Use SingleThreadedTaskQueue in DirectTransport

webrtc/test/call_test.h

Index: webrtc/test/call_test.h
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 5186afa7533881e625692f20f52962c30403a358..cf53e32f2012c1a60ace5d92bf6d26bf15da16ba 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -23,6 +23,7 @@
 #include "webrtc/test/fake_videorenderer.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
+#include "webrtc/test/single_threaded_task_queue.h"
 
 namespace webrtc {
 
@@ -162,6 +163,10 @@ class CallTest : public ::testing::Test {
   // The audio devices must outlive the voice engines.
   std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
   std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
+
+ protected:
+  // Should be last, to destruct first.
+  SingleThreadedTaskQueueForTesting task_queue_;
 };
 
 class BaseTest : public RtpRtcpObserver {
@@ -188,8 +193,11 @@ class BaseTest : public RtpRtcpObserver {
       RtpTransportControllerSend* controller);
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
 
-  virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
-  virtual test::PacketTransport* CreateReceiveTransport();
+  virtual test::PacketTransport* CreateSendTransport(
+      SingleThreadedTaskQueueForTesting* task_queue,
+      Call* sender_call);
+  virtual test::PacketTransport* CreateReceiveTransport(
+      SingleThreadedTaskQueueForTesting* task_queue);
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,