Replace scoped_ptr with unique_ptr in webrtc/api/

webrtc/api/peerconnection_unittest.cc

Index: webrtc/api/peerconnection_unittest.cc
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
index 156605c9b9b9679fde7f3042a6067a7806763a96..521486f98b8e8cd79d76148f661eefdfeca46143 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/api/peerconnection_unittest.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 #include <list>
 #include <map>
+#include <memory>
 #include <utility>
 #include <vector>
 
@@ -31,7 +32,6 @@
 #include "webrtc/api/test/mockpeerconnectionobservers.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/ssladapter.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/thread.h"
@@ -154,7 +154,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       const std::string& id,
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       bool prefer_constraint_apis,
       rtc::Thread* worker_thread) {
     PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
@@ -171,7 +171,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
       rtc::Thread* worker_thread) {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
 
@@ -184,7 +184,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       const std::string& id,
       const PeerConnectionFactory::Options* options,
       rtc::Thread* worker_thread) {
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
 
@@ -199,7 +199,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   void Negotiate() { Negotiate(true, true); }
 
   void Negotiate(bool audio, bool video) {
-    rtc::scoped_ptr<SessionDescriptionInterface> offer;
+    std::unique_ptr<SessionDescriptionInterface> offer;
     ASSERT_TRUE(DoCreateOffer(&offer));
 
     if (offer->description()->GetContentByName("audio")) {
@@ -231,7 +231,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
                          int sdp_mline_index,
                          const std::string& msg) override {
     LOG(INFO) << id_ << "ReceiveIceMessage";
-    rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
+    std::unique_ptr<webrtc::IceCandidateInterface> candidate(
         webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
     EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
   }
@@ -549,7 +549,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
 
   // Verify the CreateDtmfSender interface
   void VerifyDtmf() {
-    rtc::scoped_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
+    std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
     rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
 
     // We can't create a DTMF sender with an invalid audio track or a non local
@@ -804,7 +804,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   bool Init(
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       bool prefer_constraint_apis,
       rtc::Thread* worker_thread) {
     EXPECT_TRUE(!peer_connection_);
@@ -814,7 +814,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     }
     prefer_constraint_apis_ = prefer_constraint_apis;
 
-    rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
+    std::unique_ptr<cricket::PortAllocator> port_allocator(
         new cricket::FakePortAllocator(worker_thread, nullptr));
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
 
@@ -838,9 +838,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   }
 
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
-      rtc::scoped_ptr<cricket::PortAllocator> port_allocator,
+      std::unique_ptr<cricket::PortAllocator> port_allocator,
       const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
+      std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
     // CreatePeerConnection with RTCConfiguration.
     webrtc::PeerConnectionInterface::RTCConfiguration config;
     webrtc::PeerConnectionInterface::IceServer ice_server;
@@ -858,10 +858,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
       // If we are not sending any streams ourselves it is time to add some.
       AddMediaStream(true, true);
     }
-    rtc::scoped_ptr<SessionDescriptionInterface> desc(
+    std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("offer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
-    rtc::scoped_ptr<SessionDescriptionInterface> answer;
+    std::unique_ptr<SessionDescriptionInterface> answer;
     EXPECT_TRUE(DoCreateAnswer(&answer));
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
@@ -874,12 +874,12 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
 
   void HandleIncomingAnswer(const std::string& msg) {
     LOG(INFO) << id_ << "HandleIncomingAnswer";
-    rtc::scoped_ptr<SessionDescriptionInterface> desc(
+    std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("answer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
   }
 
-  bool DoCreateOfferAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc,
+  bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                            bool offer) {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
         observer(new rtc::RefCountedObject<
@@ -905,11 +905,11 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     return observer->result();
   }
 
-  bool DoCreateOffer(rtc::scoped_ptr<SessionDescriptionInterface>* desc) {
+  bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) {
     return DoCreateOfferAnswer(desc, true);
   }
 
-  bool DoCreateAnswer(rtc::scoped_ptr<SessionDescriptionInterface>* desc) {
+  bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) {
     return DoCreateOfferAnswer(desc, false);
   }
 
@@ -982,10 +982,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   // Needed to keep track of number of frames sent.
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   // Needed to keep track of number of frames received.
-  std::map<std::string, rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       fake_video_renderers_;
   // Needed to ensure frames aren't received for removed tracks.
-  std::vector<rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
       removed_fake_video_renderers_;
   // Needed to keep track of number of frames received when external decoder
   // used.
@@ -1002,7 +1002,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   std::vector<cricket::FakeVideoCapturer*> video_capturers_;
   webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0;
   // |local_video_renderer_| attached to the first created local video track.
-  rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
+  std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
 
   webrtc::FakeConstraints offer_answer_constraints_;
   PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
@@ -1016,7 +1016,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
   bool remove_cvo_ = false;
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
-  rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
+  std::unique_ptr<MockDataChannelObserver> data_observer_;
 };
 
 class P2PTestConductor : public testing::Test {
@@ -1253,7 +1253,7 @@ class P2PTestConductor : public testing::Test {
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
 
-    rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+    std::unique_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
                                               : nullptr);
     dtls_identity_store->use_alternate_key();
@@ -1305,11 +1305,11 @@ class P2PTestConductor : public testing::Test {
   // |worker_thread_| is used by both |initiating_client_| and
   // |receiving_client_|. Must be destroyed last.
   rtc::Thread worker_thread_;
-  rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
-  rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
+  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
+  std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
-  rtc::scoped_ptr<PeerConnectionTestClient> initiating_client_;
-  rtc::scoped_ptr<PeerConnectionTestClient> receiving_client_;
+  std::unique_ptr<PeerConnectionTestClient> initiating_client_;
+  std::unique_ptr<PeerConnectionTestClient> receiving_client_;
   bool prefer_constraint_apis_ = true;
 };
 
@@ -1405,7 +1405,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
-  rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+  std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_initializing_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();
 
@@ -1443,7 +1443,7 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
 
   // Keeping the original peer around which will still send packets to the
   // receiving client. These SRTP packets will be dropped.
-  rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+  std::unique_ptr<PeerConnectionTestClient> original_peer(
       set_receiving_client(CreateDtlsClientWithAlternateKey()));
   original_peer->pc()->Close();