From 6ab3db249b075e0e820a263d54804f521e7bc24b Mon Sep 17 00:00:00 2001 From: kwiberg Date: Wed, 11 May 2016 05:07:26 -0700 Subject: [PATCH] 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 | 3 ++ webrtc/api/audiotrack.h | 1 + webrtc/api/jsepsessiondescription.h | 1 + webrtc/api/localaudiosource.h | 1 + webrtc/api/mediastreamprovider.h | 1 + webrtc/api/peerconnection.h | 1 + webrtc/api/peerconnectionfactory.h | 1 + .../api/peerconnectioninterface_unittest.cc | 18 +++---- webrtc/api/rtpsender.h | 1 + webrtc/api/statstypes.h | 1 + webrtc/api/test/fakeaudiocapturemodule.h | 1 + webrtc/api/videocapturertracksource.h | 1 + webrtc/audio_send_stream.h | 1 + webrtc/base/BUILD.gn | 1 + webrtc/base/base.gyp | 1 + webrtc/base/messagehandler.h | 1 + webrtc/base/scoped_ptr.h | 49 +++++++++++++++++++ webrtc/common_audio/real_fourier.h | 1 + webrtc/common_video/bitrate_adjuster.cc | 1 - .../peerconnection/client/conductor.h | 1 + .../client/peer_connection_client.h | 1 + webrtc/libjingle/xmllite/xmlbuilder.h | 1 + webrtc/libjingle/xmllite/xmlelement.h | 1 + webrtc/libjingle/xmllite/xmlnsstack.h | 1 + webrtc/libjingle/xmpp/fakexmppclient.h | 1 - webrtc/libjingle/xmpp/hangoutpubsubclient.h | 1 + webrtc/libjingle/xmpp/pubsubstateclient.h | 1 + webrtc/libjingle/xmpp/xmpplogintask.h | 1 + .../libjingle/xmpp/xmpplogintask_unittest.cc | 1 - .../dummy/file_audio_device_factory.cc | 5 +- .../include/congestion_controller.h | 1 + .../cropping_window_capturer.h | 1 + .../desktop_and_cursor_composer.h | 1 + .../desktop_capture/desktop_capturer.h | 1 + .../modules/desktop_capture/shared_memory.h | 1 + .../win/screen_capturer_win_gdi.h | 1 + .../win/screen_capturer_win_magnifier.h | 1 + .../include/remote_ntp_time_estimator.h | 1 + .../rtp_rtcp/include/rtp_payload_registry.h | 1 + .../source/receive_statistics_impl.cc | 6 +-- .../rtp_rtcp/source/rtcp_receiver_help.h | 1 + webrtc/modules/rtp_rtcp/source/rtcp_utility.h | 1 + .../rtp_rtcp/source/rtp_receiver_impl.h | 1 + .../modules/rtp_rtcp/test/testAPI/test_api.h | 1 + webrtc/modules/utility/include/jvm_android.h | 1 + .../modules/utility/include/process_thread.h | 1 + webrtc/modules/video_coding/packet_buffer.h | 1 + .../modules/video_coding/test/rtp_player.cc | 6 +-- .../video_processing/util/noise_estimation.h | 1 + webrtc/p2p/base/asyncstuntcpsocket.h | 1 + .../p2p/base/dtlstransportchannel_unittest.cc | 2 +- webrtc/p2p/base/stunserver.h | 1 + webrtc/p2p/base/testrelayserver.h | 1 + webrtc/p2p/base/transportdescription.h | 1 + webrtc/p2p/client/basicportallocator.h | 1 + webrtc/p2p/client/fakeportallocator.h | 1 + webrtc/p2p/quic/quictransportchannel.h | 1 + webrtc/p2p/stunprober/stunprober.h | 1 + webrtc/pc/mediasession_unittest.cc | 4 +- .../Classes/RTCIceCandidate+Private.h | 1 + .../Classes/RTCMediaConstraints+Private.h | 1 + .../Classes/RTCMediaStreamTrack+Private.h | 1 + .../Classes/avfoundationvideocapturer.h | 1 + webrtc/system_wrappers/include/clock.h | 1 + .../system_wrappers/include/data_log_impl.h | 1 + webrtc/system_wrappers/include/utf_util_win.h | 1 + webrtc/system_wrappers/source/file_impl.h | 1 + webrtc/system_wrappers/source/trace_impl.h | 1 + webrtc/test/configurable_frame_size_encoder.h | 1 + webrtc/test/direct_transport.h | 1 + webrtc/test/fake_audio_device.h | 1 + webrtc/test/fake_network_pipe.h | 1 + webrtc/test/frame_generator_capturer.h | 1 + webrtc/test/test_suite.h | 1 + 74 files changed, 133 insertions(+), 26 deletions(-) create mode 100644 webrtc/base/scoped_ptr.h diff --git a/talk/LICENSE_THIRD_PARTY b/talk/LICENSE_THIRD_PARTY index 5a2760664e..50068efecf 100644 --- a/talk/LICENSE_THIRD_PARTY +++ b/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 diff --git a/webrtc/api/audiotrack.h b/webrtc/api/audiotrack.h index 096caf9d0e..7fde9b3e22 100644 --- a/webrtc/api/audiotrack.h +++ b/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" diff --git a/webrtc/api/jsepsessiondescription.h b/webrtc/api/jsepsessiondescription.h index 0248a07c72..56dd806110 100644 --- a/webrtc/api/jsepsessiondescription.h +++ b/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 { diff --git a/webrtc/api/localaudiosource.h b/webrtc/api/localaudiosource.h index e1c023e542..e4de650537 100644 --- a/webrtc/api/localaudiosource.h +++ b/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. diff --git a/webrtc/api/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h index 8c866f0d69..eef92846cb 100644 --- a/webrtc/api/mediastreamprovider.h +++ b/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" diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h index 4842879662..862c6fb630 100644 --- a/webrtc/api/peerconnection.h +++ b/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 { diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h index 233021c71f..1992087f75 100644 --- a/webrtc/api/peerconnectionfactory.h +++ b/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" diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc index 466c40236e..2594b6c106 100644 --- a/webrtc/api/peerconnectioninterface_unittest.cc +++ b/webrtc/api/peerconnectioninterface_unittest.cc @@ -934,20 +934,20 @@ class PeerConnectionInterfaceTest : public testing::Test { ASSERT_TRUE(stream->AddTrack(video_track)); } - std::unique_ptr CreateOfferWithOneAudioStream() { + rtc::scoped_ptr CreateOfferWithOneAudioStream() { CreatePeerConnection(); AddVoiceStream(kStreamLabel1); - std::unique_ptr offer; + rtc::scoped_ptr offer; EXPECT_TRUE(DoCreateOffer(&offer, nullptr)); return offer; } - std::unique_ptr + rtc::scoped_ptr CreateAnswerWithOneAudioStream() { - std::unique_ptr offer = + rtc::scoped_ptr offer = CreateOfferWithOneAudioStream(); EXPECT_TRUE(DoSetRemoteDescription(offer.release())); - std::unique_ptr answer; + rtc::scoped_ptr 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 offer1 = + rtc::scoped_ptr offer1 = CreateOfferWithOneAudioStream(); - std::unique_ptr offer2 = + rtc::scoped_ptr offer2 = CreateOfferWithOneAudioStream(); EXPECT_NE(GetFirstAudioStreamCname(offer1.get()), GetFirstAudioStreamCname(offer2.get())); } TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) { - std::unique_ptr answer1 = + rtc::scoped_ptr answer1 = CreateAnswerWithOneAudioStream(); - std::unique_ptr answer2 = + rtc::scoped_ptr answer2 = CreateAnswerWithOneAudioStream(); EXPECT_NE(GetFirstAudioStreamCname(answer1.get()), GetFirstAudioStreamCname(answer2.get())); diff --git a/webrtc/api/rtpsender.h b/webrtc/api/rtpsender.h index ffe5daeb01..86de7658c9 100644 --- a/webrtc/api/rtpsender.h +++ b/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 { diff --git a/webrtc/api/statstypes.h b/webrtc/api/statstypes.h index 4f58b97540..9e1c068217 100644 --- a/webrtc/api/statstypes.h +++ b/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" diff --git a/webrtc/api/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h index f89249ad2f..ca42c3b018 100644 --- a/webrtc/api/test/fakeaudiocapturemodule.h +++ b/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" diff --git a/webrtc/api/videocapturertracksource.h b/webrtc/api/videocapturertracksource.h index 92f00dc4e5..fa4ef0709b 100644 --- a/webrtc/api/videocapturertracksource.h +++ b/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" diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h index d8e98bb0ec..d5d16f2711 100644 --- a/webrtc/audio_send_stream.h +++ b/webrtc/audio_send_stream.h @@ -15,6 +15,7 @@ #include #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/config.h" #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" #include "webrtc/transport.h" diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn index 09950a8f22..63c2da533e 100644 --- a/webrtc/base/BUILD.gn +++ b/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", diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp index 5444915758..74b029c1f7 100644 --- a/webrtc/base/base.gyp +++ b/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', diff --git a/webrtc/base/messagehandler.h b/webrtc/base/messagehandler.h index 6a3c2ef740..2d964df79c 100644 --- a/webrtc/base/messagehandler.h +++ b/webrtc/base/messagehandler.h @@ -15,6 +15,7 @@ #include #include "webrtc/base/constructormagic.h" +#include "webrtc/base/scoped_ptr.h" namespace rtc { diff --git a/webrtc/base/scoped_ptr.h b/webrtc/base/scoped_ptr.h new file mode 100644 index 0000000000..af515f6801 --- /dev/null +++ b/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 +#include +#include + +#include +#include +#include + +#include "webrtc/base/constructormagic.h" +#include "webrtc/base/template_util.h" +#include "webrtc/typedefs.h" + +namespace rtc { + +template > +using scoped_ptr = std::unique_ptr; + +// These used to convert between std::unique_ptr and std::unique_ptr. Now they +// are no-ops. +template +std::unique_ptr ScopedToUnique(std::unique_ptr up) { + return up; +} +template +std::unique_ptr UniqueToScoped(std::unique_ptr up) { + return up; +} + +} // namespace rtc + +#endif // #ifndef WEBRTC_BASE_SCOPED_PTR_H__ diff --git a/webrtc/common_audio/real_fourier.h b/webrtc/common_audio/real_fourier.h index 5e83e37f70..8dbb98f776 100644 --- a/webrtc/common_audio/real_fourier.h +++ b/webrtc/common_audio/real_fourier.h @@ -14,6 +14,7 @@ #include #include +#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 diff --git a/webrtc/common_video/bitrate_adjuster.cc b/webrtc/common_video/bitrate_adjuster.cc index ada6c5db4b..9c61120afe 100644 --- a/webrtc/common_video/bitrate_adjuster.cc +++ b/webrtc/common_video/bitrate_adjuster.cc @@ -10,7 +10,6 @@ #include "webrtc/common_video/include/bitrate_adjuster.h" -#include #include #include "webrtc/base/checks.h" diff --git a/webrtc/examples/peerconnection/client/conductor.h b/webrtc/examples/peerconnection/client/conductor.h index 02351b78be..db2f77b646 100644 --- a/webrtc/examples/peerconnection/client/conductor.h +++ b/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; diff --git a/webrtc/examples/peerconnection/client/peer_connection_client.h b/webrtc/examples/peerconnection/client/peer_connection_client.h index dbf2d8ff3d..bc8b8cc9c9 100644 --- a/webrtc/examples/peerconnection/client/peer_connection_client.h +++ b/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" diff --git a/webrtc/libjingle/xmllite/xmlbuilder.h b/webrtc/libjingle/xmllite/xmlbuilder.h index 3b48f28247..08bb5ec166 100644 --- a/webrtc/libjingle/xmllite/xmlbuilder.h +++ b/webrtc/libjingle/xmllite/xmlbuilder.h @@ -15,6 +15,7 @@ #include #include #include "webrtc/libjingle/xmllite/xmlparser.h" +#include "webrtc/base/scoped_ptr.h" #ifdef EXPAT_RELATIVE_PATH #include "expat.h" diff --git a/webrtc/libjingle/xmllite/xmlelement.h b/webrtc/libjingle/xmllite/xmlelement.h index 37adfc0170..70c6f79923 100644 --- a/webrtc/libjingle/xmllite/xmlelement.h +++ b/webrtc/libjingle/xmllite/xmlelement.h @@ -15,6 +15,7 @@ #include #include "webrtc/libjingle/xmllite/qname.h" +#include "webrtc/base/scoped_ptr.h" namespace buzz { diff --git a/webrtc/libjingle/xmllite/xmlnsstack.h b/webrtc/libjingle/xmllite/xmlnsstack.h index 64e2f6ee66..07a9883de5 100644 --- a/webrtc/libjingle/xmllite/xmlnsstack.h +++ b/webrtc/libjingle/xmllite/xmlnsstack.h @@ -15,6 +15,7 @@ #include #include #include "webrtc/libjingle/xmllite/qname.h" +#include "webrtc/base/scoped_ptr.h" namespace buzz { diff --git a/webrtc/libjingle/xmpp/fakexmppclient.h b/webrtc/libjingle/xmpp/fakexmppclient.h index 63c216caf2..453a7c86f1 100644 --- a/webrtc/libjingle/xmpp/fakexmppclient.h +++ b/webrtc/libjingle/xmpp/fakexmppclient.h @@ -13,7 +13,6 @@ #ifndef WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_ #define WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_ -#include #include #include diff --git a/webrtc/libjingle/xmpp/hangoutpubsubclient.h b/webrtc/libjingle/xmpp/hangoutpubsubclient.h index fecc727604..8633424adc 100644 --- a/webrtc/libjingle/xmpp/hangoutpubsubclient.h +++ b/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" diff --git a/webrtc/libjingle/xmpp/pubsubstateclient.h b/webrtc/libjingle/xmpp/pubsubstateclient.h index 07aa26dbad..e78ddb5aff 100644 --- a/webrtc/libjingle/xmpp/pubsubstateclient.h +++ b/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" diff --git a/webrtc/libjingle/xmpp/xmpplogintask.h b/webrtc/libjingle/xmpp/xmpplogintask.h index c015c2e7a1..d47388dd7a 100644 --- a/webrtc/libjingle/xmpp/xmpplogintask.h +++ b/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 { diff --git a/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc b/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc index 18fce97645..f3060bd559 100644 --- a/webrtc/libjingle/xmpp/xmpplogintask_unittest.cc +++ b/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; diff --git a/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc b/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc index 7c6d16f129..2a6ac1ffe9 100644 --- a/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc +++ b/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 #include #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 } diff --git a/webrtc/modules/congestion_controller/include/congestion_controller.h b/webrtc/modules/congestion_controller/include/congestion_controller.h index c1a6c3fbd4..284070cc21 100644 --- a/webrtc/modules/congestion_controller/include/congestion_controller.h +++ b/webrtc/modules/congestion_controller/include/congestion_controller.h @@ -14,6 +14,7 @@ #include #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" diff --git a/webrtc/modules/desktop_capture/cropping_window_capturer.h b/webrtc/modules/desktop_capture/cropping_window_capturer.h index dfeb447e44..27957ad8e7 100644 --- a/webrtc/modules/desktop_capture/cropping_window_capturer.h +++ b/webrtc/modules/desktop_capture/cropping_window_capturer.h @@ -13,6 +13,7 @@ #include +#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" diff --git a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h index dcbe6129e6..971943b275 100644 --- a/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h +++ b/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h @@ -14,6 +14,7 @@ #include #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" diff --git a/webrtc/modules/desktop_capture/desktop_capturer.h b/webrtc/modules/desktop_capture/desktop_capturer.h index ba70e01553..103740aac5 100644 --- a/webrtc/modules/desktop_capture/desktop_capturer.h +++ b/webrtc/modules/desktop_capture/desktop_capturer.h @@ -15,6 +15,7 @@ #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/desktop_capture/desktop_capture_types.h" #include "webrtc/modules/desktop_capture/shared_memory.h" diff --git a/webrtc/modules/desktop_capture/shared_memory.h b/webrtc/modules/desktop_capture/shared_memory.h index 6e15f23f6b..e1d1e7c57b 100644 --- a/webrtc/modules/desktop_capture/shared_memory.h +++ b/webrtc/modules/desktop_capture/shared_memory.h @@ -20,6 +20,7 @@ #include #include "webrtc/base/constructormagic.h" +#include "webrtc/base/scoped_ptr.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h index 5a50580e69..f43aa0d566 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h @@ -18,6 +18,7 @@ #include #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" diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h index 623c8a3003..82ef52867b 100644 --- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h +++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h @@ -18,6 +18,7 @@ #include #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" diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h index 207e749a02..d57518a3a9 100644 --- a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h +++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h @@ -14,6 +14,7 @@ #include #include "webrtc/base/constructormagic.h" +#include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/rtp_to_ntp.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h index a199755aaf..a24fc34982 100644 --- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h +++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h @@ -15,6 +15,7 @@ #include #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" diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc index 932be1bb9e..1e7b355a87 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -12,8 +12,6 @@ #include -#include - #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 = diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h index 40d1220069..9a9c73d2cc 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h @@ -16,6 +16,7 @@ #include #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" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h index 629de4e99e..fedd1dc387 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h @@ -15,6 +15,7 @@ #include +#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" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h index 1ae1c9168a..be659fe874 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h @@ -14,6 +14,7 @@ #include #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" diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h index 44de00a55f..d8040f7902 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h +++ b/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" diff --git a/webrtc/modules/utility/include/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h index 574c977cd0..305e7cf0b4 100644 --- a/webrtc/modules/utility/include/jvm_android.h +++ b/webrtc/modules/utility/include/jvm_android.h @@ -16,6 +16,7 @@ #include #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_checker.h" #include "webrtc/modules/utility/include/helpers_android.h" diff --git a/webrtc/modules/utility/include/process_thread.h b/webrtc/modules/utility/include/process_thread.h index f6913ea316..4d774521a2 100644 --- a/webrtc/modules/utility/include/process_thread.h +++ b/webrtc/modules/utility/include/process_thread.h @@ -14,6 +14,7 @@ #include #include "webrtc/typedefs.h" +#include "webrtc/base/scoped_ptr.h" namespace webrtc { class Module; diff --git a/webrtc/modules/video_coding/packet_buffer.h b/webrtc/modules/video_coding/packet_buffer.h index 2b9e51f7e4..8a1d7069f4 100644 --- a/webrtc/modules/video_coding/packet_buffer.h +++ b/webrtc/modules/video_coding/packet_buffer.h @@ -20,6 +20,7 @@ #include #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" diff --git a/webrtc/modules/video_coding/test/rtp_player.cc b/webrtc/modules/video_coding/test/rtp_player.cc index d5fa9ae936..41cd360d13 100644 --- a/webrtc/modules/video_coding/test/rtp_player.cc +++ b/webrtc/modules/video_coding/test/rtp_player.cc @@ -12,7 +12,6 @@ #include -#include #include #include @@ -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) < - loss_rate_) { // NOLINT + } else if ((rand() + 1.0) / (RAND_MAX + 1.0) < 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); diff --git a/webrtc/modules/video_processing/util/noise_estimation.h b/webrtc/modules/video_processing/util/noise_estimation.h index 294bfb3a73..16d5587b70 100644 --- a/webrtc/modules/video_processing/util/noise_estimation.h +++ b/webrtc/modules/video_processing/util/noise_estimation.h @@ -13,6 +13,7 @@ #include +#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" diff --git a/webrtc/p2p/base/asyncstuntcpsocket.h b/webrtc/p2p/base/asyncstuntcpsocket.h index 065b4c259b..90d262b00a 100644 --- a/webrtc/p2p/base/asyncstuntcpsocket.h +++ b/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 { diff --git a/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/webrtc/p2p/base/dtlstransportchannel_unittest.cc index 486b51aec9..3e34affb4b 100644 --- a/webrtc/p2p/base/dtlstransportchannel_unittest.cc +++ b/webrtc/p2p/base/dtlstransportchannel_unittest.cc @@ -42,7 +42,7 @@ static bool IsRtpLeadByte(uint8_t b) { cricket::TransportDescription MakeTransportDescription( const rtc::scoped_refptr& cert, cricket::ConnectionRole role) { - std::unique_ptr fingerprint; + rtc::scoped_ptr fingerprint; if (cert) { std::string digest_algorithm; cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_algorithm); diff --git a/webrtc/p2p/base/stunserver.h b/webrtc/p2p/base/stunserver.h index 9d1c169a50..3f7e7de5c2 100644 --- a/webrtc/p2p/base/stunserver.h +++ b/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 { diff --git a/webrtc/p2p/base/testrelayserver.h b/webrtc/p2p/base/testrelayserver.h index 7bc0beead2..ba64008e54 100644 --- a/webrtc/p2p/base/testrelayserver.h +++ b/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" diff --git a/webrtc/p2p/base/transportdescription.h b/webrtc/p2p/base/transportdescription.h index 42e45a670f..d9cd524329 100644 --- a/webrtc/p2p/base/transportdescription.h +++ b/webrtc/p2p/base/transportdescription.h @@ -17,6 +17,7 @@ #include #include "webrtc/p2p/base/p2pconstants.h" +#include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sslfingerprint.h" namespace cricket { diff --git a/webrtc/p2p/client/basicportallocator.h b/webrtc/p2p/client/basicportallocator.h index 296d717882..c66ae596c8 100644 --- a/webrtc/p2p/client/basicportallocator.h +++ b/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 { diff --git a/webrtc/p2p/client/fakeportallocator.h b/webrtc/p2p/client/fakeportallocator.h