Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/call/rampup_tests.cc

 /*
  *  Copyright (c) 2013 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 74 matching lines @@
 
 void RampUpTester::OnVideoStreamsCreated(
     VideoSendStream* send_stream,
     const std::vector<VideoReceiveStream*>& receive_streams) {
   send_stream_ = send_stream;
 }
 
 test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
   send_transport_ = new test::PacketTransport(sender_call, this,
                                               test::PacketTransport::kSender,
+                                              MediaType::AUDIO,
                                               forward_transport_config_);
   return send_transport_;
 }
 
 size_t RampUpTester::GetNumVideoStreams() const {
   return num_video_streams_;
 }
 
 size_t RampUpTester::GetNumAudioStreams() const {
   return num_audio_streams_;
@@ skipping 538 matching lines @@
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, AudioTransportSequenceNumber) {
   RampUpTester test(0, 1, 0, 300000, 10000,
                     RtpExtension::kTransportSequenceNumberUri, false, false,
                     false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc