| /* | |
| * Copyright (c) 2016 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. | |
| */ | |
| #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_ | |
| #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_ | |
| #include <stdint.h> | |
| #include <memory> | |
| #include <optional> | |
| #include <vector> | |
| #include "api/field_trials_view.h" | |
| #include "api/network_state_predictor.h" | |
| #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" | |
| #include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h" | |
| #include "modules/congestion_controller/goog_cc/inter_arrival_delta.h" | |
| #include "modules/congestion_controller/goog_cc/link_capacity_estimator.h" | |
| #include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h" | |
| #include "modules/remote_bitrate_estimator/aimd_rate_control.h" | |
| #include "modules/remote_bitrate_estimator/inter_arrival.h" | |
| #include "rtc_base/experiments/struct_parameters_parser.h" | |
| #include "rtc_base/race_checker.h" | |
| namespace webrtc { | |
| class RtcEventLog; | |
| struct BweSeparateAudioPacketsSettings { | |
| static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets"; | |
| BweSeparateAudioPacketsSettings() = default; | |
| explicit BweSeparateAudioPacketsSettings( | |
| const FieldTrialsView* key_value_config); | |
| bool enabled = false; | |
| int packet_threshold = 10; | |
| TimeDelta time_threshold = TimeDelta::Seconds(1); | |
| std::unique_ptr<StructParametersParser> Parser(); | |
| }; | |
| class DelayBasedBwe { | |
| public: | |
| struct Result { | |
| Result(); | |
| ~Result() = default; | |
| bool updated; | |
| bool probe; | |
| DataRate target_bitrate = DataRate::Zero(); | |
| bool recovered_from_overuse; | |
| BandwidthUsage delay_detector_state; | |
| }; | |
| explicit DelayBasedBwe(const FieldTrialsView* key_value_config, | |
| RtcEventLog* event_log, | |
| NetworkStatePredictor* network_state_predictor); | |
| DelayBasedBwe() = delete; | |
| DelayBasedBwe(const DelayBasedBwe&) = delete; | |
| DelayBasedBwe& operator=(const DelayBasedBwe&) = delete; | |
| virtual ~DelayBasedBwe(); | |
| Result IncomingPacketFeedbackVector( | |
| const TransportPacketsFeedback& msg, | |
| std::optional<DataRate> acked_bitrate, | |
| std::optional<DataRate> probe_bitrate, | |
| std::optional<NetworkStateEstimate> network_estimate, | |
| bool in_alr); | |
| void OnRttUpdate(TimeDelta avg_rtt); | |
| bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const; | |
| void SetStartBitrate(DataRate start_bitrate); | |
| void SetMinBitrate(DataRate min_bitrate); | |
| TimeDelta GetExpectedBwePeriod() const; | |
| DataRate TriggerOveruse(Timestamp at_time, | |
| std::optional<DataRate> link_capacity); | |
| DataRate last_estimate() const { return prev_bitrate_; } | |
| BandwidthUsage last_state() const { return prev_state_; } | |
| private: | |
| friend class GoogCcStatePrinter; | |
| void IncomingPacketFeedback(const PacketResult& packet_feedback, | |
| Timestamp at_time); | |
| Result MaybeUpdateEstimate(std::optional<DataRate> acked_bitrate, | |
| std::optional<DataRate> probe_bitrate, | |
| std::optional<NetworkStateEstimate> state_estimate, | |
| bool recovered_from_overuse, | |
| bool in_alr, | |
| Timestamp at_time); | |
| // Updates the current remote rate estimate and returns true if a valid | |
| // estimate exists. | |
| bool UpdateEstimate(Timestamp at_time, | |
| std::optional<DataRate> acked_bitrate, | |
| DataRate* target_rate); | |
| RaceChecker network_race_; | |
| RtcEventLog* const event_log_; | |
| const FieldTrialsView* const key_value_config_; | |
| // Alternatively, run two separate overuse detectors for audio and video, | |
| // and fall back to the audio one if we haven't seen a video packet in a | |
| // while. | |
| BweSeparateAudioPacketsSettings separate_audio_; | |
| int64_t audio_packets_since_last_video_; | |
| Timestamp last_video_packet_recv_time_; | |
| NetworkStatePredictor* network_state_predictor_; | |
| std::unique_ptr<InterArrival> video_inter_arrival_; | |
| std::unique_ptr<InterArrivalDelta> video_inter_arrival_delta_; | |
| std::unique_ptr<DelayIncreaseDetectorInterface> video_delay_detector_; | |
| std::unique_ptr<InterArrival> audio_inter_arrival_; | |
| std::unique_ptr<InterArrivalDelta> audio_inter_arrival_delta_; | |
| std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_; | |
| DelayIncreaseDetectorInterface* active_delay_detector_; | |
| Timestamp last_seen_packet_; | |
| bool uma_recorded_; | |
| AimdRateControl rate_control_; | |
| DataRate prev_bitrate_; | |
| BandwidthUsage prev_state_; | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_ |