Making FakeNetworkPipe demux audio and video packets.

webrtc/test/rtp_rtcp_observer.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_RTP_RTCP_OBSERVER_H_
@@ skipping 76 matching lines @@
   const int timeout_ms_;
 };
 
 class PacketTransport : public test::DirectTransport {
  public:
   enum TransportType { kReceiver, kSender };
 
   PacketTransport(Call* send_call,
                   RtpRtcpObserver* observer,
                   TransportType transport_type,
-                  MediaType media_type,
+                  const std::map<uint8_t, MediaType>& payload_type_map,
                   const FakeNetworkPipe::Config& configuration)
-      : test::DirectTransport(configuration, send_call, media_type),
+      : test::DirectTransport(configuration, send_call, payload_type_map),
         observer_(observer),
         transport_type_(transport_type) {}
 
  private:
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
     EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
     RtpRtcpObserver::Action action;
     {
@@ skipping 33 matching lines @@
     return true;  // Will never happen, makes compiler happy.
   }
 
   RtpRtcpObserver* const observer_;
   TransportType transport_type_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_