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 05ed3aaeb8cfb7697e0ea8d54a934ea5fb2d2b45..956f058539ee3035309076e14ab20fd4545a2afa 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -20,11 +20,10 @@
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
-class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
 class RtcpRttStats;
-class PacketRouter;
+class RtpTransportControllerSenderInterface;
 
 namespace voe {
 class ChannelProxy;
@@ -37,8 +36,7 @@ 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,
+                  RtpTransportControllerSenderInterface* transport,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats);
@@ -76,7 +74,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
-  CongestionController* const congestion_controller_;
+  RtpTransportControllerSenderInterface* const transport_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };