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/*
* 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 <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/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,
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::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,
absl::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(
absl::optional<DataRate> acked_bitrate,
absl::optional<DataRate> probe_bitrate,
absl::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,
absl::optional<DataRate> acked_bitrate,
DataRate* target_rate);
rtc::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_
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