocked()
@ -161,7 +173,7 @@ class AudioProcessingImpl : public AudioProcessing {
FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
BitexactWithDisabledModules);
FRIEND_TEST_ALL_PREFIXES(
AudioProcessingImplInputVolumeControllerExperimentParametrizedTest,
AudioProcessingImplGainController2FieldTrialParametrizedTest,
ConfigAdjustedWhenExperimentEnabled);
void set_stream_analog_level_locked(int level)
@ -192,10 +204,10 @@ class AudioProcessingImpl : public AudioProcessing {
const bool use_setup_specific_default_aec3_config_;
// TODO(bugs.webrtc.org/7494): Remove when the linked field trial is removed.
// Override base on the "WebRTC-Audio-InputVolumeControllerExperiment" field
// trial for the AGC2 input volume controller configuration.
const absl::optional<InputVolumeController::Config>
input_volume_controller_config_override_;
// Override based on the "WebRTC-Audio-GainController2" field trial for the
// AGC2 input volume controller and adaptive digital controller configuration.
const absl::optional<GainController2ConfigOverride>
gain_controller2_config_override_;
const bool use_denormal_disabler_;

56
modules/audio_processing/audio_processing_impl_unittest.cc
View File

@ -1235,10 +1235,12 @@ TEST(AudioProcessingImplTest,
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigAdjustedWhenExperimentEnabledAndAgc1AnalogEnabled) {
constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
kDefaultAdaptiveDigitalConfig;
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-InputVolumeControllerExperiment/"
"WebRTC-Audio-GainController2/"
"Enabled,"
"enable_clipping_predictor:true,"
"clipped_level_min:20,"
@ -1249,7 +1251,11 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
"target_range_min_dbfs:-70,"
"update_input_volume_wait_frames:80,"
"speech_probability_threshold:0.9,"
"speech_ratio_threshold:1.0/");
"speech_ratio_threshold:1.0,"
"headroom_db:10,"
"max_gain_db:20,"
"max_gain_change_db_per_second:3,"
"max_output_noise_level_dbfs:-40/");
AudioProcessingBuilderForTesting apm_builder;
@ -1275,6 +1281,8 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
EXPECT_TRUE(adjusted_config.gain_controller2.enabled);
EXPECT_TRUE(adjusted_config.gain_controller2.adaptive_digital.enabled);
EXPECT_TRUE(adjusted_config.gain_controller2.input_volume_controller.enabled);
EXPECT_NE(adjusted_config.gain_controller2.adaptive_digital,
kDefaultAdaptiveDigitalConfig);
// Change config back and compare.
adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
@ -1285,14 +1293,18 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
config.gain_controller2.adaptive_digital.enabled;
adjusted_config.gain_controller2.input_volume_controller.enabled =
config.gain_controller2.input_volume_controller.enabled;
adjusted_config.gain_controller2.adaptive_digital =
config.gain_controller2.adaptive_digital;
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigAdjustedWhenExperimentEnabledAndHybridAgcEnabled) {
constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
kDefaultAdaptiveDigitalConfig;
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-InputVolumeControllerExperiment/"
"WebRTC-Audio-GainController2/"
"Enabled,"
"enable_clipping_predictor:true,"
"clipped_level_min:20,"
@ -1303,7 +1315,11 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
"target_range_min_dbfs:-70,"
"update_input_volume_wait_frames:80,"
"speech_probability_threshold:0.9,"
"speech_ratio_threshold:1.0/");
"speech_ratio_threshold:1.0,"
"headroom_db:10,"
"max_gain_db:20,"
"max_gain_change_db_per_second:3,"
"max_output_noise_level_dbfs:-40/");
AudioProcessingBuilderForTesting apm_builder;
@ -1331,6 +1347,8 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
EXPECT_TRUE(adjusted_config.gain_controller2.enabled);
EXPECT_TRUE(adjusted_config.gain_controller2.adaptive_digital.enabled);
EXPECT_TRUE(adjusted_config.gain_controller2.input_volume_controller.enabled);
EXPECT_NE(adjusted_config.gain_controller2.adaptive_digital,
kDefaultAdaptiveDigitalConfig);
// Change config back and compare.
adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
@ -1341,14 +1359,16 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
config.gain_controller2.adaptive_digital.enabled;
adjusted_config.gain_controller2.input_volume_controller.enabled =
config.gain_controller2.input_volume_controller.enabled;
adjusted_config.gain_controller2.adaptive_digital =
config.gain_controller2.adaptive_digital;
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentEnabledAndAgc1AnalogNotEnabled) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-InputVolumeControllerExperiment/"
"WebRTC-Audio-GainController2/"
"Enabled,"
"enable_clipping_predictor:true,"
"clipped_level_min:20,"
@ -1359,7 +1379,11 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
"target_range_min_dbfs:-70,"
"update_input_volume_wait_frames:80,"
"speech_probability_threshold:0.9,"
"speech_ratio_threshold:1.0/");
"speech_ratio_threshold:1.0,"
"headroom_db:10,"
"max_gain_db:20,"
"max_gain_change_db_per_second:3,"
"max_output_noise_level_dbfs:-40/");
AudioProcessingBuilderForTesting apm_builder;
@ -1393,10 +1417,10 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentEnabledAndHybridAgcNotEnabled) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-InputVolumeControllerExperiment/"
"WebRTC-Audio-GainController2/"
"Enabled,"
"enable_clipping_predictor:true,"
"clipped_level_min:20,"
@ -1407,7 +1431,11 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
"target_range_min_dbfs:-70,"
"update_input_volume_wait_frames:80,"
"speech_probability_threshold:0.9,"
"speech_ratio_threshold:1.0/");
"speech_ratio_threshold:1.0,"
"headroom_db:10,"
"max_gain_db:20,"
"max_gain_change_db_per_second:3,"
"max_output_noise_level_dbfs:-40/");
AudioProcessingBuilderForTesting apm_builder;
@ -1443,7 +1471,7 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentNotEnabledAndAgc1AnalogEnabled) {
AudioProcessingBuilderForTesting apm_builder;
@ -1477,7 +1505,7 @@ TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentNotEnabledAndHybridAgcEnabled) {
AudioProcessingBuilderForTesting apm_builder;