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

webrtc/test/direct_transport.h

Index: webrtc/test/direct_transport.h
diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h
index eb552c57c8e29b52bcb1f562d6f36bd98d4266dd..20a6857182b71d545b0d742df3aea61aa31138df 100644
--- a/webrtc/test/direct_transport.h
+++ b/webrtc/test/direct_transport.h
@@ -18,11 +18,11 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/call/call.h"
 #include "webrtc/test/fake_network_pipe.h"
 
 namespace webrtc {
 
-class Call;
 class Clock;
 class PacketReceiver;
 
@@ -30,8 +30,17 @@ namespace test {
 
 class DirectTransport : public Transport {
  public:
-  explicit DirectTransport(Call* send_call);
-  DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call);
+  DirectTransport(Call* send_call, MediaType media_type);

[perkj_webrtc, 2017/03/30 09:13:24]

How would you write a call test that test audio and video combined? ie, 
Test joint bwe etc? 

+  DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call,
+                  MediaType media_type);
+  // These deprecated variants always use MediaType::VIDEO.
+  RTC_DEPRECATED explicit DirectTransport(Call* send_call)
+      : DirectTransport(send_call, MediaType::VIDEO) {}
+
+  RTC_DEPRECATED DirectTransport(const FakeNetworkPipe::Config& config,
+                                 Call* send_call)
+      : DirectTransport(config, send_call, MediaType::VIDEO) {}
+
   ~DirectTransport();
 
   void SetConfig(const FakeNetworkPipe::Config& config);