Replace scoped_ptr with unique_ptr in webrtc/api/

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 2012 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 510 matching lines @@
     SignalDataChannelDestroyed();
     channel_manager_->DestroyDataChannel(data_channel_.release());
   }
   SignalDestroyed();
 
   LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
 }
 
 bool WebRtcSession::Initialize(
     const PeerConnectionFactoryInterface::Options& options,
-    rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+    std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
   rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
   transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
 
   // Obtain a certificate from RTCConfiguration if any were provided (optional).
   rtc::scoped_refptr<rtc::RTCCertificate> certificate;
   if (!rtc_configuration.certificates.empty()) {
     // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
     // just picking the first one. The decision should be made based on the DTLS
@@ skipping 126 matching lines @@
     CreateSessionDescriptionObserver* observer,
     const cricket::MediaSessionOptions& session_options) {
   webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
 }
 
 bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
                                         std::string* err_desc) {
   ASSERT(signaling_thread()->IsCurrent());
 
   // Takes the ownership of |desc| regardless of the result.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+  std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
 
   // Validate SDP.
   if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
     return false;
   }
 
   // Update the initial_offerer flag if this session is the initial_offerer.
   Action action = GetAction(desc->type());
   if (state() == STATE_INIT && action == kOffer) {
     initial_offerer_ = true;
@@ skipping 35 matching lines @@
     return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);
   }
   return true;
 }
 
 bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
                                          std::string* err_desc) {
   ASSERT(signaling_thread()->IsCurrent());
 
   // Takes the ownership of |desc| regardless of the result.
-  rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+  std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
 
   // Validate SDP.
   if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
     return false;
   }
 
-  rtc::scoped_ptr<SessionDescriptionInterface> old_remote_desc(
+  std::unique_ptr<SessionDescriptionInterface> old_remote_desc(
       remote_desc_.release());
   remote_desc_.reset(desc_temp.release());
 
   // Transport and Media channels will be created only when offer is set.
   Action action = GetAction(desc->type());
   if (action == kOffer && !CreateChannels(desc->description())) {
     // TODO(mallinath) - Handle CreateChannel failure, as new local description
     // is applied. Restore back to old description.
     return BadRemoteSdp(desc->type(), kCreateChannelFailed, err_desc);
   }
@@ skipping 477 matching lines @@
     LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
     return;
   }
 
   if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
     ASSERT(false);
   }
 }
 
 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
-                                    rtc::scoped_ptr<AudioSinkInterface> sink) {
+                                    std::unique_ptr<AudioSinkInterface> sink) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_)
     return;
 
   voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
 }
 
 RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
   ASSERT(signaling_thread()->IsCurrent());
   if (voice_channel_) {
@@ skipping 905 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc