Parse FlexFEC RTP headers in Call and add integration with BWE.

webrtc/modules/congestion_controller/include/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 18 matching lines @@
 
 namespace webrtc {
 
 class BitrateController;
 class Clock;
 class ProbeController;
 class RateLimiter;
 class RemoteBitrateEstimator;
 class RemoteBitrateObserver;
 class RtcEventLog;
+class RtpHeaderExtensionMap;
 class TransportFeedbackObserver;
 
 class CongestionController : public CallStatsObserver, public Module {
  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 {
@@ skipping 44 matching lines @@
   virtual RemoteBitrateEstimator* GetRemoteBitrateEstimator(
       bool send_side_bwe);
   virtual int64_t GetPacerQueuingDelayMs() const;
   virtual PacedSender* pacer() { return pacer_.get(); }
   // TODO(nisse): Deprecated, but still used by downstream projects.
   virtual PacketRouter* packet_router() { return packet_router_; }
   virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
   RateLimiter* GetRetransmissionRateLimiter();
   void EnablePeriodicAlrProbing(bool enable);
 
+  // Are the necessary RTCP and RTP header extensions enabled for send-side BWE?
+  static bool UseSendSideBwe(
+      bool transport_cc,

[danilchap, 2016/12/06 15:04:30]

what does transport_cc mean?

[brandtr, 2016/12/12 13:51:07]

Before this comment, I actually didn't know. Now I know that it stands for
"transport-wide congestion control". The variable name |transport_cc| is used in
the configs of AudioReceiveStream and VideoReceiveStream, so I just reused the
same here.

+      const RtpHeaderExtensionMap& rtp_header_extensions);
+
   // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
   // settings.
   // |min_send_bitrate_bps| is the total minimum send bitrate required by all
   // sending streams.  This is the minimum bitrate the PacedSender will use.
   // Note that CongestionController::OnNetworkChanged can still be called with
   // a lower bitrate estimate.
   // |max_padding_bitrate_bps| is the max bitrate the send streams request for
   // padding. This can be higher than the current network estimate and tells
   // the PacedSender how much it should max pad unless there is real packets to
   // send.
@@ skipping 37 matching lines @@
   uint8_t last_reported_fraction_loss_ GUARDED_BY(critsect_);
   int64_t last_reported_rtt_ GUARDED_BY(critsect_);
   NetworkState network_state_ GUARDED_BY(critsect_);
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionController);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_INCLUDE_CONGESTION_CONTROLLER_H_