Deprecate WebRTC-Audio-GainController2 fieldtrial
Bug: webrtc:7494 Change-Id: I315a6e5d203a7f7f86e27d5b1b1f7dd72ccf1b08 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354100 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42468}
This commit is contained in:
Hanna Silen
WebRTC LUCI CQ
parent
b2c4f5469c
commit
7ee37cf839
@ -324,229 +324,10 @@ constexpr int kUnspecifiedDataDumpInputVolume = -100;
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// Throughout webrtc, it's assumed that success is represented by zero.
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static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
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absl::optional<AudioProcessingImpl::GainController2ExperimentParams>
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AudioProcessingImpl::GetGainController2ExperimentParams() {
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constexpr char kFieldTrialName[] = "WebRTC-Audio-GainController2";
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if (!field_trial::IsEnabled(kFieldTrialName)) {
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return absl::nullopt;
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}
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FieldTrialFlag enabled("Enabled", false);
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// Whether the gain control should switch to AGC2. Enabled by default.
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FieldTrialParameter<bool> switch_to_agc2("switch_to_agc2", true);
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// AGC2 input volume controller configuration.
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constexpr InputVolumeController::Config kDefaultInputVolumeControllerConfig;
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FieldTrialConstrained<int> min_input_volume(
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"min_input_volume", kDefaultInputVolumeControllerConfig.min_input_volume,
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0, 255);
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FieldTrialConstrained<int> clipped_level_min(
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"clipped_level_min",
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kDefaultInputVolumeControllerConfig.clipped_level_min, 0, 255);
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FieldTrialConstrained<int> clipped_level_step(
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"clipped_level_step",
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kDefaultInputVolumeControllerConfig.clipped_level_step, 0, 255);
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FieldTrialConstrained<double> clipped_ratio_threshold(
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"clipped_ratio_threshold",
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kDefaultInputVolumeControllerConfig.clipped_ratio_threshold, 0, 1);
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FieldTrialConstrained<int> clipped_wait_frames(
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"clipped_wait_frames",
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kDefaultInputVolumeControllerConfig.clipped_wait_frames, 0,
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absl::nullopt);
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FieldTrialParameter<bool> enable_clipping_predictor(
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"enable_clipping_predictor",
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kDefaultInputVolumeControllerConfig.enable_clipping_predictor);
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FieldTrialConstrained<int> target_range_max_dbfs(
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"target_range_max_dbfs",
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kDefaultInputVolumeControllerConfig.target_range_max_dbfs, -90, 30);
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FieldTrialConstrained<int> target_range_min_dbfs(
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"target_range_min_dbfs",
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kDefaultInputVolumeControllerConfig.target_range_min_dbfs, -90, 30);
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FieldTrialConstrained<int> update_input_volume_wait_frames(
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"update_input_volume_wait_frames",
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kDefaultInputVolumeControllerConfig.update_input_volume_wait_frames, 0,
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absl::nullopt);
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FieldTrialConstrained<double> speech_probability_threshold(
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"speech_probability_threshold",
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kDefaultInputVolumeControllerConfig.speech_probability_threshold, 0, 1);
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FieldTrialConstrained<double> speech_ratio_threshold(
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"speech_ratio_threshold",
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kDefaultInputVolumeControllerConfig.speech_ratio_threshold, 0, 1);
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// AGC2 adaptive digital controller configuration.
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constexpr AudioProcessing::Config::GainController2::AdaptiveDigital
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kDefaultAdaptiveDigitalConfig;
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FieldTrialConstrained<double> headroom_db(
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"headroom_db", kDefaultAdaptiveDigitalConfig.headroom_db, 0,
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absl::nullopt);
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FieldTrialConstrained<double> max_gain_db(
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"max_gain_db", kDefaultAdaptiveDigitalConfig.max_gain_db, 0,
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absl::nullopt);
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FieldTrialConstrained<double> initial_gain_db(
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"initial_gain_db", kDefaultAdaptiveDigitalConfig.initial_gain_db, 0,
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absl::nullopt);
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FieldTrialConstrained<double> max_gain_change_db_per_second(
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"max_gain_change_db_per_second",
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kDefaultAdaptiveDigitalConfig.max_gain_change_db_per_second, 0,
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absl::nullopt);
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FieldTrialConstrained<double> max_output_noise_level_dbfs(
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"max_output_noise_level_dbfs",
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kDefaultAdaptiveDigitalConfig.max_output_noise_level_dbfs, absl::nullopt,
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0);
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// Transient suppressor.
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FieldTrialParameter<bool> disallow_transient_suppressor_usage(
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"disallow_transient_suppressor_usage", false);
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// Field-trial based override for the input volume controller and adaptive
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// digital configs.
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ParseFieldTrial(
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{&enabled, &switch_to_agc2, &min_input_volume, &clipped_level_min,
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&clipped_level_step, &clipped_ratio_threshold, &clipped_wait_frames,
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&enable_clipping_predictor, &target_range_max_dbfs,
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&target_range_min_dbfs, &update_input_volume_wait_frames,
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&speech_probability_threshold, &speech_ratio_threshold, &headroom_db,
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&max_gain_db, &initial_gain_db, &max_gain_change_db_per_second,
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&max_output_noise_level_dbfs, &disallow_transient_suppressor_usage},
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field_trial::FindFullName(kFieldTrialName));
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// Checked already by `IsEnabled()` before parsing, therefore always true.
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RTC_DCHECK(enabled);
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const bool do_not_change_agc_config = !switch_to_agc2.Get();
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if (do_not_change_agc_config && !disallow_transient_suppressor_usage.Get()) {
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// Return an unspecifed value since, in this case, both the AGC2 and TS
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// configurations won't be adjusted.
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return absl::nullopt;
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}
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using Params = AudioProcessingImpl::GainController2ExperimentParams;
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if (do_not_change_agc_config) {
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// Return a value that leaves the AGC2 config unchanged and that always
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// disables TS.
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return Params{.agc2_config = absl::nullopt,
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.disallow_transient_suppressor_usage = true};
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}
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// Return a value that switches all the gain control to AGC2.
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return Params{
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.agc2_config =
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Params::Agc2Config{
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.input_volume_controller =
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{
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.min_input_volume = min_input_volume.Get(),
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.clipped_level_min = clipped_level_min.Get(),
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.clipped_level_step = clipped_level_step.Get(),
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.clipped_ratio_threshold =
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static_cast<float>(clipped_ratio_threshold.Get()),
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.clipped_wait_frames = clipped_wait_frames.Get(),
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.enable_clipping_predictor =
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enable_clipping_predictor.Get(),
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.target_range_max_dbfs = target_range_max_dbfs.Get(),
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.target_range_min_dbfs = target_range_min_dbfs.Get(),
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.update_input_volume_wait_frames =
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update_input_volume_wait_frames.Get(),
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.speech_probability_threshold = static_cast<float>(
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speech_probability_threshold.Get()),
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.speech_ratio_threshold =
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static_cast<float>(speech_ratio_threshold.Get()),
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},
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.adaptive_digital_controller =
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{
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.headroom_db = static_cast<float>(headroom_db.Get()),
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.max_gain_db = static_cast<float>(max_gain_db.Get()),
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.initial_gain_db =
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static_cast<float>(initial_gain_db.Get()),
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.max_gain_change_db_per_second = static_cast<float>(
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max_gain_change_db_per_second.Get()),
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.max_output_noise_level_dbfs =
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static_cast<float>(max_output_noise_level_dbfs.Get()),
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}},
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.disallow_transient_suppressor_usage =
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disallow_transient_suppressor_usage.Get()};
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}
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AudioProcessing::Config AudioProcessingImpl::AdjustConfig(
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const AudioProcessing::Config& config,
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const absl::optional<AudioProcessingImpl::GainController2ExperimentParams>&
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experiment_params) {
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if (!experiment_params.has_value() ||
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(!experiment_params->agc2_config.has_value() &&
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!experiment_params->disallow_transient_suppressor_usage)) {
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// When the experiment parameters are unspecified or when the AGC and TS
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// configuration are not overridden, return the unmodified configuration.
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return config;
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}
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AudioProcessing::Config adjusted_config = config;
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// Override the transient suppressor configuration.
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if (experiment_params->disallow_transient_suppressor_usage) {
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adjusted_config.transient_suppression.enabled = false;
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}
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// Override the auto gain control configuration if the AGC1 analog gain
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// controller is active and `experiment_params->agc2_config` is specified.
