Replace scoped_ptr with unique_ptr in webrtc/media/

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 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_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/audio_state.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stream.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/pc/channel.h"
 
@@ skipping 75 matching lines @@
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The primary instance of WebRtc VoiceEngine.
-  rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
+  std::unique_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_ = nullptr;
   std::vector<AudioCodec> codecs_;
   std::vector<WebRtcVoiceMediaChannel*> channels_;
   webrtc::Config voe_config_;
   bool initialized_ = false;
   bool is_dumping_aec_ = false;
 
   webrtc::AgcConfig default_agc_config_;
@@ skipping 54 matching lines @@
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   bool GetStats(VoiceMediaInfo* info) override;
 
   void SetRawAudioSink(
       uint32_t ssrc,
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   // implements Transport interface
   bool SendRtp(const uint8_t* data,
                size_t len,
                const webrtc::PacketOptions& options) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     rtc::PacketOptions rtc_options;
     rtc_options.packet_id = options.packet_id;
     return VoiceMediaChannel::SendPacket(&packet, rtc_options);
@@ skipping 31 matching lines @@
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
-  rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
+  std::unique_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
   bool desired_playout_ = false;
   bool nack_enabled_ = false;
   bool transport_cc_enabled_ = false;
   bool playout_ = false;
   SendFlags desired_send_ = SEND_NOTHING;
   SendFlags send_ = SEND_NOTHING;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
   // Sink for unsignalled stream, which may be set before the stream exists.
-  rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_;
+  std::unique_ptr<webrtc::AudioSinkInterface> default_sink_;
   // Default SSRC to use for RTCP receiver reports in case of no signaled
   // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
   // and https://code.google.com/p/chromium/issues/detail?id=547661
   uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
 
   class WebRtcAudioSendStream;
   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_