Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )

Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!

Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}

talk/LICENSE_THIRD_PARTY

@@ -14,6 +14,9 @@ Governed by http://www.fourmilab.ch/md5/ (Public domain):
 base/md5.c
 base/md5.h
 
+Governed by http://www.boost.org/LICENSE_1_0.txt (Boost license):
+base/scoped_ptr.h
+
 Governed by license within files (Public domain):
 Originally downloaded from http://svn.ghostscript.com/jbig2dec/tags/release_0_02/sha1.*
 base/sha1.c

webrtc/api/audiotrack.h

@@ -17,6 +17,7 @@
 #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"
 

webrtc/api/jsepsessiondescription.h

@@ -20,6 +20,7 @@
 #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 {

webrtc/api/localaudiosource.h

@@ -14,6 +14,7 @@
 #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.

webrtc/api/mediastreamprovider.h

@@ -15,6 +15,7 @@
 
 #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"
 

webrtc/api/peerconnection.h

@@ -24,6 +24,7 @@
 #include "webrtc/api/statscollector.h"
 #include "webrtc/api/streamcollection.h"
 #include "webrtc/api/webrtcsession.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 

webrtc/api/peerconnectionfactory.h

@@ -18,6 +18,7 @@
 #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"

webrtc/api/peerconnectioninterface_unittest.cc

@@ -934,20 +934,20 @@ class PeerConnectionInterfaceTest : public testing::Test {
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
-  std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
+  rtc::scoped_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
     CreatePeerConnection();
     AddVoiceStream(kStreamLabel1);
-    std::unique_ptr<SessionDescriptionInterface> offer;
+    rtc::scoped_ptr<SessionDescriptionInterface> offer;
     EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
     return offer;
   }
 
-  std::unique_ptr<SessionDescriptionInterface>
+  rtc::scoped_ptr<SessionDescriptionInterface>
   CreateAnswerWithOneAudioStream() {
-    std::unique_ptr<SessionDescriptionInterface> offer =
+    rtc::scoped_ptr<SessionDescriptionInterface> offer =
         CreateOfferWithOneAudioStream();
     EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
-    std::unique_ptr<SessionDescriptionInterface> answer;
+    rtc::scoped_ptr<SessionDescriptionInterface> answer;
     EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
     return answer;
   }
@@ -973,18 +973,18 @@ class PeerConnectionInterfaceTest : public testing::Test {
 // 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) {
-  std::unique_ptr<SessionDescriptionInterface> offer1 =
+  rtc::scoped_ptr<SessionDescriptionInterface> offer1 =
       CreateOfferWithOneAudioStream();
-  std::unique_ptr<SessionDescriptionInterface> offer2 =
+  rtc::scoped_ptr<SessionDescriptionInterface> offer2 =
       CreateOfferWithOneAudioStream();
   EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
             GetFirstAudioStreamCname(offer2.get()));
 }
 
 TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
-  std::unique_ptr<SessionDescriptionInterface> answer1 =
+  rtc::scoped_ptr<SessionDescriptionInterface> answer1 =
       CreateAnswerWithOneAudioStream();
-  std::unique_ptr<SessionDescriptionInterface> answer2 =
+  rtc::scoped_ptr<SessionDescriptionInterface> answer2 =
       CreateAnswerWithOneAudioStream();
   EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
             GetFirstAudioStreamCname(answer2.get()));

webrtc/api/rtpsender.h

@@ -23,6 +23,7 @@
 #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 {

webrtc/api/statstypes.h

@@ -24,6 +24,7 @@
 #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"

webrtc/api/test/fakeaudiocapturemodule.h

@@ -25,6 +25,7 @@
 #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"

webrtc/api/videocapturertracksource.h

@@ -16,6 +16,7 @@
 #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"

webrtc/audio_send_stream.h

@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/transport.h"

webrtc/base/BUILD.gn

@@ -144,6 +144,7 @@ static_library("rtc_base_approved") {
     "refcount.h",
     "safe_conversions.h",
     "safe_conversions_impl.h",
+    "scoped_ptr.h",
     "scoped_ref_ptr.h",
     "stringencode.cc",
     "stringencode.h",

webrtc/base/base.gyp

@@ -75,6 +75,7 @@
         'refcount.h',
         'safe_conversions.h',
         'safe_conversions_impl.h',
+        'scoped_ptr.h',
         'scoped_ref_ptr.h',
         'stringencode.cc',
         'stringencode.h',

webrtc/base/messagehandler.h

@@ -15,6 +15,7 @@
 #include <utility>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 

webrtc/base/scoped_ptr.h

@@ -0,0 +1,49 @@
+/*
+ *  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.
+ */
+
+// This entire file is deprecated, and will be removed in XXXX 2016. Use
+// std::unique_ptr instead!
+
+#ifndef WEBRTC_BASE_SCOPED_PTR_H__
+#define WEBRTC_BASE_SCOPED_PTR_H__
+
+// All these #includes are left to maximize backwards compatibility.
+
+#include <assert.h>
+#include <stddef.h>
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cstddef>
+#include <memory>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/template_util.h"
+#include "webrtc/typedefs.h"
+
+namespace rtc {
+
+template <typename T, typename Deleter = std::default_delete<T>>
+using scoped_ptr = std::unique_ptr<T, Deleter>;
+
+// These used to convert between std::unique_ptr and std::unique_ptr. Now they
+// are no-ops.
+template <typename T>
+std::unique_ptr<T> ScopedToUnique(std::unique_ptr<T> up) {
+  return up;
+}
+template <typename T>
+std::unique_ptr<T> UniqueToScoped(std::unique_ptr<T> up) {
+  return up;
+}
+
+}  // namespace rtc
+
+#endif  // #ifndef WEBRTC_BASE_SCOPED_PTR_H__

webrtc/common_audio/real_fourier.h

@@ -14,6 +14,7 @@
 #include <complex>
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/aligned_malloc.h"
 
 // Uniform interface class for the real DFT and its inverse, for power-of-2

webrtc/common_video/bitrate_adjuster.cc

@@ -10,7 +10,6 @@
 
 #include "webrtc/common_video/include/bitrate_adjuster.h"
 
-#include <algorithm>
 #include <cmath>
 
 #include "webrtc/base/checks.h"

webrtc/examples/peerconnection/client/conductor.h

@@ -21,6 +21,7 @@
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/examples/peerconnection/client/main_wnd.h"
 #include "webrtc/examples/peerconnection/client/peer_connection_client.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 class VideoCaptureModule;

webrtc/examples/peerconnection/client/peer_connection_client.h

@@ -18,6 +18,7 @@
 
 #include "webrtc/base/nethelpers.h"
 #include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/signalthread.h"
 #include "webrtc/base/sigslot.h"
 

webrtc/libjingle/xmllite/xmlbuilder.h

@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 #include "webrtc/libjingle/xmllite/xmlparser.h"
+#include "webrtc/base/scoped_ptr.h"
 
 #ifdef EXPAT_RELATIVE_PATH
 #include "expat.h"

webrtc/libjingle/xmllite/xmlelement.h

@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 

webrtc/libjingle/xmllite/xmlnsstack.h

@@ -15,6 +15,7 @@
 #include <string>
 #include <vector>
 #include "webrtc/libjingle/xmllite/qname.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 

webrtc/libjingle/xmpp/fakexmppclient.h

@@ -13,7 +13,6 @@
 #ifndef WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
 #define WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
 
-#include <algorithm>
 #include <string>
 #include <vector>
 

webrtc/libjingle/xmpp/hangoutpubsubclient.h

@@ -19,6 +19,7 @@
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/pubsubclient.h"
 #include "webrtc/libjingle/xmpp/pubsubstateclient.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sigslotrepeater.h"
 

webrtc/libjingle/xmpp/pubsubstateclient.h

@@ -22,6 +22,7 @@
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/pubsubclient.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sigslotrepeater.h"
 

webrtc/libjingle/xmpp/xmpplogintask.h

@@ -18,6 +18,7 @@
 #include "webrtc/libjingle/xmpp/jid.h"
 #include "webrtc/libjingle/xmpp/xmppengine.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace buzz {
 

webrtc/libjingle/xmpp/xmpplogintask_unittest.cc

@@ -22,7 +22,6 @@
 #include "webrtc/base/common.h"
 #include "webrtc/base/cryptstring.h"
 #include "webrtc/base/gunit.h"
-#include "webrtc/typedefs.h"
 
 using buzz::Jid;
 using buzz::QName;

webrtc/modules/audio_device/dummy/file_audio_device_factory.cc

@@ -10,7 +10,6 @@
 
 #include "webrtc/modules/audio_device/dummy/file_audio_device_factory.h"
 
-#include <cstdlib>
 #include <cstring>
 
 #include "webrtc/modules/audio_device/dummy/file_audio_device.h"
@@ -27,7 +26,7 @@ FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice(
   if (!_isConfigured) {
     printf("Was compiled with WEBRTC_DUMMY_AUDIO_PLAY_STATIC_FILE "
            "but did not set input/output files to use. Bailing out.\n");
-    std::exit(1);
+    exit(1);
   }
   return new FileAudioDevice(id, _inputAudioFilename, _outputAudioFilename);
 }
@@ -46,7 +45,7 @@ void FileAudioDeviceFactory::SetFilenamesToUse(
   // Sanity: must be compiled with the right define to run this.
   printf("Trying to use dummy file devices, but is not compiled "
          "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
-  std::exit(1);
+  exit(1);
 #endif
 }
 

webrtc/modules/congestion_controller/include/congestion_controller.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/include/module_common_types.h"

webrtc/modules/desktop_capture/cropping_window_capturer.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_options.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 #include "webrtc/modules/desktop_capture/window_capturer.h"

webrtc/modules/desktop_capture/desktop_and_cursor_composer.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capturer.h"
 #include "webrtc/modules/desktop_capture/mouse_cursor_monitor.h"
 

webrtc/modules/desktop_capture/desktop_capturer.h

@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/desktop_capture_types.h"
 #include "webrtc/modules/desktop_capture/shared_memory.h"
 

webrtc/modules/desktop_capture/shared_memory.h

@@ -20,6 +20,7 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h

@@ -18,6 +18,7 @@
 #include <windows.h>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
 #include "webrtc/modules/desktop_capture/shared_desktop_frame.h"

webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h

@@ -18,6 +18,7 @@
 #include <wincodec.h>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
 #include "webrtc/modules/desktop_capture/screen_capturer.h"
 #include "webrtc/modules/desktop_capture/screen_capturer_helper.h"

webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 
 namespace webrtc {

webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

@@ -15,6 +15,7 @@
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 

webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc

@@ -12,8 +12,6 @@
 
 #include <math.h>
 
-#include <cstdlib>
-
 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 
@@ -115,7 +113,7 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
       (header.timestamp - last_received_timestamp_);
 
-  time_diff_samples = std::abs(time_diff_samples);
+  time_diff_samples = abs(time_diff_samples);
 
   // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
   // If this happens, don't update jitter value. Use 5 secs video frequency
@@ -135,7 +133,7 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
      (last_received_timestamp_ +
       last_received_transmission_time_offset_));
 
-  time_diff_samples_ext = std::abs(time_diff_samples_ext);
+  time_diff_samples_ext = abs(time_diff_samples_ext);
 
   if (time_diff_samples_ext < 450000) {
     int32_t jitter_diffQ4TransmissionTimeOffset =

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h

@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPReportBlock
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"

webrtc/modules/rtp_rtcp/source/rtcp_utility.h

@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"

webrtc/modules/rtp_rtcp/test/testAPI/test_api.h

@@ -11,6 +11,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_TEST_TESTAPI_TEST_API_H_
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"

webrtc/modules/utility/include/jvm_android.h

@@ -16,6 +16,7 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/utility/include/helpers_android.h"
 

webrtc/modules/utility/include/process_thread.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/typedefs.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 class Module;

webrtc/modules/video_coding/packet_buffer.h

@@ -20,6 +20,7 @@
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/packet.h"

webrtc/modules/video_coding/test/rtp_player.cc

@@ -12,7 +12,6 @@
 
 #include <stdio.h>
 
-#include <cstdlib>
 #include <map>
 #include <memory>
 
@@ -342,7 +341,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
     assert(packet_source);
     assert(packet_source->get());
     packet_source_.swap(*packet_source);
-    std::srand(321);
+    srand(321);
   }
 
   virtual ~RtpPlayerImpl() {}
@@ -435,8 +434,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
 
       if (no_loss_startup_ > 0) {
         no_loss_startup_--;
-      } else if ((std::rand() + 1.0) / (RAND_MAX + 1.0) <
+      } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < loss_rate_) {  // NOLINT
-                 loss_rate_) {  // NOLINT
         uint16_t seq_num = header.sequenceNumber;
         lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
         DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);

webrtc/modules/video_processing/util/noise_estimation.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_processing/include/video_processing_defines.h"
 #include "webrtc/modules/video_processing/util/denoiser_filter.h"

webrtc/p2p/base/asyncstuntcpsocket.h

@@ -13,6 +13,7 @@
 
 #include "webrtc/base/asynctcpsocket.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketfactory.h"
 