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const bool agc1_analog_enabled =
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config.gain_controller1.enabled &&
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(config.gain_controller1.mode ==
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AudioProcessing::Config::GainController1::kAdaptiveAnalog ||
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config.gain_controller1.analog_gain_controller.enabled);
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if (agc1_analog_enabled && experiment_params->agc2_config.has_value()) {
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// Check that the unadjusted AGC config meets the preconditions.
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const bool hybrid_agc_config_detected =
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config.gain_controller1.enabled &&
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config.gain_controller1.analog_gain_controller.enabled &&
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!config.gain_controller1.analog_gain_controller
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.enable_digital_adaptive &&
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config.gain_controller2.enabled &&
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config.gain_controller2.adaptive_digital.enabled;
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const bool full_agc1_config_detected =
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config.gain_controller1.enabled &&
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config.gain_controller1.analog_gain_controller.enabled &&
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config.gain_controller1.analog_gain_controller
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.enable_digital_adaptive &&
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!config.gain_controller2.enabled;
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const bool one_and_only_one_input_volume_controller =
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hybrid_agc_config_detected != full_agc1_config_detected;
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const bool agc2_input_volume_controller_enabled =
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config.gain_controller2.enabled &&
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config.gain_controller2.input_volume_controller.enabled;
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if (!one_and_only_one_input_volume_controller ||
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agc2_input_volume_controller_enabled) {
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RTC_LOG(LS_ERROR) << "Cannot adjust AGC config (precondition failed)";
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if (!one_and_only_one_input_volume_controller)
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RTC_LOG(LS_ERROR)
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<< "One and only one input volume controller must be enabled.";
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if (agc2_input_volume_controller_enabled)
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RTC_LOG(LS_ERROR)
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<< "The AGC2 input volume controller must be disabled.";
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} else {
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adjusted_config.gain_controller1.enabled = false;
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adjusted_config.gain_controller1.analog_gain_controller.enabled = false;
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adjusted_config.gain_controller2.enabled = true;
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adjusted_config.gain_controller2.input_volume_controller.enabled = true;
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adjusted_config.gain_controller2.adaptive_digital =
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experiment_params->agc2_config->adaptive_digital_controller;
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adjusted_config.gain_controller2.adaptive_digital.enabled = true;
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}
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}
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return adjusted_config;
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}
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bool AudioProcessingImpl::UseApmVadSubModule(
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const AudioProcessing::Config& config,
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const absl::optional<GainController2ExperimentParams>& experiment_params) {
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// The VAD as an APM sub-module is needed only in one case, that is when TS
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// and AGC2 are both enabled and when the AGC2 experiment is running and its
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// parameters require to fully switch the gain control to AGC2.
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return config.transient_suppression.enabled &&
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config.gain_controller2.enabled &&
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(config.gain_controller2.input_volume_controller.enabled ||
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config.gain_controller2.adaptive_digital.enabled) &&
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experiment_params.has_value() &&
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experiment_params->agc2_config.has_value();
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const AudioProcessing::Config& config) {
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// Without "WebRTC-Audio-GainController2" always return false.
