Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/test/direct_transport.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.
  */
 #include "webrtc/test/direct_transport.h"
 
 #include "webrtc/call/call.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 namespace test {
 
-DirectTransport::DirectTransport(Call* send_call, MediaType media_type)
-    : DirectTransport(FakeNetworkPipe::Config(), send_call, media_type) {}
+DirectTransport::DirectTransport(Call* send_call)
+    : DirectTransport(FakeNetworkPipe::Config(), send_call) {}
 
 DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
-                                 Call* send_call, MediaType media_type)
+                                 Call* send_call)
     : send_call_(send_call),
       packet_event_(false, false),
       thread_(NetworkProcess, this, "NetworkProcess"),
       clock_(Clock::GetRealTimeClock()),
       shutting_down_(false),
-      fake_network_(clock_, config, media_type) {
+      fake_network_(clock_, config) {
   thread_.Start();
   if (send_call_) {
     send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
     send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
   }
 }
 
 DirectTransport::~DirectTransport() { StopSending(); }
 
 void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
@@ skipping 45 matching lines @@
   fake_network_.Process();
   int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
   if (wait_time_ms > 0) {
     packet_event_.Wait(static_cast<int>(wait_time_ms));
   }
   rtc::CritScope crit(&lock_);
   return shutting_down_ ? false : true;
 }
 }  // namespace test
 }  // namespace webrtc