Add CreateSend/ReceiveTransport() methods to CallTest.

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_
@@ skipping 40 matching lines @@
   static const uint32_t kVideoSendSsrcs[kNumSsrcs];
   static const uint32_t kAudioSendSsrc;
   static const uint32_t kReceiverLocalVideoSsrc;
   static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
 
  protected:
   // RunBaseTest overwrites the audio_state and the voice_engine of the send and
   // receive Call configs to simplify test code and avoid having old VoiceEngine
   // APIs in the tests.
-  void RunBaseTest(BaseTest* test, const FakeNetworkPipe::Config& config);
+  void RunBaseTest(BaseTest* test);
 
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         Transport* send_transport);
@@ skipping 72 matching lines @@
 
   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;
 
   virtual size_t GetNumVideoStreams() const;
   virtual size_t GetNumAudioStreams() const;
 
   virtual Call::Config GetSenderCallConfig();
   virtual Call::Config GetReceiverCallConfig();
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
-  virtual void OnTransportsCreated(PacketTransport* send_transport,
-                                   PacketTransport* receive_transport);
+
+  virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
+  virtual test::PacketTransport* CreateReceiveTransport();
 
   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);
 
   virtual void ModifyAudioConfigs(
@@ skipping 18 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_