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return false;
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}
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AudioProcessingImpl::SubmoduleStates::SubmoduleStates(
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@ -666,14 +447,13 @@ AudioProcessingImpl::AudioProcessingImpl(
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: data_dumper_(new ApmDataDumper(instance_count_.fetch_add(1) + 1)),
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use_setup_specific_default_aec3_config_(
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UseSetupSpecificDefaultAec3Congfig()),
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gain_controller2_experiment_params_(GetGainController2ExperimentParams()),
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transient_suppressor_vad_mode_(TransientSuppressor::VadMode::kDefault),
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capture_runtime_settings_(RuntimeSettingQueueSize()),
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render_runtime_settings_(RuntimeSettingQueueSize()),
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capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
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render_runtime_settings_enqueuer_(&render_runtime_settings_),
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echo_control_factory_(std::move(echo_control_factory)),
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config_(AdjustConfig(config, gain_controller2_experiment_params_)),
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config_(config),
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submodule_states_(!!capture_post_processor,
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!!render_pre_processor,
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!!capture_analyzer),
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@ -908,52 +688,44 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
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MutexLock lock_render(&mutex_render_);
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MutexLock lock_capture(&mutex_capture_);
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const auto adjusted_config =
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AdjustConfig(config, gain_controller2_experiment_params_);
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RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: "
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<< adjusted_config.ToString();
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RTC_LOG(LS_INFO) << "AudioProcessing::ApplyConfig: " << config.ToString();
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const bool pipeline_config_changed =
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config_.pipeline.multi_channel_render !=
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adjusted_config.pipeline.multi_channel_render ||
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config.pipeline.multi_channel_render ||
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config_.pipeline.multi_channel_capture !=
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adjusted_config.pipeline.multi_channel_capture ||
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config.pipeline.multi_channel_capture ||
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config_.pipeline.maximum_internal_processing_rate !=
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adjusted_config.pipeline.maximum_internal_processing_rate ||
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config.pipeline.maximum_internal_processing_rate ||
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config_.pipeline.capture_downmix_method !=
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adjusted_config.pipeline.capture_downmix_method;
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config.pipeline.capture_downmix_method;
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const bool aec_config_changed =
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config_.echo_canceller.enabled !=
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adjusted_config.echo_canceller.enabled ||
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config_.echo_canceller.mobile_mode !=
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adjusted_config.echo_canceller.mobile_mode;
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config_.echo_canceller.enabled != config.echo_canceller.enabled ||
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config_.echo_canceller.mobile_mode != config.echo_canceller.mobile_mode;
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const bool agc1_config_changed =
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config_.gain_controller1 != adjusted_config.gain_controller1;
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config_.gain_controller1 != config.gain_controller1;
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const bool agc2_config_changed =
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config_.gain_controller2 != adjusted_config.gain_controller2;
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config_.gain_controller2 != config.gain_controller2;
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const bool ns_config_changed =
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config_.noise_suppression.enabled !=
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adjusted_config.noise_suppression.enabled ||
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config_.noise_suppression.level !=
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adjusted_config.noise_suppression.level;
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config_.noise_suppression.enabled != config.noise_suppression.enabled ||
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config_.noise_suppression.level != config.noise_suppression.level;
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const bool ts_config_changed = config_.transient_suppression.enabled !=
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adjusted_config.transient_suppression.enabled;
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config.transient_suppression.enabled;
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const bool pre_amplifier_config_changed =
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config_.pre_amplifier.enabled != adjusted_config.pre_amplifier.enabled ||
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config_.pre_amplifier.enabled != config.pre_amplifier.enabled ||
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config_.pre_amplifier.fixed_gain_factor !=
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adjusted_config.pre_amplifier.fixed_gain_factor;
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config.pre_amplifier.fixed_gain_factor;
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const bool gain_adjustment_config_changed =
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config_.capture_level_adjustment !=
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adjusted_config.capture_level_adjustment;
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config_.capture_level_adjustment != config.capture_level_adjustment;
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config_ = adjusted_config;
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config_ = config;
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if (aec_config_changed) {
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InitializeEchoController();
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@ -2157,7 +1929,7 @@ void AudioProcessingImpl::InitializeTransientSuppressor() {
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const TransientSuppressor::VadMode previous_vad_mode =
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transient_suppressor_vad_mode_;
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transient_suppressor_vad_mode_ = TransientSuppressor::VadMode::kDefault;
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if (UseApmVadSubModule(config_, gain_controller2_experiment_params_)) {
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if (UseApmVadSubModule(config_)) {
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transient_suppressor_vad_mode_ = TransientSuppressor::VadMode::kRnnVad;
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}
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const bool vad_mode_changed =
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@ -2365,20 +2137,13 @@ void AudioProcessingImpl::InitializeGainController2() {
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submodules_.gain_controller2.reset();
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return;
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}
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// Override the input volume controller configuration if the AGC2 experiment
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// is running and its parameters require to fully switch the gain control to
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// Input volume controller configuration if the AGC2 is running
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// and its parameters require to fully switch the gain control to
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// AGC2.
