Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/test/mock_voe_channel_proxy.h

 /*
  *  Copyright (c) 2015 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 26 matching lines @@
                void(PacketRouter* packet_router));
   MOCK_METHOD0(ResetCongestionControlObjects, void());
   MOCK_CONST_METHOD0(GetRTCPStatistics, CallStatistics());
   MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
   MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
   MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int32_t());
   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
   MOCK_METHOD1(SetSendTelephoneEventPayloadType, bool(int payload_type));
   MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
+  MOCK_METHOD1(RegisterExternalTransport, void(Transport* transport));
+  MOCK_METHOD0(DeRegisterExternalTransport, void());
+  MOCK_METHOD3(ReceivedRTPPacket,bool(const uint8_t* packet,

[the sun, 2016/04/22 12:40:32]

nit: space after ,

[mflodman, 2016/04/27 13:42:17]

Done.

+                                      size_t length,
+                                      const PacketTime& packet_time));
+  MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_MOCK_VOE_CHANNEL_PROXY_H_