APM: Add a field trial for input volume controller
Add a field trial WebRTC-Audio-InputVolumeControllerExperiment and
a mechanism to adjust the config accordingly. Pass the additional
input volume controller config to GainController2.
Bug: webrtc:7494
Change-Id: I3dd624df1f4774cb533417747627995e1f60aa68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284101
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38780}
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 8edf6fe..e1f6877 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -193,6 +193,7 @@
"../../rtc_base:sanitizer",
"../../rtc_base:swap_queue",
"../../rtc_base:timeutils",
+ "../../rtc_base/experiments:field_trial_parser",
"../../rtc_base/synchronization:mutex",
"../../rtc_base/system:rtc_export",
"../../system_wrappers",
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 96193fb..52f2fcb 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -31,6 +31,7 @@
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"
#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
@@ -144,8 +145,6 @@
audio.channels_const()[0] + audio.num_frames());
}
-constexpr int kUnspecifiedDataDumpInputVolume = -100;
-
// Options for gracefully handling processing errors.
enum class FormatErrorOutputOption {
kOutputExactCopyOfInput,
@@ -326,6 +325,125 @@
return error_code;
}
+const absl::optional<InputVolumeController::Config>
+GetInputVolumeControllerConfigOverride() {
+ constexpr char kInputVolumeControllerFieldTrial[] =
+ "WebRTC-Audio-InputVolumeControllerExperiment";
+
+ if (!field_trial::IsEnabled(kInputVolumeControllerFieldTrial)) {
+ return absl::nullopt;
+ }
+
+ constexpr InputVolumeController::Config kDefaultConfig;
+
+ FieldTrialFlag enabled("Enabled", false);
+ FieldTrialConstrained<int> clipped_level_min(
+ "clipped_level_min", kDefaultConfig.clipped_level_min, 0, 255);
+ FieldTrialConstrained<int> clipped_level_step(
+ "clipped_level_step", kDefaultConfig.clipped_level_step, 0, 255);
+ FieldTrialConstrained<double> clipped_ratio_threshold(
+ "clipped_ratio_threshold", kDefaultConfig.clipped_ratio_threshold, 0, 1);
+ FieldTrialConstrained<int> clipped_wait_frames(
+ "clipped_wait_frames", kDefaultConfig.clipped_wait_frames, 0,
+ absl::nullopt);
+ FieldTrialParameter<bool> enable_clipping_predictor(
+ "enable_clipping_predictor", kDefaultConfig.enable_clipping_predictor);
+ FieldTrialConstrained<int> target_range_max_dbfs(
+ "target_range_max_dbfs", kDefaultConfig.target_range_max_dbfs, -90, 30);
+ FieldTrialConstrained<int> target_range_min_dbfs(
+ "target_range_min_dbfs", kDefaultConfig.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);
+ FieldTrialConstrained<double> speech_probability_threshold(
+ "speech_probability_threshold",
+ kDefaultConfig.speech_probability_threshold, 0, 1);
+ FieldTrialConstrained<double> speech_ratio_threshold(
+ "speech_ratio_threshold", kDefaultConfig.speech_ratio_threshold, 0, 1);
+
+ // Field-trial based override for the input volume controller config.
+ const std::string field_trial_name =
+ field_trial::FindFullName(kInputVolumeControllerFieldTrial);
+
+ 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},
+ 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()),
+ };
+}
+
+// Switches all gain control to AGC2 if experimenting with input volume
+// controller.
+const AudioProcessing::Config AdjustConfig(
+ const AudioProcessing::Config& config,
+ const absl::optional<InputVolumeController::Config>&
+ input_volume_controller_config_override) {
+ const bool analog_agc_enabled =
+ config.gain_controller1.enabled &&
+ (config.gain_controller1.mode ==
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
+ config.gain_controller1.analog_gain_controller.enabled);
+
+ // Do not update the config if none of the analog AGCs is active
+ // regardless of the input volume controller override.
+ if (!analog_agc_enabled ||
+ !input_volume_controller_config_override.has_value()) {
+ return config;
+ }
+
+ const bool hybrid_agc_config_detected =
+ config.gain_controller1.enabled &&
+ config.gain_controller1.analog_gain_controller.enabled &&
+ !config.gain_controller1.analog_gain_controller.enable_digital_adaptive &&
+ config.gain_controller2.enabled &&
+ config.gain_controller2.adaptive_digital.enabled;
+
+ const bool full_agc1_config_detected =
+ config.gain_controller1.enabled &&
+ config.gain_controller1.analog_gain_controller.enabled &&
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive &&
+ !config.gain_controller2.enabled;
+
+ if (hybrid_agc_config_detected == full_agc1_config_detected ||
+ config.gain_controller2.input_volume_controller.enabled) {
+ RTC_LOG(LS_ERROR) << "Unexpected AGC config: Config not adjusted.";
+ return config;
+ }
+
+ AudioProcessing::Config adjusted_config = config;
+ 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;
+
+ return adjusted_config;
+}
+
+constexpr int kUnspecifiedDataDumpInputVolume = -100;
+
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
@@ -448,6 +566,8 @@
: data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
use_setup_specific_default_aec3_config_(
UseSetupSpecificDefaultAec3Congfig()),
+ input_volume_controller_config_override_(
+ GetInputVolumeControllerConfigOverride()),
use_denormal_disabler_(
!field_trial::IsEnabled("WebRTC-ApmDenormalDisablerKillSwitch")),
transient_suppressor_vad_mode_(GetTransientSuppressorVadMode()),
@@ -456,7 +576,7 @@
capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
render_runtime_settings_enqueuer_(&render_runtime_settings_),
echo_control_factory_(std::move(echo_control_factory)),
- config_(config),
+ config_(AdjustConfig(config, input_volume_controller_config_override_)),
submodule_states_(!!capture_post_processor,
!!render_pre_processor,
!!capture_analyzer),
@@ -490,6 +610,8 @@
RTC_LOG(LS_INFO) << "Denormal disabler unsupported";
}
+ RTC_LOG(LS_INFO) << "AudioProcessing: " << config_.ToString();
+
// Mark Echo Controller enabled if a factory is injected.
capture_nonlocked_.echo_controller_enabled =
static_cast<bool>(echo_control_factory_);
@@ -681,46 +803,57 @@
}
void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
- RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
-
// Run in a single-threaded manner when applying the settings.
MutexLock lock_render(&mutex_render_);
MutexLock lock_capture(&mutex_capture_);
+ // TODO(bugs.webrtc.org/7494): Replace `adjusted_config` with `config` after
+ // "WebRTC-Audio-InputVolumeControllerExperiment" field trial is removed.
+ const auto adjusted_config =
+ AdjustConfig(config, input_volume_controller_config_override_);
+
+ RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: "
+ << adjusted_config.ToString();
+
const bool pipeline_config_changed =
config_.pipeline.multi_channel_render !=
- config.pipeline.multi_channel_render ||
+ adjusted_config.pipeline.multi_channel_render ||
config_.pipeline.multi_channel_capture !=
- config.pipeline.multi_channel_capture ||
+ adjusted_config.pipeline.multi_channel_capture ||
config_.pipeline.maximum_internal_processing_rate !=
- config.pipeline.maximum_internal_processing_rate;
+ adjusted_config.pipeline.maximum_internal_processing_rate;
const bool aec_config_changed =
- config_.echo_canceller.enabled != config.echo_canceller.enabled ||
- config_.echo_canceller.mobile_mode != config.echo_canceller.mobile_mode;
+ config_.echo_canceller.enabled !=
+ adjusted_config.echo_canceller.enabled ||
+ config_.echo_canceller.mobile_mode !=
+ adjusted_config.echo_canceller.mobile_mode;
const bool agc1_config_changed =
- config_.gain_controller1 != config.gain_controller1;
+ config_.gain_controller1 != adjusted_config.gain_controller1;
const bool agc2_config_changed =
- config_.gain_controller2 != config.gain_controller2;
+ config_.gain_controller2 != adjusted_config.gain_controller2;
const bool ns_config_changed =
- config_.noise_suppression.enabled != config.noise_suppression.enabled ||
- config_.noise_suppression.level != config.noise_suppression.level;
+ config_.noise_suppression.enabled !=
+ adjusted_config.noise_suppression.enabled ||
+ config_.noise_suppression.level !=
+ adjusted_config.noise_suppression.level;
const bool ts_config_changed = config_.transient_suppression.enabled !=
- config.transient_suppression.enabled;
+ adjusted_config.transient_suppression.enabled;
const bool pre_amplifier_config_changed =
- config_.pre_amplifier.enabled != config.pre_amplifier.enabled ||
+ config_.pre_amplifier.enabled != adjusted_config.pre_amplifier.enabled ||
config_.pre_amplifier.fixed_gain_factor !=
- config.pre_amplifier.fixed_gain_factor;
+ adjusted_config.pre_amplifier.fixed_gain_factor;
const bool gain_adjustment_config_changed =
- config_.capture_level_adjustment != config.capture_level_adjustment;
+ config_.capture_level_adjustment !=
+ adjusted_config.capture_level_adjustment;
- config_ = config;
+ config_ = adjusted_config;
if (aec_config_changed) {
InitializeEchoController();
@@ -2123,8 +2256,10 @@
const bool use_internal_vad =
transient_suppressor_vad_mode_ != TransientSuppressor::VadMode::kRnnVad;
submodules_.gain_controller2 = std::make_unique<GainController2>(
- config_.gain_controller2, proc_fullband_sample_rate_hz(),
- num_input_channels(), use_internal_vad);
+ config_.gain_controller2,
+ input_volume_controller_config_override_.value_or(
+ 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 191a3ee..9a30c8b 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -160,6 +160,9 @@
ReinitializeTransientSuppressor);
FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
BitexactWithDisabledModules);
+ FRIEND_TEST_ALL_PREFIXES(
+ AudioProcessingImplInputVolumeControllerExperimentParametrizedTest,
+ ConfigAdjustedWhenExperimentEnabled);
void set_stream_analog_level_locked(int level)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
@@ -188,6 +191,12 @@
static std::atomic<int> instance_count_;
const bool use_setup_specific_default_aec3_config_;
+ // TODO(bugs.webrtc.org/7494): Remove the the config when the field trial is
+ // removed. "WebRTC-Audio-InputVolumeControllerExperiment" field trial
+ // override for the input volume controller config.
+ const absl::optional<InputVolumeController::Config>
+ input_volume_controller_config_override_;
+
const bool use_denormal_disabler_;
const TransientSuppressor::VadMode transient_suppressor_vad_mode_;
diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index fea7a8c..ea61dae 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -1188,4 +1188,282 @@
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigAdjustedWhenExperimentEnabledAndAgc1AnalogEnabled) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-InputVolumeControllerExperiment/"
+ "Enabled,"
+ "enable_clipping_predictor:true,"
+ "clipped_level_min:20,"
+ "clipped_level_step:30,"
+ "clipped_ratio_threshold:0.4,"
+ "clipped_wait_frames:50,"
+ "target_range_max_dbfs:-6,"
+ "target_range_min_dbfs:-70,"
+ "update_input_volume_wait_frames:80,"
+ "speech_probability_threshold:0.9,"
+ "speech_ratio_threshold:1.0/");
+
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with analog AGC1 enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
+ config.gain_controller2.enabled = false;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ // Expect the config to be adjusted.
+ EXPECT_FALSE(adjusted_config.gain_controller1.enabled);
+ EXPECT_FALSE(adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ 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);
+
+ // Change config back and compare.
+ adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
+ adjusted_config.gain_controller1.analog_gain_controller.enabled =
+ config.gain_controller1.analog_gain_controller.enabled;
+ adjusted_config.gain_controller2.enabled = config.gain_controller2.enabled;
+ adjusted_config.gain_controller2.adaptive_digital.enabled =
+ config.gain_controller2.adaptive_digital.enabled;
+ adjusted_config.gain_controller2.input_volume_controller.enabled =
+ config.gain_controller2.input_volume_controller.enabled;
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigAdjustedWhenExperimentEnabledAndHybridAgcEnabled) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-InputVolumeControllerExperiment/"
+ "Enabled,"
+ "enable_clipping_predictor:true,"
+ "clipped_level_min:20,"
+ "clipped_level_step:30,"
+ "clipped_ratio_threshold:0.4,"
+ "clipped_wait_frames:50,"
+ "target_range_max_dbfs:-6,"
+ "target_range_min_dbfs:-70,"
+ "update_input_volume_wait_frames:80,"
+ "speech_probability_threshold:0.9,"
+ "speech_ratio_threshold:1.0/");
+
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with hybrid AGC enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
+ false;
+ config.gain_controller2.enabled = true;
+ config.gain_controller2.adaptive_digital.enabled = true;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ // Expect the config to be adjusted.
+ EXPECT_FALSE(adjusted_config.gain_controller1.enabled);
+ EXPECT_FALSE(adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ 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);
+
+ // Change config back and compare.
+ adjusted_config.gain_controller1.enabled = config.gain_controller1.enabled;
+ adjusted_config.gain_controller1.analog_gain_controller.enabled =
+ config.gain_controller1.analog_gain_controller.enabled;
+ adjusted_config.gain_controller2.enabled = config.gain_controller2.enabled;
+ adjusted_config.gain_controller2.adaptive_digital.enabled =
+ config.gain_controller2.adaptive_digital.enabled;
+ adjusted_config.gain_controller2.input_volume_controller.enabled =
+ config.gain_controller2.input_volume_controller.enabled;
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigNotAdjustedWhenExperimentEnabledAndAgc1AnalogNotEnabled) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-InputVolumeControllerExperiment/"
+ "Enabled,"
+ "enable_clipping_predictor:true,"
+ "clipped_level_min:20,"
+ "clipped_level_step:30,"
+ "clipped_ratio_threshold:0.4,"
+ "clipped_wait_frames:50,"
+ "target_range_max_dbfs:-6,"
+ "target_range_min_dbfs:-70,"
+ "update_input_volume_wait_frames:80,"
+ "speech_probability_threshold:0.9,"
+ "speech_ratio_threshold:1.0/");
+
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with analog AGC1 not enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = false;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
+ config.gain_controller2.enabled = false;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ EXPECT_EQ(config.gain_controller1.enabled,
+ adjusted_config.gain_controller1.enabled);
+ EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
+ adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ EXPECT_EQ(config.gain_controller2.enabled,
+ adjusted_config.gain_controller2.enabled);
+ EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
+ adjusted_config.gain_controller2.adaptive_digital.enabled);
+ EXPECT_FALSE(
+ adjusted_config.gain_controller2.input_volume_controller.enabled);
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigNotAdjustedWhenExperimentEnabledAndHybridAgcNotEnabled) {
+ webrtc::test::ScopedFieldTrials field_trials(
+ "WebRTC-Audio-InputVolumeControllerExperiment/"
+ "Enabled,"
+ "enable_clipping_predictor:true,"
+ "clipped_level_min:20,"
+ "clipped_level_step:30,"
+ "clipped_ratio_threshold:0.4,"
+ "clipped_wait_frames:50,"
+ "target_range_max_dbfs:-6,"
+ "target_range_min_dbfs:-70,"
+ "update_input_volume_wait_frames:80,"
+ "speech_probability_threshold:0.9,"
+ "speech_ratio_threshold:1.0/");
+
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with hybrid AGC analog not enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = false;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
+ false;
+ config.gain_controller2.enabled = true;
+ config.gain_controller2.adaptive_digital.enabled = true;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ EXPECT_EQ(config.gain_controller1.enabled,
+ adjusted_config.gain_controller1.enabled);
+ EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
+ adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ EXPECT_EQ(config.gain_controller2.enabled,
+ adjusted_config.gain_controller2.enabled);
+ EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
+ adjusted_config.gain_controller2.adaptive_digital.enabled);
+ EXPECT_FALSE(
+ adjusted_config.gain_controller2.input_volume_controller.enabled);
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigNotAdjustedWhenExperimentNotEnabledAndAgc1AnalogEnabled) {
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with analog AGC1 analog enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive = true;
+ config.gain_controller2.enabled = false;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ EXPECT_EQ(config.gain_controller1.enabled,
+ adjusted_config.gain_controller1.enabled);
+ EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
+ adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ EXPECT_EQ(config.gain_controller2.enabled,
+ adjusted_config.gain_controller2.enabled);
+ EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
+ adjusted_config.gain_controller2.adaptive_digital.enabled);
+ EXPECT_FALSE(
+ adjusted_config.gain_controller2.input_volume_controller.enabled);
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
+TEST(AudioProcessingImplInputVolumeControllerExperimentTest,
+ ConfigNotAdjustedWhenExperimentNotEnabledAndHybridAgcEnabled) {
+ AudioProcessingBuilderForTesting apm_builder;
+
+ // Set a config with hybrid AGC enabled.
+ AudioProcessing::Config config;
+ config.gain_controller1.enabled = true;
+ config.gain_controller1.analog_gain_controller.enabled = true;
+ config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
+ false;
+ config.gain_controller2.enabled = true;
+ config.gain_controller2.adaptive_digital.enabled = true;
+ config.gain_controller1.mode =
+ AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+
+ EXPECT_FALSE(config.gain_controller2.input_volume_controller.enabled);
+
+ apm_builder.SetConfig(config);
+
+ auto apm = apm_builder.Create();
+ auto adjusted_config = apm->GetConfig();
+
+ EXPECT_EQ(config.gain_controller1.enabled,
+ adjusted_config.gain_controller1.enabled);
+ EXPECT_EQ(config.gain_controller1.analog_gain_controller.enabled,
+ adjusted_config.gain_controller1.analog_gain_controller.enabled);
+ EXPECT_EQ(config.gain_controller2.enabled,
+ adjusted_config.gain_controller2.enabled);
+ EXPECT_EQ(config.gain_controller2.adaptive_digital.enabled,
+ adjusted_config.gain_controller2.adaptive_digital.enabled);
+ EXPECT_FALSE(
+ adjusted_config.gain_controller2.input_volume_controller.enabled);
+
+ EXPECT_THAT(adjusted_config.ToString(), ::testing::StrEq(config.ToString()));
+}
+
} // namespace webrtc
diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc
index 70b598d..2a9c862 100644
--- a/modules/audio_processing/gain_controller2.cc
+++ b/modules/audio_processing/gain_controller2.cc
@@ -27,6 +27,7 @@
namespace {
using Agc2Config = AudioProcessing::Config::GainController2;
+using InputVolumeControllerConfig = InputVolumeController::Config;
constexpr int kLogLimiterStatsPeriodMs = 30'000;
constexpr int kFrameLengthMs = 10;
@@ -64,10 +65,10 @@
// Creates an input volume controller if `enabled` is true.
std::unique_ptr<InputVolumeController> CreateInputVolumeController(
bool enabled,
+ const InputVolumeControllerConfig& config,
int num_channels) {
if (enabled) {
- return std::make_unique<InputVolumeController>(
- num_channels, InputVolumeController::Config());
+ return std::make_unique<InputVolumeController>(num_channels, config);
}
return nullptr;
}
@@ -76,10 +77,12 @@
std::atomic<int> GainController2::instance_count_(0);
-GainController2::GainController2(const Agc2Config& config,
- int sample_rate_hz,
- int num_channels,
- bool use_internal_vad)
+GainController2::GainController2(
+ const Agc2Config& config,
+ const InputVolumeControllerConfig& input_volume_controller_config,
+ int sample_rate_hz,
+ int num_channels,
+ bool use_internal_vad)
: cpu_features_(GetAllowedCpuFeatures()),
data_dumper_(instance_count_.fetch_add(1) + 1),
fixed_gain_applier_(
@@ -92,6 +95,7 @@
&data_dumper_)),
input_volume_controller_(
CreateInputVolumeController(config.input_volume_controller.enabled,
+ input_volume_controller_config,
num_channels)),
limiter_(sample_rate_hz, &data_dumper_, /*histogram_name_prefix=*/"Agc2"),
calls_since_last_limiter_log_(0) {
diff --git a/modules/audio_processing/gain_controller2.h b/modules/audio_processing/gain_controller2.h
index 3341cd2..0d41eaa 100644
--- a/modules/audio_processing/gain_controller2.h
+++ b/modules/audio_processing/gain_controller2.h
@@ -34,10 +34,12 @@
public:
// Ctor. If `use_internal_vad` is true, an internal voice activity
// detector is used for digital adaptive gain.
