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

webrtc/voice_engine/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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/voice_engine/channel_manager.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 namespace webrtc {
 
 class AudioSinkInterface;
 class PacketRouter;
 class RtpPacketSender;
+class Transport;
 class TransportFeedbackObserver;
 
 namespace voe {
 
 class Channel;
 
 // This class provides the "view" of a voe::Channel that we need to implement
 // webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
 // purposes:
 //  1. Allow mocking just the interfaces used, instead of the entire
@@ skipping 27 matching lines @@
   virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
   virtual NetworkStatistics GetNetworkStatistics() const;
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int32_t GetSpeechOutputLevelFullRange() const;
   virtual uint32_t GetDelayEstimate() const;
 
   virtual bool SetSendTelephoneEventPayloadType(int payload_type);
   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;
 
   rtc::ThreadChecker thread_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_