pub struct FieldTrials {Show 26 fields
pub robust_throughput_estimator_settings: RobustThroughputEstimatorSettings,
pub probing_configuration: ProbeControllerConfig,
pub rapid_recovery_experiment: bool,
pub alr_detector_parameters: AlrDetectorConfig,
pub alr_experiment_settings: AlrExperimentSettings,
pub add_pacing_to_congestion_window_pushback: bool,
pub congestion_window: CongestionWindowConfig,
pub video_rate_control: VideoRateControlConfig,
pub vp8_base_heavy_tl3_alloc: bool,
pub separate_audio_packets: BweSeparateAudioPacketsSettings,
pub loss_based_control: LossBasedControlConfig,
pub loss_based_bwe_v2: LossBasedBweV2Config,
pub max_rtt_limit: RttBasedBackoffConfig,
pub trendline_estimator_settings: TrendlineEstimatorSettings,
pub bwe_back_off_factor: BweBackOffFactor,
pub no_bitrate_increase_in_alr: bool,
pub estimate_bounded_increase: EstimateBoundedIncrease,
pub min_alloc_as_lower_bound: bool,
pub ignore_probes_lower_than_network_state_estimate: bool,
pub limit_probes_lower_than_throughput_estimate: bool,
pub pace_at_max_of_bwe_and_lower_link_capacity: bool,
pub limit_pacing_factor_by_upper_link_capacity_estimate: bool,
pub safe_reset_on_route_change: SafeResetOnRouteChange,
pub bwe_throughput_window_config: BitrateEstimatorConfig,
pub loss_experiment: BweLossExperiment,
pub receiver_limit_caps_only: bool,
}
Expand description
Current field trials for WebRTC that impact GoogCC.
These have been painstakingly extracted from the WebRTC source code. Normally these field trials are enabled via bespoke string parsing, but that is very difficult to port correctly. I decided to make everything type safe which doubles as documentation; ur welcome.
Fields§
§robust_throughput_estimator_settings: RobustThroughputEstimatorSettings
WebRTC-Bwe-RobustThroughputEstimatorSettings
probing_configuration: ProbeControllerConfig
WebRTC-Bwe-ProbingConfiguration
rapid_recovery_experiment: bool
WebRTC-BweRapidRecoveryExperiment
alr_detector_parameters: AlrDetectorConfig
WebRTC-AlrDetectorParameters
alr_experiment_settings: AlrExperimentSettings
WebRTC-ProbingScreenshareBwe or WebRTC-StrictPacingAndProbing (legacy?)
add_pacing_to_congestion_window_pushback: bool
WebRTC-AddPacingToCongestionWindowPushback
congestion_window: CongestionWindowConfig
WebRTC-CongestionWindow
video_rate_control: VideoRateControlConfig
WebRTC-VideoRateControl
vp8_base_heavy_tl3_alloc: bool
WebRTC-UseBaseHeavyVP8TL3RateAllocation
separate_audio_packets: BweSeparateAudioPacketsSettings
WebRTC-Bwe-SeparateAudioPackets
loss_based_control: LossBasedControlConfig
WebRTC-Bwe-LossBasedControl
loss_based_bwe_v2: LossBasedBweV2Config
LossBasedBweV2Config
max_rtt_limit: RttBasedBackoffConfig
WebRTC-Bwe-MaxRttLimit
trendline_estimator_settings: TrendlineEstimatorSettings
WebRTC-Bwe-TrendlineEstimatorSettings
bwe_back_off_factor: BweBackOffFactor
WebRTC-BweBackOffFactor
no_bitrate_increase_in_alr: bool
WebRTC-DontIncreaseDelayBasedBweInAlr
estimate_bounded_increase: EstimateBoundedIncrease
WebRTC-Bwe-EstimateBoundedIncrease
min_alloc_as_lower_bound: bool
WebRTC-Bwe-MinAllocAsLowerBound
ignore_probes_lower_than_network_state_estimate: bool
WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate
limit_probes_lower_than_throughput_estimate: bool
WebRTC-Bwe-LimitProbesLowerThanThroughputEstimate
pace_at_max_of_bwe_and_lower_link_capacity: bool
WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity
limit_pacing_factor_by_upper_link_capacity_estimate: bool
WebRTC-Bwe-LimitPacingFactorByUpperLinkCapacityEstimate
safe_reset_on_route_change: SafeResetOnRouteChange
WebRTC-Bwe-SafeResetOnRouteChange
bwe_throughput_window_config: BitrateEstimatorConfig
WebRTC-BweThroughputWindowConfig
loss_experiment: BweLossExperiment
WebRTC-BweLossExperiment
receiver_limit_caps_only: bool
WebRTC-Bwe-ReceiverLimitCapsOnly
Trait Implementations§
Source§impl Clone for FieldTrials
impl Clone for FieldTrials
Source§fn clone(&self) -> FieldTrials
fn clone(&self) -> FieldTrials
1.0.0 · Source§const fn clone_from(&mut self, source: &Self)
const fn clone_from(&mut self, source: &Self)
source
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