/*
 *  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 <stddef.h>
#include <stdint.h>

#include <memory>
#include <vector>

#include "absl/types/optional.h"
#include "api/network_state_predictor.h"
#include "api/transport/network_types.h"
#include "api/transport/webrtc_key_value_config.h"
#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.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 BweIgnoreSmallPacketsSettings {
  static constexpr char kKey[] = "WebRTC-BweIgnoreSmallPacketsFix";

  BweIgnoreSmallPacketsSettings() = default;
  explicit BweIgnoreSmallPacketsSettings(
      const WebRtcKeyValueConfig* key_value_config);

  double smoothing_factor = 0.1;
  double fraction_large = 1.0;
  DataSize large_threshold = DataSize::Zero();
  DataSize small_threshold = DataSize::Zero();

  std::unique_ptr<StructParametersParser> Parser();
};

struct BweSeparateAudioPacketsSettings {
  static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";

  BweSeparateAudioPacketsSettings() = default;
  explicit BweSeparateAudioPacketsSettings(
      const WebRtcKeyValueConfig* 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(bool probe, DataRate target_bitrate);
    ~Result() = default;
    bool updated;
    bool probe;
    DataRate target_bitrate = DataRate::Zero();
    bool recovered_from_overuse;
    bool backoff_in_alr;
  };

  explicit DelayBasedBwe(const WebRtcKeyValueConfig* 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;
  void SetAlrLimitedBackoffExperiment(bool enabled);
  DataRate TriggerOveruse(Timestamp at_time,
                          absl::optional<DataRate> link_capacity);
  DataRate last_estimate() const { return prev_bitrate_; }

 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 now,
                      absl::optional<DataRate> acked_bitrate,
                      DataRate* target_bitrate);

  rtc::RaceChecker network_race_;
  RtcEventLog* const event_log_;
  const WebRtcKeyValueConfig* const key_value_config_;

  // Filtering out small packets. Intention is to base the detection only
  // on video packets even if we have TWCC sequence numbers for audio.
  BweIgnoreSmallPacketsSettings ignore_small_;
  double fraction_large_packets_;

  // 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<DelayIncreaseDetectorInterface> video_delay_detector_;
  std::unique_ptr<InterArrival> audio_inter_arrival_;
  std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_;
  DelayIncreaseDetectorInterface* active_delay_detector_;

  Timestamp last_seen_packet_;
  bool uma_recorded_;
  AimdRateControl rate_control_;
  DataRate prev_bitrate_;
  bool has_once_detected_overuse_;
  BandwidthUsage prev_state_;
  bool alr_limited_backoff_enabled_;
};

}  // namespace webrtc

#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_