Prepare the AudioSendStream to be hooked up to send-side BWE.

webrtc/voice_engine/channel_proxy.h

Index: webrtc/voice_engine/channel_proxy.h
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 6b916a54c0eecc73473c831b37fe017a2ee7217e..4e5ae021260ab95fc0571877ecd7c9779c21ef5f 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -19,6 +19,11 @@
 #include <vector>
 
 namespace webrtc {
+
+class RtpPacketSender;
+class TransportFeedbackObserver;
+class PacketRouter;
+
 namespace voe {
 
 class Channel;
@@ -43,6 +48,11 @@ class ChannelProxy {
   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
   virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
   virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+  virtual void SetSendTransportSequenceNumber(int id);
+  virtual void SetCongestionControlObjects(
+      RtpPacketSender* rtp_packet_sender,
+      TransportFeedbackObserver* transport_feedback_observer,
+      PacketRouter* packet_router);
 
   virtual CallStatistics GetRTCPStatistics() const;
   virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;