Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/test/call_test.cc

 /*
  *  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 440 matching lines @@
 
 Call::Config BaseTest::GetReceiverCallConfig() {
   return Call::Config(&event_log_);
 }
 
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
 test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
   return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
+                             MediaType::VIDEO,
                              FakeNetworkPipe::Config());
 }
 
 test::PacketTransport* BaseTest::CreateReceiveTransport() {
   return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
+                             MediaType::VIDEO,
                              FakeNetworkPipe::Config());
 }
 
 size_t BaseTest::GetNumVideoStreams() const {
   return 1;
 }
 
 size_t BaseTest::GetNumAudioStreams() const {
   return 0;
 }
@@ skipping 47 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc