Add CreateSend/ReceiveTransport() methods to CallTest.

webrtc/call/rampup_tests.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.
  */
 
@@ skipping 54 matching lines @@
   VideoSendStream* send_stream_;
   test::PacketTransport* send_transport_;
 
  private:
   typedef std::map<uint32_t, uint32_t> SsrcMap;
 
   Call::Config GetSenderCallConfig() override;
   void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams) override;
-  void OnTransportsCreated(test::PacketTransport* send_transport,
-                           test::PacketTransport* receive_transport) override;
+  test::PacketTransport* CreateSendTransport(Call* sender_call) override;
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override;
   void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
 
   static bool BitrateStatsPollingThread(void* obj);
 
   const int start_bitrate_bps_;
   bool start_bitrate_verified_;
@@ skipping 35 matching lines @@
   std::string GetModifierString() const;
   void EvolveTestState(int bitrate_bps, bool suspended);
 
   TestStates test_state_;
   int64_t state_start_ms_;
   int64_t interval_start_ms_;
   int sent_bytes_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_CALL_RAMPUP_TESTS_H_