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/*
* 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_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
#define MODULES_BITRATE_CONTROLLER_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/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/bitrate_controller/loss_based_bandwidth_estimation.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/experiments/field_trial_parser.h"
namespace webrtc {
class RtcEventLog;
class LinkCapacityTracker {
public:
LinkCapacityTracker();
~LinkCapacityTracker();
void OnOveruse(DataRate acknowledged_rate, Timestamp at_time);
void OnStartingRate(DataRate start_rate);
void OnRateUpdate(DataRate acknowledged, 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();
};
class RttBasedBackoff {
public:
RttBasedBackoff();
~RttBasedBackoff();
void OnRouteChange();
void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
TimeDelta CorrectedRtt(Timestamp at_time) const;
FieldTrialParameter<TimeDelta> rtt_limit_;
FieldTrialParameter<double> drop_fraction_;
FieldTrialParameter<TimeDelta> drop_interval_;
FieldTrialFlag persist_on_route_change_;
FieldTrialParameter<bool> safe_timeout_;
public:
Timestamp last_propagation_rtt_update_;
TimeDelta last_propagation_rtt_;
Timestamp last_packet_sent_;
};
class SendSideBandwidthEstimation {
public:
SendSideBandwidthEstimation() = delete;
explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
~SendSideBandwidthEstimation();
void OnRouteChange();
void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
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 UpdateReceiverBlock(uint8_t fraction_loss,
TimeDelta rtt_ms,
int number_of_packets,
Timestamp at_time);
// Call when we receive a RTCP message with a ReceiveBlock.
void UpdatePacketsLost(int packets_lost,
int 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:
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);
DataRate MaybeRampupOrBackoff(DataRate new_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 CapBitrateToThresholds(Timestamp at_time, DataRate bitrate);
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_bitrate_;
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_;
Timestamp last_timeout_;
uint8_t last_fraction_loss_;
uint8_t last_logged_fraction_loss_;
TimeDelta last_round_trip_time_;
DataRate bwe_incoming_;
DataRate delay_based_bitrate_;
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* event_log_;
Timestamp last_rtc_event_log_;
bool in_timeout_experiment_;
float low_loss_threshold_;
float high_loss_threshold_;
DataRate bitrate_threshold_;
LossBasedBandwidthEstimation loss_based_bandwidth_estimation_;
};
} // namespace webrtc
#endif // MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_