APM: Add field trial parameters and rename
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}
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index c1a2756..3200ea4 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -325,83 +325,124 @@
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 @@
}
// 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 @@
} 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 @@
: 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 @@
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 @@
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 @@
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);
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index 66e98dc..189ed03 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -138,6 +138,18 @@
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 @@
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 @@
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_;
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 7a45c45..b394e93 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -1235,10 +1235,12 @@
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 @@
"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 @@
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 @@
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 @@
"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 @@
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 @@
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 @@
"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 @@
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 @@
"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 @@
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
-TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentNotEnabledAndAgc1AnalogEnabled) {
AudioProcessingBuilderForTesting apm_builder;
@@ -1477,7 +1505,7 @@
EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
}
-TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+TEST(AudioProcessingImplGainController2FieldTrialTest,
ConfigNotAdjustedWhenExperimentNotEnabledAndHybridAgcEnabled) {
AudioProcessingBuilderForTesting apm_builder;