Remove webrtc/base/scoped_ptr.h

BUG=webrtc:5520

NOTRY=True

Review-Url: https://codereview.webrtc.org/1942823002
Cr-Commit-Position: refs/heads/master@{#12684}
diff --git a/webrtc/api/audiotrack.h b/webrtc/api/audiotrack.h
index 7fde9b3..096caf9 100644
--- a/webrtc/api/audiotrack.h
+++ b/webrtc/api/audiotrack.h
@@ -17,7 +17,6 @@
 #include "webrtc/api/mediastreamtrack.h"
 #include "webrtc/api/notifier.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_checker.h"
 
diff --git a/webrtc/api/jsepsessiondescription.h b/webrtc/api/jsepsessiondescription.h
index 56dd806..0248a07 100644
--- a/webrtc/api/jsepsessiondescription.h
+++ b/webrtc/api/jsepsessiondescription.h
@@ -20,7 +20,6 @@
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/p2p/base/candidate.h"
 
 namespace cricket {
diff --git a/webrtc/api/localaudiosource.h b/webrtc/api/localaudiosource.h
index e4de650..e1c023e 100644
--- a/webrtc/api/localaudiosource.h
+++ b/webrtc/api/localaudiosource.h
@@ -14,7 +14,6 @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/notifier.h"
 #include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/mediachannel.h"
 
 // LocalAudioSource implements AudioSourceInterface.
diff --git a/webrtc/api/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h
index eef9284..8c866f0 100644
--- a/webrtc/api/mediastreamprovider.h
+++ b/webrtc/api/mediastreamprovider.h
@@ -15,7 +15,6 @@
 
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/media/base/videosourceinterface.h"
 
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index 862c6fb..4842879 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -24,7 +24,6 @@
 #include "webrtc/api/statscollector.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/webrtcsession.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h
index 1992087..233021c 100644
--- a/webrtc/api/peerconnectionfactory.h
+++ b/webrtc/api/peerconnectionfactory.h
@@ -18,7 +18,6 @@
 #include "webrtc/api/mediacontroller.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/pc/channelmanager.h"
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 2594b6c..466c402 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -934,20 +934,20 @@
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
-  rtc::scoped_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
+  std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
     CreatePeerConnection();
     AddVoiceStream(kStreamLabel1);
-    rtc::scoped_ptr<SessionDescriptionInterface> offer;
+    std::unique_ptr<SessionDescriptionInterface> offer;
     EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
     return offer;
   }
 
-  rtc::scoped_ptr<SessionDescriptionInterface>
+  std::unique_ptr<SessionDescriptionInterface>
   CreateAnswerWithOneAudioStream() {
-    rtc::scoped_ptr<SessionDescriptionInterface> offer =
+    std::unique_ptr<SessionDescriptionInterface> offer =
         CreateOfferWithOneAudioStream();
     EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
-    rtc::scoped_ptr<SessionDescriptionInterface> answer;
+    std::unique_ptr<SessionDescriptionInterface> answer;
     EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
     return answer;
   }
@@ -973,18 +973,18 @@
 // The CNAMEs are expected to be generated randomly. It is possible
 // that the test fails, though the possibility is very low.
 TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
-  rtc::scoped_ptr<SessionDescriptionInterface> offer1 =
+  std::unique_ptr<SessionDescriptionInterface> offer1 =
       CreateOfferWithOneAudioStream();
-  rtc::scoped_ptr<SessionDescriptionInterface> offer2 =
+  std::unique_ptr<SessionDescriptionInterface> offer2 =
       CreateOfferWithOneAudioStream();
   EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
             GetFirstAudioStreamCname(offer2.get()));
 }
 
 TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
-  rtc::scoped_ptr<SessionDescriptionInterface> answer1 =
+  std::unique_ptr<SessionDescriptionInterface> answer1 =
       CreateAnswerWithOneAudioStream();
-  rtc::scoped_ptr<SessionDescriptionInterface> answer2 =
+  std::unique_ptr<SessionDescriptionInterface> answer2 =
       CreateAnswerWithOneAudioStream();
   EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
             GetFirstAudioStreamCname(answer2.get()));
diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h
index 86de765..ffe5dae 100644
--- a/webrtc/api/rtpsender.h
+++ b/webrtc/api/rtpsender.h
@@ -23,7 +23,6 @@
 #include "webrtc/api/statscollector.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/audiosource.h"
 
 namespace webrtc {
diff --git a/webrtc/api/statstypes.h b/webrtc/api/statstypes.h
index 9e1c068..4f58b97 100644
--- a/webrtc/api/statstypes.h
+++ b/webrtc/api/statstypes.h
@@ -24,7 +24,6 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/linked_ptr.h"
 #include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/thread_checker.h"
diff --git a/webrtc/api/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h
index ca42c3b..f89249a 100644
--- a/webrtc/api/test/fakeaudiocapturemodule.h
+++ b/webrtc/api/test/fakeaudiocapturemodule.h
@@ -25,7 +25,6 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/messagehandler.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
diff --git a/webrtc/api/videocapturertracksource.h b/webrtc/api/videocapturertracksource.h
index fa4ef07..92f00dc 100644
--- a/webrtc/api/videocapturertracksource.h
+++ b/webrtc/api/videocapturertracksource.h
@@ -16,7 +16,6 @@
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/videotracksource.h"
 #include "webrtc/base/asyncinvoker.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/media/base/videocommon.h"