Wire up BitrateAllocation to be sent as RTCP TargetBitrate

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index f59676685dbe81e27843953680f3b21ef155ff02..56c05e010cabba214554484697ff28d362e46963 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -150,6 +150,7 @@ class RTCPSender {
   void SetCsrcs(const std::vector<uint32_t>& csrcs);
 
   void SetTargetBitrate(unsigned int target_bitrate);
+  void SetVideoBitrateAllocation(const BitrateAllocation& bitrate);
   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
 
  private:
@@ -197,6 +198,9 @@ class RTCPSender {
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   std::unique_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
+  std::unique_ptr<rtcp::RtcpPacket> BuildTargetBitrate(
+      const RtcpContext& context)
+      EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
 
  private:
   const bool audio_;
@@ -265,6 +269,9 @@ class RTCPSender {
 
   RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
 
+  rtc::Optional<BitrateAllocation> video_bitrate_allocation_

[danilchap, 2016/11/29 10:39:31]

#include "webrtc/base/optional.h"

[sprang_webrtc, 2016/11/29 12:24:01]

Done.

+      GUARDED_BY(critical_section_rtcp_sender_);
+
   void SetFlag(RTCPPacketType type, bool is_volatile)
       EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
   void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)