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

webrtc/voice_engine/channel_proxy.cc

Index: webrtc/voice_engine/channel_proxy.cc
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 10c88212024aa2ea911c09ccb97bcb2fd15ca65e..c4f38ba54d40b8f97aaa0bae2385522cf24ad383 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -158,6 +158,24 @@ void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   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();