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

webrtc/test/mock_voe_channel_proxy.h

Index: webrtc/test/mock_voe_channel_proxy.h
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index c2211f85547a11d9c1555b1d50e66294968b1117..a27a739006edfaffb9aff16f23abf87872cb571d 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -44,6 +44,12 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   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,
+                                       size_t length,
+                                       const PacketTime& packet_time));
+  MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
 };
 }  // namespace test
 }  // namespace webrtc