Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  Copyright (c) 2017 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 62 matching lines @@
 }
 
 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
   return FakeNetworkPipe::Config();
 }
 
 test::PacketTransport* AudioQualityTest::CreateSendTransport(
     Call* sender_call) {
   return new test::PacketTransport(
       sender_call, this, test::PacketTransport::kSender,
+      MediaType::AUDIO,
       GetNetworkPipeConfig());
 }
 
 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
   return new test::PacketTransport(nullptr, this,
-      test::PacketTransport::kReceiver, GetNetworkPipeConfig());
+      test::PacketTransport::kReceiver, MediaType::AUDIO,
+      GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
   send_config->send_codec_spec.codec_inst = kDefaultCodec;
 }
 
 void AudioQualityTest::PerformTest() {
   // Wait until the input audio file is done...
@@ skipping 43 matching lines @@
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
 }  // namespace test
 }  // namespace webrtc