Wire up BitrateAllocation to be sent as RTCP TargetBitrate

webrtc/video/payload_router.h

 /*
  *  Copyright (c) 2015 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 22 matching lines @@
  public:
   // Rtp modules are assumed to be sorted in simulcast index order.
   PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
                 int payload_type);
   ~PayloadRouter();
 
   static size_t DefaultMaxPayloadLength();
 
   // PayloadRouter will only route packets if being active, all packets will be
   // dropped otherwise.
-  void set_active(bool active);
-  bool active();
+  void SetActive(bool active);
+  bool IsActive();
 
   // Implements EncodedImageCallback.
   // Returns 0 if the packet was routed / sent, -1 otherwise.
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) override;
 
   // Returns the maximum allowed data payload length, given the configured MTU
   // and RTP headers.
   size_t MaxPayloadLength() const;
 
+  void OnBitrateAllocationUpdated(const BitrateAllocation& bitrate);
+
  private:
   void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   rtc::CriticalSection crit_;
   bool active_ GUARDED_BY(crit_);
 
   // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
   const std::vector<RtpRtcp*> rtp_modules_;
   const int payload_type_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_