Unify Transport and newapi::Transport interfaces.

webrtc/audio_send_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 11 matching lines @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioSendStream : public SendStream {
  public:
   struct Stats {};
 
   struct Config {
     Config() = delete;
-    explicit Config(newapi::Transport* send_transport)
+    explicit Config(Transport* send_transport)
         : send_transport(send_transport) {}
 
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
     struct Rtp {
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
 
       // RTP header extensions used for the received stream.
       std::vector<RtpExtension> extensions;
     } rtp;
 
     // Transport for outgoing packets.
-    newapi::Transport* send_transport = nullptr;
+    Transport* send_transport = nullptr;
 
     rtc::scoped_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_