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

webrtc/audio_receive_stream.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.
  */
 
@@ skipping 65 matching lines @@
       // Enable feedback for send side bandwidth estimation.
       // See
       // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
       // for details.
       bool transport_cc = false;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
-    Transport* receive_transport = nullptr;
     Transport* rtcp_send_transport = nullptr;
 
     // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
     // level components.
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     // of Call.
     int voe_channel_id = -1;
 
     // Identifier for an A/V synchronization group. Empty string to disable.
     // TODO(pbos): Synchronize streams in a sync group, not just one video
@@ skipping 15 matching lines @@
   // Only one sink can be set and passing a null sink clears an existing one.
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_