Define RtpTransportControllerSendInterface.

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 5ee49da91a7a1469285d1fcf675c39c53a43ac5c..fb31342ff49c0a45fcf2cb188f4782d9c28ab75a 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -20,12 +20,11 @@
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
-class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
 class RtcpBandwidthObserver;
 class RtcpRttStats;
-class PacketRouter;
+class RtpTransportControllerSendInterface;
 
 namespace voe {
 class ChannelProxy;
@@ -38,8 +37,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
-                  PacketRouter* packet_router,
-                  CongestionController* congestion_controller,
+                  RtpTransportControllerSendInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats);
@@ -77,7 +75,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
-  CongestionController* const congestion_controller_;
+  RtpTransportControllerSendInterface* const transport_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);