Refactored incoming bitrate estimator.

webrtc/modules/congestion_controller/include/send_side_congestion_controller.h

 /*
  *  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.
  */
 
@@ skipping 16 matching lines @@
 #include "webrtc/modules/pacing/packet_router.h"
 
 namespace rtc {
 struct SentPacket;
 }
 
 namespace webrtc {
 
 class BitrateController;
 class Clock;
+class AcknowledgedBitrateEstimator;
 class ProbeController;
 class RateLimiter;
 class RtcEventLog;
 
 class SendSideCongestionController : public CallStatsObserver,
                                      public Module,
                                      public TransportFeedbackObserver {
  public:
   // Observer class for bitrate changes announced due to change in bandwidth
   // estimate or due to that the send pacer is full. Fraction loss and rtt is
   // also part of this callback to allow the observer to optimize its settings
   // for different types of network environments. The bitrate does not include
   // packet headers and is measured in bits per second.
   class Observer {
    public:
     virtual void OnNetworkChanged(uint32_t bitrate_bps,
                                   uint8_t fraction_loss,  // 0 - 255.
                                   int64_t rtt_ms,
-                                  int64_t probing_interval_ms) = 0;
+                                  int64_t probinbweg_interval_ms) = 0;

[terelius, 2017/06/02 12:14:05]

Please fix :)

[tschumi, 2017/06/02 13:13:04]

Done.

 
    protected:
     virtual ~Observer() {}
   };
   // TODO(nisse): Consider deleting the |observer| argument to constructors
   // once CongestionController is deleted.
   SendSideCongestionController(const Clock* clock,
                                Observer* observer,
                                RtcEventLog* event_log,
                                PacketRouter* packet_router);
@@ skipping 68 matching lines @@
   bool IsNetworkDown() const;
   bool HasNetworkParametersToReportChanged(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt);
   const Clock* const clock_;
   rtc::CriticalSection observer_lock_;
   Observer* observer_ GUARDED_BY(observer_lock_);
   RtcEventLog* const event_log_;
   const std::unique_ptr<PacedSender> pacer_;
   const std::unique_ptr<BitrateController> bitrate_controller_;
+  const std::unique_ptr<AcknowledgedBitrateEstimator>
+      acknowledged_bitrate_estimator_;
   const std::unique_ptr<ProbeController> probe_controller_;
   const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
   TransportFeedbackAdapter transport_feedback_adapter_;
   rtc::CriticalSection network_state_lock_;
   uint32_t last_reported_bitrate_bps_ GUARDED_BY(network_state_lock_);
   uint8_t last_reported_fraction_loss_ GUARDED_BY(network_state_lock_);
   int64_t last_reported_rtt_ GUARDED_BY(network_state_lock_);
   NetworkState network_state_ GUARDED_BY(network_state_lock_);
   rtc::CriticalSection bwe_lock_;
   int min_bitrate_bps_ GUARDED_BY(bwe_lock_);
   std::unique_ptr<DelayBasedBwe> delay_based_bwe_ GUARDED_BY(bwe_lock_);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendSideCongestionController);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_H_