blob: 8c1e1d04d8723d7094bca19f9a69cc86f97dba4e [file] [log] [blame]
* 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
#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 "rtc_base/experiments/field_trial_parser.h"
namespace webrtc {
class RtcEventLog;
class 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;
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 {
explicit RttBasedBackoff(const WebRtcKeyValueConfig* key_value_config);
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_;
TimeDelta rtt_limit_;
Timestamp last_propagation_rtt_update_;
TimeDelta last_propagation_rtt_;
Timestamp last_packet_sent_;
class SendSideBandwidthEstimation {
SendSideBandwidthEstimation() = delete;
SendSideBandwidthEstimation(const WebRtcKeyValueConfig* key_value_config,
RtcEventLog* event_log);
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);
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);
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_estimation_;
FieldTrialFlag disable_receiver_limit_caps_only_;
} // namespace webrtc