Replace scoped_ptr with unique_ptr everywhere

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.
  */
 
 #ifndef WEBRTC_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ skipping 53 matching lines @@
 
     // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
     // components.
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     // of Call.
     int voe_channel_id = -1;
 
     // Ownership of the encoder object is transferred to Call when the config is
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
-    // rtc::scoped_ptr<AudioEncoder> encoder;
+    // std::unique_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.
   };
 
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_SEND_STREAM_H_