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

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index b670efee3cbfad5074cbbc77050ed7e590f5230f..0cf5c6d9a2ddcc124b9132ccaf241b5cd2c343ac 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -17,6 +17,7 @@
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
+class CongestionController;
 class VoiceEngine;
 
 namespace voe {
@@ -27,7 +28,8 @@ namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
-                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  CongestionController* congestion_controller);
   ~AudioSendStream() override;
 
   // webrtc::SendStream implementation.