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 119 matching lines @@
   void DestroyVoiceEngines();
 
   VoiceEngineState voe_send_;
   VoiceEngineState voe_recv_;
   rtc::scoped_refptr<AudioProcessing> apm_send_;
   rtc::scoped_refptr<AudioProcessing> apm_recv_;
 
   // 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.

[nisse-webrtc, 2017/08/21 09:07:07]

Explain briefly *why* it has to be destructed first. I guess it will refer to
some task which in turn will refer to other instance variables?

[eladalon, 2017/08/21 10:56:53]

Actually, come to think of it, putting this last does not provide enough of a
guarantee, because this class is not final; sub-classes might add tasks which
refer to their own member-variables.

+  SingleThreadedTaskQueueForTesting task_queue_;
 };
 
 class BaseTest : public RtpRtcpObserver {
  public:
   BaseTest();
   explicit BaseTest(unsigned int timeout_ms);
   virtual ~BaseTest();
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
   virtual size_t GetNumVideoStreams() const;
   virtual size_t GetNumAudioStreams() const;
   virtual size_t GetNumFlexfecStreams() const;
 
   virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
   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_