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

webrtc/voice_engine/channel_proxy.cc

 /*
  *  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 140 matching lines @@
 bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
 }
 
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));
 }
 
+void ChannelProxy::RegisterExternalTransport(Transport* transport) {
+  channel()->RegisterExternalTransport(*transport);
+}
+
+void ChannelProxy::DeRegisterExternalTransport() {
+  channel()->DeRegisterExternalTransport();
+}
+
+bool ChannelProxy::ReceivedRTPPacket(const uint8_t* packet,
+                                     size_t length,
+                                     const PacketTime& packet_time) {
+  return channel()->ReceivedRTPPacket(packet, length, packet_time) == 0;
+}
+
+bool ChannelProxy::ReceivedRTCPPacket(const uint8_t* packet, size_t length) {
+  return channel()->ReceivedRTCPPacket(packet, length) == 0;
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
 
 }  // namespace voe
 }  // namespace webrtc