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const bool input_volume_controller_config_overridden =
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gain_controller2_experiment_params_.has_value() &&
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gain_controller2_experiment_params_->agc2_config.has_value();
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const InputVolumeController::Config input_volume_controller_config =
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input_volume_controller_config_overridden
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? gain_controller2_experiment_params_->agc2_config
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->input_volume_controller
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: InputVolumeController::Config{};
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InputVolumeController::Config{};
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// If the APM VAD sub-module is not used, let AGC2 use its internal VAD.
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const bool use_internal_vad =
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!UseApmVadSubModule(config_, gain_controller2_experiment_params_);
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const bool use_internal_vad = !UseApmVadSubModule(config_);
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submodules_.gain_controller2 = std::make_unique<GainController2>(
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config_.gain_controller2, input_volume_controller_config,
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proc_fullband_sample_rate_hz(), num_output_channels(), use_internal_vad);
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@ -2387,7 +2152,7 @@ void AudioProcessingImpl::InitializeGainController2() {
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}
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void AudioProcessingImpl::InitializeVoiceActivityDetector() {
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if (!UseApmVadSubModule(config_, gain_controller2_experiment_params_)) {
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if (!UseApmVadSubModule(config_)) {
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submodules_.voice_activity_detector.reset();
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return;
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}
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@ -163,9 +163,6 @@ class AudioProcessingImpl : public AudioProcessing {
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ReinitializeTransientSuppressor);
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FRIEND_TEST_ALL_PREFIXES(ApmWithSubmodulesExcludedTest,
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BitexactWithDisabledModules);
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FRIEND_TEST_ALL_PREFIXES(
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AudioProcessingImplGainController2FieldTrialParametrizedTest,
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ConfigAdjustedWhenExperimentEnabled);
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void set_stream_analog_level_locked(int level)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_capture_);
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@ -194,46 +191,9 @@ class AudioProcessingImpl : public AudioProcessing {
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static std::atomic<int> instance_count_;
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const bool use_setup_specific_default_aec3_config_;
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// Parameters for the "GainController2" experiment which determines whether
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// the following APM sub-modules are created and, if so, their configurations:
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// AGC2 (`gain_controller2`), AGC1 (`gain_control`, `agc_manager`) and TS
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// (`transient_suppressor`).
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// TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
|
||||
// field trial is removed.
|
||||
struct GainController2ExperimentParams {
|
||||
struct Agc2Config {
|
||||
InputVolumeController::Config input_volume_controller;
|
||||
AudioProcessing::Config::GainController2::AdaptiveDigital
|
||||
adaptive_digital_controller;
|
||||
};
|
||||
// When `agc2_config` is specified, all gain control switches to AGC2 and
|
||||
// the configuration is overridden.
|
||||
absl::optional<Agc2Config> agc2_config;
|
||||
// When true, the transient suppressor submodule is never created regardless
|
||||
// of the APM configuration.
|
||||
bool disallow_transient_suppressor_usage;
|
||||
};
|
||||
// Specified when the "WebRTC-Audio-GainController2" field trial is specified.
|
||||
// TODO(bugs.webrtc.org/7494): Remove when the "WebRTC-Audio-GainController2"
|
||||
// field trial is removed.
|
||||
const absl::optional<GainController2ExperimentParams>
|
||||
gain_controller2_experiment_params_;
|
||||
|
||||
// Parses the "WebRTC-Audio-GainController2" field trial. If disabled, returns
|
||||
// an unspecified value.
|
||||
static absl::optional<GainController2ExperimentParams>
|
||||
GetGainController2ExperimentParams();
|
||||
|
||||
// When `experiment_params` is specified, returns an APM configuration
|
||||
// modified according to the experiment parameters. Otherwise returns
|
||||
// `config`.
|
||||
static AudioProcessing::Config AdjustConfig(
|
||||
const AudioProcessing::Config& config,
|
||||
const absl::optional<GainController2ExperimentParams>& experiment_params);
|
||||
// Returns true if the APM VAD sub-module should be used.
|
||||
static bool UseApmVadSubModule(
|
||||
const AudioProcessing::Config& config,
|
||||
const absl::optional<GainController2ExperimentParams>& experiment_params);
|
||||
// Deprecated.
|
||||
// TODO(bugs.webrtc.org/7494): Remove.
|
||||
static bool UseApmVadSubModule(const AudioProcessing::Config& config);
|
||||
|
||||
TransientSuppressor::VadMode transient_suppressor_vad_mode_;
|
||||
|
||||
|
||||
@ -520,64 +520,6 @@ TEST(AudioProcessingImplTest,
|
||||
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Disabled/");
|
||||
auto apm = AudioProcessingBuilder()
|
||||
.SetConfig({.gain_controller1{.enabled = false}})
|
||||
.Create();
|
||||
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
|
||||
webrtc::AudioProcessing::Config apm_config;
|
||||
apm_config.gain_controller1.enabled = false;
|
||||
apm_config.gain_controller2.enabled = true;
|
||||
apm_config.gain_controller2.adaptive_digital.enabled = true;
|
||||
apm_config.transient_suppression.enabled = true;
|
||||
apm->ApplyConfig(apm_config);
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
float* channel_pointers[] = {buffer.data()};
|
||||
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
|
||||
/*num_channels=*/kNumChannels);
|
||||
Random random_generator(2341U);
|
||||
constexpr int kFramesToProcess = 10;
|
||||
for (int i = 0; i < kFramesToProcess; ++i) {
|
||||
RandomizeSampleVector(&random_generator, buffer);
|
||||
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
||||
channel_pointers),
|
||||
kNoErr);
|
||||
}
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
|
||||
rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
|
||||
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
|
||||
webrtc::AudioProcessing::Config apm_config;
|
||||
apm_config.gain_controller1.enabled = false;
|
||||
apm_config.gain_controller2.enabled = true;
|
||||
apm_config.gain_controller2.adaptive_digital.enabled = true;
|
||||
apm_config.transient_suppression.enabled = true;
|
||||
apm->ApplyConfig(apm_config);
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
float* channel_pointers[] = {buffer.data()};
|
||||
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
|
||||
/*num_channels=*/kNumChannels);
|
||||
Random random_generator(2341U);
|
||||
constexpr int kFramesToProcess = 10;
|
||||
for (int i = 0; i < kFramesToProcess; ++i) {
|
||||
RandomizeSampleVector(&random_generator, buffer);
|
||||
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
||||
channel_pointers),
|
||||
kNoErr);
|
||||
}
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) {
|
||||
// Tests that the echo controller observes an echo path gain change when a
|
||||
// playout volume change is reported.
|
||||
@ -1060,320 +1002,11 @@ TEST(AudioProcessingImplTest,
|
||||
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
Agc2FieldTrialDoNotSwitchToFullAgc2WhenNoAgcIsActive) {
|
||||
constexpr AudioProcessing::Config kOriginal{
|
||||
.gain_controller1{.enabled = false},
|
||||
.gain_controller2{.enabled = false},
|
||||
};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
Agc2FieldTrialDoNotSwitchToFullAgc2WithAgc1Agc2InputVolumeControllers) {
|
||||
constexpr AudioProcessing::Config kOriginal{
|
||||
.gain_controller1{.enabled = true,
|
||||
.analog_gain_controller{.enabled = true}},
|
||||
.gain_controller2{.enabled = true,
|
||||
.input_volume_controller{.enabled = true}},
|
||||
};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
|
||||
}
|
||||
|
||||
class Agc2FieldTrialParametrizedTest
|
||||
class Agc2ParametrizedTest
|
||||
: public ::testing::TestWithParam<AudioProcessing::Config> {};
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfDisabled) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Disabled/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfNoOverride) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"switch_to_agc2:false,"
|
||||
"disallow_transient_suppressor_usage:false/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, DoNotSwitchToFullAgc2) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:false/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
|
||||
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, SwitchToFullAgc2) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest,
|
||||
SwitchToFullAgc2AndOverrideInputVolumeControllerParameters) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
|
||||
"min_input_volume:123,"
|
||||
"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:80,"
|
||||
"speech_probability_threshold:0.9,"
|
||||
"speech_ratio_threshold:1.0/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest,
|
||||
SwitchToFullAgc2AndOverrideAdaptiveDigitalControllerParameters) {
|
||||
const AudioProcessing::Config original = GetParam();
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
|
||||
"headroom_db:10,"
|
||||
"max_gain_db:20,"
|
||||
"initial_gain_db:7,"
|
||||
"max_gain_change_db_per_second:5,"
|
||||
"max_output_noise_level_dbfs:-40/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
