Don't hardcode MediaType::ANY in FakeNetworkPipe.

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 154 matching lines @@
 
   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 OnCallsCreated(Call* sender_call, Call* receiver_call);
 
+  // The default implementation creates MediaType::VIDEO transports.
   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 ModifyVideoCaptureStartResolution(int* width,
                                                  int* heigt,
                                                  int* frame_rate);
@@ skipping 33 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_