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

webrtc/voice_engine/channel_proxy.h

Index: webrtc/voice_engine/channel_proxy.h
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index f8f0659af8742766f247c6265e61d7a177391b78..df0c3f22ef67cd0c5bf2d2ab34f92cd47c8a7751 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -25,6 +25,7 @@ namespace webrtc {
 class AudioSinkInterface;
 class PacketRouter;
 class RtpPacketSender;
+class Transport;
 class TransportFeedbackObserver;
 
 namespace voe {
@@ -72,6 +73,13 @@ class ChannelProxy {
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
+  virtual void RegisterExternalTransport(Transport* transport);
+  virtual void DeRegisterExternalTransport();
+  virtual bool ReceivedRTPPacket(const uint8_t* packet,

[tommi, 2016/04/23 16:00:09]

prefer using CopyOnWriteBuffer here instead of forcing buffer copies due to the
interface.

[mflodman, 2016/04/27 13:42:18]

I think that as something we should look at, but not in this CL. The contract
has been to copy the incoming data to store it in a buffer used by the decoder
anyway and I'm not how much we really would gain by doing this change. But we
can take a look at it.

+                                 size_t length,
+                                 const PacketTime& packet_time);
+  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
+
  private:
   Channel* channel() const;