- GainController2(const AudioProcessing::Config::GainController2& config,
- int sample_rate_hz,
- int num_channels,
- bool use_internal_vad);
+ GainController2(
+ const AudioProcessing::Config::GainController2& config,
+ const InputVolumeController::Config& input_volume_controller_config,
+ int sample_rate_hz,
+ int num_channels,
+ bool use_internal_vad);
GainController2(const GainController2&) = delete;
GainController2& operator=(const GainController2&) = delete;
~GainController2();
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index 7fb0c26..f7e5db2 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -33,6 +33,7 @@
using ::testing::Optional;
using Agc2Config = AudioProcessing::Config::GainController2;
+using InputVolumeControllerConfig = InputVolumeController::Config;
// Sets all the samples in `ab` to `value`.
void SetAudioBufferSamples(float value, AudioBuffer& ab) {
@@ -73,11 +74,25 @@
config.adaptive_digital.enabled = false;
config.fixed_digital.gain_db = fixed_gain_db;
EXPECT_TRUE(GainController2::Validate(config));
- return std::make_unique<GainController2>(config, sample_rate_hz,
- /*num_channels=*/1,
- /*use_internal_vad=*/true);
+ return std::make_unique<GainController2>(
+ config, InputVolumeControllerConfig{}, sample_rate_hz,
+ /*num_channels=*/1,
+ /*use_internal_vad=*/true);
}
+constexpr InputVolumeControllerConfig kTestInputVolumeControllerConfig{
+ .clipped_level_min = 20,
+ .clipped_level_step = 30,
+ .clipped_ratio_threshold = 0.4,
+ .clipped_wait_frames = 50,
+ .enable_clipping_predictor = true,
+ .target_range_max_dbfs = -6,
+ .target_range_min_dbfs = -70,
+ .update_input_volume_wait_frames = 100,
+ .speech_probability_threshold = 0.9,
+ .speech_ratio_threshold = 1,
+};
+
} // namespace
TEST(GainController2, CheckDefaultConfig) {
@@ -160,9 +175,41 @@
Agc2Config config;
config.input_volume_controller.enabled = false;
- auto gain_controller =
- std::make_unique<GainController2>(config, kSampleRateHz, kNumChannels,
- /*use_internal_vad=*/true);
+ auto gain_controller = std::make_unique<GainController2>(
+ config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels,
+ /*use_internal_vad=*/true);
+
+ EXPECT_FALSE(gain_controller->GetRecommendedInputVolume().has_value());
+
+ // Run AGC for a signal with no clipping or detected speech.
+ RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+ kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+ EXPECT_FALSE(gain_controller->GetRecommendedInputVolume().has_value());
+
+ // Run AGC for a signal with clipping.
+ RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+ kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+ EXPECT_FALSE(gain_controller->GetRecommendedInputVolume().has_value());
+}
+
+TEST(
+ GainController2,
+ CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabledAndSpecificConfigUsed) {
+ constexpr float kHighInputLevel = 32767.0f;
+ constexpr float kLowInputLevel = 1000.0f;
+ constexpr int kInitialInputVolume = 100;
+ constexpr int kNumChannels = 2;
+ constexpr int kNumFrames = 5;
+ constexpr int kSampleRateHz = 16000;
+
+ Agc2Config config;
+ config.input_volume_controller.enabled = false;
+
+ auto gain_controller = std::make_unique<GainController2>(
+ config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels,
+ /*use_internal_vad=*/true);
EXPECT_FALSE(gain_controller->GetRecommendedInputVolume().has_value());
@@ -192,9 +239,42 @@
config.input_volume_controller.enabled = true;
config.adaptive_digital.enabled = true;
- auto gain_controller =
- std::make_unique<GainController2>(config, kSampleRateHz, kNumChannels,
- /*use_internal_vad=*/true);
+ auto gain_controller = std::make_unique<GainController2>(
+ config, InputVolumeControllerConfig{}, kSampleRateHz, kNumChannels,
+ /*use_internal_vad=*/true);
+
+ EXPECT_TRUE(gain_controller->GetRecommendedInputVolume().has_value());
+
+ // Run AGC for a signal with no clipping or detected speech.
+ RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames,
+ kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+ EXPECT_TRUE(gain_controller->GetRecommendedInputVolume().has_value());
+
+ // Run AGC for a signal with clipping.
+ RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames,
+ kSampleRateHz, kNumChannels, kInitialInputVolume);
+
+ EXPECT_TRUE(gain_controller->GetRecommendedInputVolume().has_value());
+}
+
+TEST(
+ GainController2,
+ CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabledAndSpecificConfigUsed) {
+ constexpr float kHighInputLevel = 32767.0f;
+ constexpr float kLowInputLevel = 1000.0f;
+ constexpr int kInitialInputVolume = 100;
+ constexpr int kNumChannels = 2;
+ constexpr int kNumFrames = 5;
+ constexpr int kSampleRateHz = 16000;
+
+ Agc2Config config;
+ config.input_volume_controller.enabled = true;
+ config.adaptive_digital.enabled = true;
+
+ auto gain_controller = std::make_unique<GainController2>(
+ config, kTestInputVolumeControllerConfig, kSampleRateHz, kNumChannels,
+ /*use_internal_vad=*/true);
EXPECT_TRUE(gain_controller->GetRecommendedInputVolume().has_value());
@@ -214,7 +294,8 @@
// Checks that the default config is applied.
TEST(GainController2, ApplyDefaultConfig) {
auto gain_controller2 = std::make_unique<GainController2>(
- Agc2Config{}, /*sample_rate_hz=*/16000, /*num_channels=*/2,
+ Agc2Config{}, InputVolumeControllerConfig{},
+ /*sample_rate_hz=*/16000, /*num_channels=*/2,
/*use_internal_vad=*/true);
EXPECT_TRUE(gain_controller2.get());
}
@@ -330,7 +411,8 @@
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
- GainController2 agc2(config, kSampleRateHz, kStereo,
+ GainController2 agc2(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
test::InputAudioFile input_file(
@@ -385,9 +467,11 @@
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
- GainController2 agc2(config, kSampleRateHz, kStereo,
+ GainController2 agc2(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
- GainController2 agc2_reference(config, kSampleRateHz, kStereo,
+ GainController2 agc2_reference(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
test::InputAudioFile input_file(
@@ -452,9 +536,11 @@
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
- GainController2 agc2(config, kSampleRateHz, kStereo,
+ GainController2 agc2(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/false);
- GainController2 agc2_reference(config, kSampleRateHz, kStereo,
+ GainController2 agc2_reference(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
test::InputAudioFile input_file(
@@ -521,9 +607,11 @@
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
- GainController2 agc2(config, kSampleRateHz, kStereo,
+ GainController2 agc2(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/false);
- GainController2 agc2_reference(config, kSampleRateHz, kStereo,
+ GainController2 agc2_reference(config, /*input_volume_controller_config=*/{},
+ kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
VoiceActivityDetectorWrapper vad(config.adaptive_digital.vad_reset_period_ms,
GetAvailableCpuFeatures(), kSampleRateHz);