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[feat] use original webrtc header defines
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@@ -1,11 +1,15 @@
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/*
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* @Author: DI JUNKUN
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* @Date: 2025-01-14
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* Copyright (c) 2025 by DI JUNKUN, All Rights Reserved.
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef _DELAY_BASED_BWE_H_
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#define _DELAY_BASED_BWE_H_
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#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
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#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
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#include <stdint.h>
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@@ -13,27 +17,29 @@
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#include <optional>
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#include <vector>
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#include "aimd_rate_control.h"
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#include "api/units/data_rate.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "bandwidth_usage.h"
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#include "delay_increase_detector_interface.h"
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#include "inter_arrival.h"
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#include "inter_arrival_delta.h"
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#include "link_capacity_estimator.h"
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#include "network_types.h"
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#include "probe_bitrate_estimator.h"
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enum class BandwidthUsage {
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kBwNormal = 0,
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kBwUnderusing = 1,
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kBwOverusing = 2,
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kLast
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};
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namespace webrtc {
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class RtcEventLog;
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struct BweSeparateAudioPacketsSettings {
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static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";
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BweSeparateAudioPacketsSettings() = default;
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explicit BweSeparateAudioPacketsSettings(
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const FieldTrialsView* key_value_config);
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bool enabled = false;
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int packet_threshold = 10;
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int64_t time_threshold = int64_t::Seconds(1);
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std::unique_ptr<StructParametersParser> Parser();
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TimeDelta time_threshold = TimeDelta::Seconds(1);
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};
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class DelayBasedBwe {
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@@ -43,60 +49,51 @@ class DelayBasedBwe {
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~Result() = default;
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bool updated;
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bool probe;
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int64_t target_bitrate = int64_t::Zero();
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DataRate target_bitrate = DataRate::Zero();
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bool recovered_from_overuse;
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BandwidthUsage delay_detector_state;
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};
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explicit DelayBasedBwe(const FieldTrialsView* key_value_config,
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RtcEventLog* event_log,
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NetworkStatePredictor* network_state_predictor);
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DelayBasedBwe() = delete;
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DelayBasedBwe();
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DelayBasedBwe(const DelayBasedBwe&) = delete;
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DelayBasedBwe& operator=(const DelayBasedBwe&) = delete;
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virtual ~DelayBasedBwe();
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Result IncomingPacketFeedbackVector(
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const TransportPacketsFeedback& msg, std::optional<int64_t> acked_bitrate,
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std::optional<int64_t> probe_bitrate,
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std::optional<NetworkStateEstimate> network_estimate, bool in_alr);
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void OnRttUpdate(int64_t avg_rtt);
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bool LatestEstimate(std::vector<uint32_t>* ssrcs, int64_t* bitrate) const;
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void SetStartBitrate(int64_t start_bitrate);
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void SetMinBitrate(int64_t min_bitrate);
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int64_t GetExpectedBwePeriod() const;
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int64_t TriggerOveruse(int64_t at_time, std::optional<int64_t> link_capacity);
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int64_t last_estimate() const { return prev_bitrate_; }
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Result IncomingPacketFeedbackVector(const TransportPacketsFeedback& msg,
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std::optional<DataRate> acked_bitrate,
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std::optional<DataRate> probe_bitrate,
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bool in_alr);
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void OnRttUpdate(TimeDelta avg_rtt);
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bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
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void SetStartBitrate(DataRate start_bitrate);
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void SetMinBitrate(DataRate min_bitrate);
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TimeDelta GetExpectedBwePeriod() const;
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DataRate TriggerOveruse(Timestamp at_time,
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std::optional<DataRate> link_capacity);
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DataRate last_estimate() const { return prev_bitrate_; }
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BandwidthUsage last_state() const { return prev_state_; }
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private:
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friend class GoogCcStatePrinter;
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void IncomingPacketFeedback(const PacketResult& packet_feedback,
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int64_t at_time);
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Result MaybeUpdateEstimate(std::optional<int64_t> acked_bitrate,
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std::optional<int64_t> probe_bitrate,
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std::optional<NetworkStateEstimate> state_estimate,
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Timestamp at_time);
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Result MaybeUpdateEstimate(std::optional<DataRate> acked_bitrate,
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std::optional<DataRate> probe_bitrate,
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bool recovered_from_overuse, bool in_alr,
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int64_t at_time);
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Timestamp at_time);
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// Updates the current remote rate estimate and returns true if a valid
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// estimate exists.
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bool UpdateEstimate(int64_t at_time, std::optional<int64_t> acked_bitrate,
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int64_t* target_rate);
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rtc::RaceChecker network_race_;
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RtcEventLog* const event_log_;
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const FieldTrialsView* const key_value_config_;
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bool UpdateEstimate(Timestamp at_time, std::optional<DataRate> acked_bitrate,
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DataRate* target_rate);
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// Alternatively, run two separate overuse detectors for audio and video,
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// and fall back to the audio one if we haven't seen a video packet in a
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// while.
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BweSeparateAudioPacketsSettings separate_audio_;
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int64_t audio_packets_since_last_video_;
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int64_t last_video_packet_recv_time_;
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Timestamp last_video_packet_recv_time_;
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NetworkStatePredictor* network_state_predictor_;
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std::unique_ptr<InterArrival> video_inter_arrival_;
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std::unique_ptr<InterArrivalDelta> video_inter_arrival_delta_;
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std::unique_ptr<DelayIncreaseDetectorInterface> video_delay_detector_;
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@@ -105,11 +102,13 @@ class DelayBasedBwe {
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std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_;
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DelayIncreaseDetectorInterface* active_delay_detector_;
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int64_t last_seen_packet_;
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Timestamp last_seen_packet_;
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bool uma_recorded_;
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AimdRateControl rate_control_;
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int64_t prev_bitrate_;
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DataRate prev_bitrate_;
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BandwidthUsage prev_state_;
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};
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#endif
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} // namespace webrtc
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#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
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