|
||||
original.gain_controller2.adaptive_digital);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
|
||||
EXPECT_EQ(
|
||||
adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
|
||||
5);
|
||||
EXPECT_EQ(
|
||||
adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
|
||||
-40);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(original);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(adjusted.gain_controller1.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
|
||||
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
|
||||
ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
|
||||
original.gain_controller2.adaptive_digital);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
|
||||
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
|
||||
EXPECT_EQ(
|
||||
adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
|
||||
5);
|
||||
EXPECT_EQ(
|
||||
adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
|
||||
-40);
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithTs) {
|
||||
AudioProcessing::Config config = GetParam();
|
||||
if (!config.transient_suppression.enabled) {
|
||||
GTEST_SKIP() << "TS is disabled, skip.";
|
||||
}
|
||||
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Disabled/");
|
||||
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
|
||||
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
float* channel_pointers[] = {buffer.data()};
|
||||
StreamConfig stream_config(kSampleRateHz, kNumChannels);
|
||||
Random random_generator(2341U);
|
||||
constexpr int kFramesToProcess = 10;
|
||||
int volume = 100;
|
||||
for (int i = 0; i < kFramesToProcess; ++i) {
|
||||
SCOPED_TRACE(i);
|
||||
RandomizeSampleVector(&random_generator, buffer);
|
||||
apm->set_stream_analog_level(volume);
|
||||
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
||||
channel_pointers),
|
||||
kNoErr);
|
||||
volume = apm->recommended_stream_analog_level();
|
||||
}
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithoutTs) {
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"switch_to_agc2:false,"
|
||||
"disallow_transient_suppressor_usage:true/");
|
||||
TEST_P(Agc2ParametrizedTest, ProcessSucceedsWhenOneAgcEnabled) {
|
||||
auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
|
||||
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
float* channel_pointers[] = {buffer.data()};
|
||||
StreamConfig stream_config(kSampleRateHz, kNumChannels);
|
||||
Random random_generator(2341U);
|
||||
constexpr int kFramesToProcess = 10;
|
||||
int volume = 100;
|
||||
for (int i = 0; i < kFramesToProcess; ++i) {
|
||||
SCOPED_TRACE(i);
|
||||
RandomizeSampleVector(&random_generator, buffer);
|
||||
apm->set_stream_analog_level(volume);
|
||||
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
||||
channel_pointers),
|
||||
kNoErr);
|
||||
volume = apm->recommended_stream_analog_level();
|
||||
}
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest,
|
||||
ProcessSucceedsWhenSwitchToFullAgc2WithTs) {
|
||||
AudioProcessing::Config config = GetParam();
|
||||
if (!config.transient_suppression.enabled) {
|
||||
GTEST_SKIP() << "TS is disabled, skip.";
|
||||
}
|
||||
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"switch_to_agc2:true,"
|
||||
"disallow_transient_suppressor_usage:false/");
|
||||
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
|
||||
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
float* channel_pointers[] = {buffer.data()};
|
||||
StreamConfig stream_config(kSampleRateHz, kNumChannels);
|
||||
Random random_generator(2341U);
|
||||
constexpr int kFramesToProcess = 10;
|
||||
int volume = 100;
|
||||
for (int i = 0; i < kFramesToProcess; ++i) {
|
||||
SCOPED_TRACE(i);
|
||||
RandomizeSampleVector(&random_generator, buffer);
|
||||
apm->set_stream_analog_level(volume);
|
||||
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
|
||||
channel_pointers),
|
||||
kNoErr);
|
||||
volume = apm->recommended_stream_analog_level();
|
||||
}
|
||||
}
|
||||
|
||||
TEST_P(Agc2FieldTrialParametrizedTest,
|
||||
ProcessSucceedsWhenSwitchToFullAgc2WithoutTs) {
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"switch_to_agc2:true,"
|
||||
"disallow_transient_suppressor_usage:true/");
|
||||
auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
|
||||
|
||||
constexpr int kSampleRateHz = 48000;
|
||||
constexpr int kNumChannels = 1;
|
||||
std::array<float, kSampleRateHz / 100> buffer;
|
||||
@ -1395,7 +1028,7 @@ TEST_P(Agc2FieldTrialParametrizedTest,
|
||||
|
||||
INSTANTIATE_TEST_SUITE_P(
|
||||
AudioProcessingImplTest,
|
||||
Agc2FieldTrialParametrizedTest,
|
||||
Agc2ParametrizedTest,
|
||||
::testing::Values(
|
||||
// Full AGC1, TS disabled.
|
||||
AudioProcessing::Config{
|
||||
@ -1430,6 +1063,26 @@ INSTANTIATE_TEST_SUITE_P(
|
||||
.analog_gain_controller = {.enabled = true,
|
||||
.enable_digital_adaptive = false}},
|
||||
.gain_controller2 = {.enabled = true,
|
||||
.adaptive_digital = {.enabled = true}}},
|
||||
// Full AGC2, TS disabled.
|
||||
AudioProcessing::Config{
|
||||
.transient_suppression = {.enabled = false},
|
||||
.gain_controller1 =
|
||||
{.enabled = false,
|
||||
.analog_gain_controller = {.enabled = false,
|
||||
.enable_digital_adaptive = false}},
|
||||
.gain_controller2 = {.enabled = true,
|
||||
.input_volume_controller = {.enabled = true},
|
||||
.adaptive_digital = {.enabled = true}}},
|
||||
// Full AGC2, TS enabled.
|
||||
AudioProcessing::Config{
|
||||
.transient_suppression = {.enabled = true},
|
||||
.gain_controller1 =
|
||||
{.enabled = false,
|
||||
.analog_gain_controller = {.enabled = false,
|
||||
.enable_digital_adaptive = false}},
|
||||
.gain_controller2 = {.enabled = true,
|
||||
.input_volume_controller = {.enabled = true},
|
||||
.adaptive_digital = {.enabled = true}}}));
|
||||
|
||||
TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) {
|
||||
@ -1464,100 +1117,4 @@ TEST(AudioProcessingImplTest, CanEnableTs) {
|
||||
EXPECT_TRUE(adjusted.transient_suppression.enabled);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest, CanDisableTsWithAgc2FieldTrialDisabled) {
|
||||
constexpr AudioProcessing::Config kOriginal = {
|
||||
.transient_suppression = {.enabled = false}};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Disabled/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.transient_suppression.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest, CanEnableTsWithAgc2FieldTrialDisabled) {
|
||||
constexpr AudioProcessing::Config kOriginal = {
|
||||
.transient_suppression = {.enabled = true}};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Disabled/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_TRUE(adjusted.transient_suppression.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_TRUE(adjusted.transient_suppression.enabled);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
CanDisableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
|
||||
constexpr AudioProcessing::Config kOriginal = {
|
||||
.transient_suppression = {.enabled = false}};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"disallow_transient_suppressor_usage:false/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.transient_suppression.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(adjusted.transient_suppression.enabled);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
CanEnableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
|
||||
constexpr AudioProcessing::Config kOriginal = {
|
||||
.transient_suppression = {.enabled = true}};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"disallow_transient_suppressor_usage:false/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_TRUE(adjusted.transient_suppression.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_TRUE(adjusted.transient_suppression.enabled);
|
||||
}
|
||||
|
||||
TEST(AudioProcessingImplTest,
|
||||
CannotEnableTsWithAgc2FieldTrialEnabledAndUsageDisallowed) {
|
||||
constexpr AudioProcessing::Config kOriginal = {
|
||||
.transient_suppression = {.enabled = true}};
|
||||
webrtc::test::ScopedFieldTrials field_trials(
|
||||
"WebRTC-Audio-GainController2/Enabled,"
|
||||
"disallow_transient_suppressor_usage:true/");
|
||||
|
||||
// Test config application via `AudioProcessing` ctor.
|
||||
auto adjusted =
|
||||
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
|
||||
EXPECT_FALSE(adjusted.transient_suppression.enabled);
|
||||
|
||||
// Test config application via `AudioProcessing::ApplyConfig()`.
|
||||
auto apm = AudioProcessingBuilder().Create();
|
||||
apm->ApplyConfig(kOriginal);
|
||||
adjusted = apm->GetConfig();
|
||||
EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user