Replace scoped_ptr with unique_ptr in webrtc/pc/

webrtc/pc/channel.h

 /*
  *  Copyright 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_PC_CHANNEL_H_
 #define WEBRTC_PC_CHANNEL_H_
 
 #include <map>
+#include <memory>
 #include <set>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/sigslot.h"
@@ skipping 272 matching lines @@
   const std::string content_name_;
   std::string transport_name_;
   bool rtcp_transport_enabled_;
   TransportChannel* transport_channel_;
   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
   TransportChannel* rtcp_transport_channel_;
   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
   SrtpFilter srtp_filter_;
   RtcpMuxFilter rtcp_mux_filter_;
   BundleFilter bundle_filter_;
-  rtc::scoped_ptr<ConnectionMonitor> connection_monitor_;
+  std::unique_ptr<ConnectionMonitor> connection_monitor_;
   bool enabled_;
   bool writable_;
   bool rtp_ready_to_send_;
   bool rtcp_ready_to_send_;
   bool was_ever_writable_;
   MediaContentDirection local_content_direction_;
   MediaContentDirection remote_content_direction_;
   bool has_received_packet_;
   bool dtls_keyed_;
   bool secure_required_;
@@ skipping 34 matching lines @@
   // Returns if the telephone-event has been negotiated.
   bool CanInsertDtmf();
   // Send and/or play a DTMF |event| according to the |flags|.
   // The DTMF out-of-band signal will be used on sending.
   // The |ssrc| should be either 0 or a valid send stream ssrc.
   // The valid value for the |event| are 0 which corresponding to DTMF
   // event 0-9, *, #, A-D.
   bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
   bool SetOutputVolume(uint32_t ssrc, double volume);
   void SetRawAudioSink(uint32_t ssrc,
-                       rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
+                       std::unique_ptr<webrtc::AudioSinkInterface> sink);
 
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
   // Monitoring functions
   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
@@ skipping 31 matching lines @@
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
 
   static const int kEarlyMediaTimeout = 1000;
   MediaEngineInterface* media_engine_;
   bool received_media_;
-  rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_;
-  rtc::scoped_ptr<AudioMonitor> audio_monitor_;
+  std::unique_ptr<VoiceMediaMonitor> media_monitor_;
+  std::unique_ptr<AudioMonitor> audio_monitor_;
 
   // Last AudioSendParameters sent down to the media_channel() via
   // SetSendParameters.
   AudioSendParameters last_send_params_;
   // Last AudioRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   AudioRecvParameters last_recv_params_;
 };
 
 // VideoChannel is a specialization for video.
@@ skipping 38 matching lines @@
                                   std::string* error_desc);
   bool GetStats_w(VideoMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VideoMediaChannel* media_channel, const VideoMediaInfo& info);
 
-  rtc::scoped_ptr<VideoMediaMonitor> media_monitor_;
+  std::unique_ptr<VideoMediaMonitor> media_monitor_;
 
   // Last VideoSendParameters sent down to the media_channel() via
   // SetSendParameters.
   VideoSendParameters last_send_params_;
   // Last VideoRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   VideoRecvParameters last_recv_params_;
 };
 
 // DataChannel is a specialization for data.
@@ skipping 95 matching lines @@
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       DataMediaChannel* media_channel, const DataMediaInfo& info);
   virtual bool ShouldSetupDtlsSrtp() const;
   void OnDataReceived(
       const ReceiveDataParams& params, const char* data, size_t len);
   void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
   void OnDataChannelReadyToSend(bool writable);
   void OnStreamClosedRemotely(uint32_t sid);
 
-  rtc::scoped_ptr<DataMediaMonitor> media_monitor_;
+  std::unique_ptr<DataMediaMonitor> media_monitor_;
   // TODO(pthatcher): Make a separate SctpDataChannel and
   // RtpDataChannel instead of using this.
   DataChannelType data_channel_type_;
   bool ready_to_send_data_;
 
   // Last DataSendParameters sent down to the media_channel() via
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNEL_H_