Allow mocking SendSideCongestionController for Call tests.

webrtc/call/call.h

Index: webrtc/call/call.h
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index caf0ee2d881951e5c28320086b6cf74c2c48ef92..f67b6907e5f7a1c3465b984dc23de768f6130631 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -10,6 +10,7 @@
 #ifndef WEBRTC_CALL_CALL_H_
 #define WEBRTC_CALL_CALL_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
@@ -20,6 +21,7 @@
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/flexfec_receive_stream.h"
+#include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
@@ -98,6 +100,11 @@ class Call {
 
   static Call* Create(const Call::Config& config);
 
+  // Allows mocking |transport_send| for testing.
+  static Call* Create(
+      const Call::Config& config,
+      std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
+
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;