Use SingleThreadedTaskQueue in DirectTransport

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
 #define WEBRTC_TEST_CALL_TEST_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/call/call.h"
 #include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #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 {
 
 class VoEBase;
 
 namespace test {
 
 class BaseTest;
 
 class CallTest : public ::testing::Test {
@@ skipping 93 matching lines @@
   std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
   std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
   size_t num_video_streams_;
   size_t num_audio_streams_;
   size_t num_flexfec_streams_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
   test::FakeVideoRenderer fake_renderer_;
 
+  SingleThreadedTaskQueueForTesting task_queue_;
+
  private:
   // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
   // These methods are used to set up legacy voice engines and channels which is
   // necessary while voice engine is being refactored to the new stream API.
   struct VoiceEngineState {
     VoiceEngineState()
         : voice_engine(nullptr),
           base(nullptr),
           channel_id(-1) {}
 
@@ skipping 32 matching lines @@
   virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
   virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
                                          FakeAudioDevice* recv_audio_device);
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnRtpTransportControllerSendCreated(
       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,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void ModifyVideoCaptureStartResolution(int* width,
                                                  int* heigt,
                                                  int* frame_rate);
   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
@@ skipping 31 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_