 namespace cricket {

webrtc/p2p/base/dtlstransportchannel_unittest.cc

@@ -42,7 +42,7 @@ static bool IsRtpLeadByte(uint8_t b) {
 cricket::TransportDescription MakeTransportDescription(
     const rtc::scoped_refptr<rtc::RTCCertificate>& cert,
     cricket::ConnectionRole role) {
-  std::unique_ptr<rtc::SSLFingerprint> fingerprint;
+  rtc::scoped_ptr<rtc::SSLFingerprint> fingerprint;
   if (cert) {
     std::string digest_algorithm;
     cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm);

webrtc/p2p/base/stunserver.h

@@ -15,6 +15,7 @@
 
 #include "webrtc/p2p/base/stun.h"
 #include "webrtc/base/asyncudpsocket.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace cricket {
 

webrtc/p2p/base/testrelayserver.h

@@ -15,6 +15,7 @@
 
 #include "webrtc/p2p/base/relayserver.h"
 #include "webrtc/base/asynctcpsocket.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/thread.h"

webrtc/p2p/base/transportdescription.h

@@ -17,6 +17,7 @@
 #include <vector>
 
 #include "webrtc/p2p/base/p2pconstants.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sslfingerprint.h"
 
 namespace cricket {

webrtc/p2p/client/basicportallocator.h

@@ -18,6 +18,7 @@
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/base/messagequeue.h"
 #include "webrtc/base/network.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 
 namespace cricket {

webrtc/p2p/client/fakeportallocator.h

@@ -17,6 +17,7 @@
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/udpport.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace rtc {
 class SocketFactory;

webrtc/p2p/quic/quictransportchannel.h

@@ -19,6 +19,7 @@
 #include "net/quic/quic_packet_writer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/p2p/base/transportchannelimpl.h"
 #include "webrtc/p2p/quic/quicconnectionhelper.h"
 #include "webrtc/p2p/quic/quicsession.h"

webrtc/p2p/stunprober/stunprober.h

@@ -22,6 +22,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/ipaddress.h"
 #include "webrtc/base/network.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"

webrtc/pc/mediasession_unittest.cc

@@ -2404,9 +2404,9 @@ class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
     f1_.set_secure(SEC_ENABLED);
     f2_.set_secure(SEC_ENABLED);
     tdf1_.set_certificate(rtc::RTCCertificate::Create(
-        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
+        rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
     tdf2_.set_certificate(rtc::RTCCertificate::Create(
-        std::unique_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
+        rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
     tdf1_.set_secure(SEC_ENABLED);
     tdf2_.set_secure(SEC_ENABLED);
   }

webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h

@@ -13,6 +13,7 @@
 #include <memory>
 
 #include "webrtc/api/jsep.h"
+#include "webrtc/base/scoped_ptr.h"
 
 NS_ASSUME_NONNULL_BEGIN
 

webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h

@@ -13,6 +13,7 @@
 #include <memory>
 
 #include "webrtc/api/mediaconstraintsinterface.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 

webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h

@@ -11,6 +11,7 @@
 #import "WebRTC/RTCMediaStreamTrack.h"
 
 #include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/base/scoped_ptr.h"
 
 typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
   RTCMediaStreamTrackTypeAudio,

webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h

@@ -13,6 +13,7 @@
 
 #import <AVFoundation/AVFoundation.h>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/video_frame.h"
 

webrtc/system_wrappers/include/clock.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/typedefs.h"
 

webrtc/system_wrappers/include/data_log_impl.h

@@ -24,6 +24,7 @@
 #include <vector>
 
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/system_wrappers/include/utf_util_win.h

@@ -19,6 +19,7 @@
 #include <memory>
 #include <string>
 
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 

webrtc/system_wrappers/source/file_impl.h

@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 namespace webrtc {

webrtc/system_wrappers/source/trace_impl.h

@@ -14,6 +14,7 @@
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/static_instance.h"

webrtc/test/configurable_frame_size_encoder.h

@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/video_encoder.h"
 
 namespace webrtc {

webrtc/test/direct_transport.h

@@ -17,6 +17,7 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/fake_network_pipe.h"
 #include "webrtc/transport.h"
 

webrtc/test/fake_audio_device.h

@@ -15,6 +15,7 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/typedefs.h"

webrtc/test/fake_network_pipe.h

@@ -19,6 +19,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {

webrtc/test/frame_generator_capturer.h

@@ -15,6 +15,7 @@
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_video/rotation.h"
 #include "webrtc/test/video_capturer.h"
 #include "webrtc/typedefs.h"

webrtc/test/test_suite.h

@@ -20,6 +20,7 @@
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 namespace test {