/* * 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 #include #include #include #include #include #include "api/transport/bandwidth_usage.h" #include "api/transport/network_types.h" #include "api/units/data_rate.h" #include "api/units/time_delta.h" #include "api/units/timestamp.h" namespace webrtc { enum class LossBasedState { kIncreasing = 0, // TODO(bugs.webrtc.org/12707): Remove one of the increasing states once we // have decided if padding is usefull for ramping up when BWE is loss // limited. kIncreaseUsingPadding = 1, kDecreasing = 2, kDelayBasedEstimate = 3 }; class LinkCapacityTracker { public: LinkCapacityTracker() = default; ~LinkCapacityTracker() = default; // 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(std::optional acknowledged, DataRate target, Timestamp at_time); void OnRttBackoff(DataRate backoff_rate, Timestamp at_time); DataRate estimate() const; private: double capacity_estimate_bps_ = 0; Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity(); DataRate last_delay_based_estimate_ = DataRate::PlusInfinity(); }; class RttBasedBackoff { public: explicit RttBasedBackoff(); ~RttBasedBackoff(); void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt); bool IsRttAboveLimit() const; bool disabled_; TimeDelta configured_limit_; double drop_fraction_; TimeDelta drop_interval_; DataRate bandwidth_floor_; public: TimeDelta rtt_limit_; Timestamp last_propagation_rtt_update_; TimeDelta last_propagation_rtt_; Timestamp last_packet_sent_; private: TimeDelta CorrectedRtt() const; }; class SendSideBandwidthEstimation { public: SendSideBandwidthEstimation(); ~SendSideBandwidthEstimation(); void OnRouteChange(); DataRate target_rate() const; LossBasedState loss_based_state() const; // Return whether the current rtt is higher than the rtt limited configured in // RttBasedBackoff. bool IsRttAboveLimit() 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(std::optional 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(std::optional acknowledged_rate, Timestamp at_time); void UpdateLossBasedEstimator(const TransportPacketsFeedback& report, BandwidthUsage delay_detector_state, std::optional probe_bitrate, bool in_alr); bool PaceAtLossBasedEstimate() const; 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); // 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); RttBasedBackoff rtt_backoff_; LinkCapacityTracker link_capacity_; std::deque > min_bitrate_history_; // incoming filters int lost_packets_since_last_loss_update_; int expected_packets_since_last_loss_update_; std::optional 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 rampup_uma_stats_updated_; float low_loss_threshold_; float high_loss_threshold_; DataRate bitrate_threshold_; bool disable_receiver_limit_caps_only_; LossBasedState loss_based_state_; }; } // namespace webrtc #endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_