Making FakeNetworkPipe demux audio and video packets.

webrtc/test/direct_transport.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_DIRECT_TRANSPORT_H_
@@ skipping 12 matching lines @@
 
 namespace webrtc {
 
 class Clock;
 class PacketReceiver;
 
 namespace test {
 
 class DirectTransport : public Transport {
  public:
-  DirectTransport(Call* send_call, MediaType media_type);
-  DirectTransport(const FakeNetworkPipe::Config& config, Call* send_call,
-                  MediaType media_type);
+  DirectTransport(Call* send_call,
+                  const std::map<uint8_t, MediaType>& payload_type_map);
+  DirectTransport(const FakeNetworkPipe::Config& config,
+                  Call* send_call,
+                  const std::map<uint8_t, MediaType>& payload_type_map);
   // These deprecated variants always use MediaType::VIDEO.
   RTC_DEPRECATED explicit DirectTransport(Call* send_call)
-      : DirectTransport(send_call, MediaType::VIDEO) {}
+      : DirectTransport(send_call, {{}}) {}
 
   RTC_DEPRECATED DirectTransport(const FakeNetworkPipe::Config& config,
                                  Call* send_call)
-      : DirectTransport(config, send_call, MediaType::VIDEO) {}
+      : DirectTransport(config, send_call, {{}}) {}
 
   ~DirectTransport();
 
   void SetConfig(const FakeNetworkPipe::Config& config);
 
   virtual void StopSending();
   // TODO(holmer): Look into moving this to the constructor.
   virtual void SetReceiver(PacketReceiver* receiver);
 
   bool SendRtp(const uint8_t* data,
@@ skipping 14 matching lines @@
   Clock* const clock_;
 
   bool shutting_down_;
 
   FakeNetworkPipe fake_network_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_DIRECT_TRANSPORT_H_