| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| * |
| * FEC and NACK added bitrate is handled outside class |
| */ |
| |
| #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
| #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |
| |
| #include <stdint.h> |
| |
| #include <deque> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/transport/network_types.h" |
| #include "api/transport/webrtc_key_value_config.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h" |
| #include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| |
| namespace webrtc { |
| |
| class RtcEventLog; |
| |
| class LinkCapacityTracker { |
| public: |
| LinkCapacityTracker(); |
| ~LinkCapacityTracker(); |
| // Call when a new delay-based estimate is available. |
| void UpdateDelayBasedEstimate(Timestamp at_time, |
| DataRate delay_based_bitrate); |
| void OnStartingRate(DataRate start_rate); |
| void OnRateUpdate(absl::optional<DataRate> acknowledged, |
| DataRate target, |
| Timestamp at_time); |
| void OnRttBackoff(DataRate backoff_rate, Timestamp at_time); |
| DataRate estimate() const; |
| |
| private: |
| FieldTrialParameter<TimeDelta> tracking_rate; |
| double capacity_estimate_bps_ = 0; |
| Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity(); |
| DataRate last_delay_based_estimate_ = DataRate::PlusInfinity(); |
| }; |
| |
| class RttBasedBackoff { |
| public: |
| explicit RttBasedBackoff(const WebRtcKeyValueConfig* key_value_config); |
| ~RttBasedBackoff(); |
| void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt); |
| TimeDelta CorrectedRtt(Timestamp at_time) const; |
| |
| FieldTrialFlag disabled_; |
| FieldTrialParameter<TimeDelta> configured_limit_; |
| FieldTrialParameter<double> drop_fraction_; |
| FieldTrialParameter<TimeDelta> drop_interval_; |
| FieldTrialParameter<DataRate> bandwidth_floor_; |
| |
| public: |
| TimeDelta rtt_limit_; |
| Timestamp last_propagation_rtt_update_; |
| TimeDelta last_propagation_rtt_; |
| Timestamp last_packet_sent_; |
| }; |
| |
| class SendSideBandwidthEstimation { |
| public: |
| SendSideBandwidthEstimation() = delete; |
| SendSideBandwidthEstimation(const WebRtcKeyValueConfig* key_value_config, |
| RtcEventLog* event_log); |
| ~SendSideBandwidthEstimation(); |
| |
| void OnRouteChange(); |
| |
| DataRate target_rate() const; |
| uint8_t fraction_loss() const { return last_fraction_loss_; } |
| TimeDelta round_trip_time() const { return last_round_trip_time_; } |
| |
| DataRate GetEstimatedLinkCapacity() const; |
| // Call periodically to update estimate. |
| void UpdateEstimate(Timestamp at_time); |
| void OnSentPacket(const SentPacket& sent_packet); |
| void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt); |
| |
| // Call when we receive a RTCP message with TMMBR or REMB. |
| void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth); |
| |
| // Call when a new delay-based estimate is available. |
| void UpdateDelayBasedEstimate(Timestamp at_time, DataRate bitrate); |
| |
| // Call when we receive a RTCP message with a ReceiveBlock. |
| void UpdatePacketsLost(int64_t packets_lost, |
| int64_t number_of_packets, |
| Timestamp at_time); |
| |
| // Call when we receive a RTCP message with a ReceiveBlock. |
| void UpdateRtt(TimeDelta rtt, Timestamp at_time); |
| |
| void SetBitrates(absl::optional<DataRate> send_bitrate, |
| DataRate min_bitrate, |
| DataRate max_bitrate, |
| Timestamp at_time); |
| void SetSendBitrate(DataRate bitrate, Timestamp at_time); |
| void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate); |
| int GetMinBitrate() const; |
| void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate, |
| Timestamp at_time); |
| void IncomingPacketFeedbackVector(const TransportPacketsFeedback& report); |
| |
| private: |
| friend class GoogCcStatePrinter; |
| |
| enum UmaState { kNoUpdate, kFirstDone, kDone }; |
| |
| bool IsInStartPhase(Timestamp at_time) const; |
| |
| void UpdateUmaStatsPacketsLost(Timestamp at_time, int packets_lost); |
| |
| // Updates history of min bitrates. |
| // After this method returns min_bitrate_history_.front().second contains the |
| // min bitrate used during last kBweIncreaseIntervalMs. |
| void UpdateMinHistory(Timestamp at_time); |
| |
| // Gets the upper limit for the target bitrate. This is the minimum of the |
| // delay based limit, the receiver limit and the loss based controller limit. |
| DataRate GetUpperLimit() const; |
| // Prints a warning if `bitrate` if sufficiently long time has past since last |
| // warning. |
| void MaybeLogLowBitrateWarning(DataRate bitrate, Timestamp at_time); |
| // Stores an update to the event log if the loss rate has changed, the target |
| // has changed, or sufficient time has passed since last stored event. |
| void MaybeLogLossBasedEvent(Timestamp at_time); |
| |
| // Cap `bitrate` to [min_bitrate_configured_, max_bitrate_configured_] and |
| // set `current_bitrate_` to the capped value and updates the event log. |
| void UpdateTargetBitrate(DataRate bitrate, Timestamp at_time); |
| // Applies lower and upper bounds to the current target rate. |
| // TODO(srte): This seems to be called even when limits haven't changed, that |
| // should be cleaned up. |
| void ApplyTargetLimits(Timestamp at_time); |
| |
| bool LossBasedBandwidthEstimatorV1Enabled() const; |
| bool LossBasedBandwidthEstimatorV2Enabled() const; |
| |
| bool LossBasedBandwidthEstimatorV1ReadyForUse() const; |
| bool LossBasedBandwidthEstimatorV2ReadyForUse() const; |
| |
| RttBasedBackoff rtt_backoff_; |
| LinkCapacityTracker link_capacity_; |
| |
| std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_; |
| |
| // incoming filters |
| int lost_packets_since_last_loss_update_; |
| int expected_packets_since_last_loss_update_; |
| |
| absl::optional<DataRate> acknowledged_rate_; |
| DataRate current_target_; |
| DataRate last_logged_target_; |
| DataRate min_bitrate_configured_; |
| DataRate max_bitrate_configured_; |
| Timestamp last_low_bitrate_log_; |
| |
| bool has_decreased_since_last_fraction_loss_; |
| Timestamp last_loss_feedback_; |
| Timestamp last_loss_packet_report_; |
| uint8_t last_fraction_loss_; |
| uint8_t last_logged_fraction_loss_; |
| TimeDelta last_round_trip_time_; |
| |
| // The max bitrate as set by the receiver in the call. This is typically |
| // signalled using the REMB RTCP message and is used when we don't have any |
| // send side delay based estimate. |
| DataRate receiver_limit_; |
| DataRate delay_based_limit_; |
| Timestamp time_last_decrease_; |
| Timestamp first_report_time_; |
| int initially_lost_packets_; |
| DataRate bitrate_at_2_seconds_; |
| UmaState uma_update_state_; |
| UmaState uma_rtt_state_; |
| std::vector<bool> rampup_uma_stats_updated_; |
| RtcEventLog* const event_log_; |
| Timestamp last_rtc_event_log_; |
| float low_loss_threshold_; |
| float high_loss_threshold_; |
| DataRate bitrate_threshold_; |
| LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_; |
| LossBasedBweV2 loss_based_bandwidth_estimator_v2_; |
| FieldTrialFlag disable_receiver_limit_caps_only_; |
| }; |
| } // namespace webrtc |
| #endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ |