Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 216 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
new file mode 100644
index 0000000000..3a4efc47c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
@@ -0,0 +1,216 @@
+/*
+ *  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/field_trials_view.h"
+#include "api/network_state_predictor.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/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 FieldTrialsView* key_value_config);
+  ~RttBasedBackoff();
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+  bool IsRttAboveLimit() 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_;
+
+ private:
+  TimeDelta CorrectedRtt() const;
+};
+
+class SendSideBandwidthEstimation {
+ public:
+  SendSideBandwidthEstimation() = delete;
+  SendSideBandwidthEstimation(const FieldTrialsView* key_value_config,
+                              RtcEventLog* event_log);
+  ~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(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 UpdateLossBasedEstimator(const TransportPacketsFeedback& report,
+                                BandwidthUsage delay_detector_state,
+                                absl::optional<DataRate> probe_bitrate,
+                                bool in_alr);
+
+ 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_;
+  LossBasedState loss_based_state_;
+  FieldTrialFlag disable_receiver_limit_caps_only_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_