index 1e5f46eea2..76357a5ffa 100644 --- a/webrtc/p2p/client/fakeportallocator.h +++ b/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; diff --git a/webrtc/p2p/quic/quictransportchannel.h b/webrtc/p2p/quic/quictransportchannel.h index dec24d2154..2ce17f8f90 100644 --- a/webrtc/p2p/quic/quictransportchannel.h +++ b/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" diff --git a/webrtc/p2p/stunprober/stunprober.h b/webrtc/p2p/stunprober/stunprober.h index dbb67c6167..3f100c6dc1 100644 --- a/webrtc/p2p/stunprober/stunprober.h +++ b/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" diff --git a/webrtc/pc/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc index b14bd801cb..b1c4044b70 100644 --- a/webrtc/pc/mediasession_unittest.cc +++ b/webrtc/pc/mediasession_unittest.cc @@ -2404,9 +2404,9 @@ class MediaProtocolTest : public ::testing::TestWithParam { f1_.set_secure(SEC_ENABLED); f2_.set_secure(SEC_ENABLED); tdf1_.set_certificate(rtc::RTCCertificate::Create( - std::unique_ptr(new rtc::FakeSSLIdentity("id1")))); + rtc::scoped_ptr(new rtc::FakeSSLIdentity("id1")))); tdf2_.set_certificate(rtc::RTCCertificate::Create( - std::unique_ptr(new rtc::FakeSSLIdentity("id2")))); + rtc::scoped_ptr(new rtc::FakeSSLIdentity("id2")))); tdf1_.set_secure(SEC_ENABLED); tdf2_.set_secure(SEC_ENABLED); } diff --git a/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h b/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h index b00c8da3a3..04858cfbc3 100644 --- a/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h +++ b/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h @@ -13,6 +13,7 @@ #include #include "webrtc/api/jsep.h" +#include "webrtc/base/scoped_ptr.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h b/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h index 606a132da6..6ad3b6d899 100644 --- a/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h +++ b/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h @@ -13,6 +13,7 @@ #include #include "webrtc/api/mediaconstraintsinterface.h" +#include "webrtc/base/scoped_ptr.h" namespace webrtc { diff --git a/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h b/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h index d5261266b8..fd98cb6e03 100644 --- a/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h +++ b/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, diff --git a/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h b/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h index c523b527bb..5a70238b7b 100644 --- a/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h +++ b/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h @@ -13,6 +13,7 @@ #import +#include "webrtc/base/scoped_ptr.h" #include "webrtc/media/base/videocapturer.h" #include "webrtc/video_frame.h" diff --git a/webrtc/system_wrappers/include/clock.h b/webrtc/system_wrappers/include/clock.h index a209770261..8245ecd9f4 100644 --- a/webrtc/system_wrappers/include/clock.h +++ b/webrtc/system_wrappers/include/clock.h @@ -13,6 +13,7 @@ #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/system_wrappers/include/data_log_impl.h b/webrtc/system_wrappers/include/data_log_impl.h index 6d59fa8c66..c68c82985e 100644 --- a/webrtc/system_wrappers/include/data_log_impl.h +++ b/webrtc/system_wrappers/include/data_log_impl.h @@ -24,6 +24,7 @@ #include #include "webrtc/base/platform_thread.h" +#include "webrtc/base/scoped_ptr.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/system_wrappers/include/utf_util_win.h b/webrtc/system_wrappers/include/utf_util_win.h index ac91fe30fe..730aa4649e 100644 --- a/webrtc/system_wrappers/include/utf_util_win.h +++ b/webrtc/system_wrappers/include/utf_util_win.h @@ -19,6 +19,7 @@ #include #include +#include "webrtc/base/scoped_ptr.h" namespace webrtc { diff --git a/webrtc/system_wrappers/source/file_impl.h b/webrtc/system_wrappers/source/file_impl.h index 51103d648b..8764f720f3 100644 --- a/webrtc/system_wrappers/source/file_impl.h +++ b/webrtc/system_wrappers/source/file_impl.h @@ -15,6 +15,7 @@ #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/file_wrapper.h" namespace webrtc { diff --git a/webrtc/system_wrappers/source/trace_impl.h b/webrtc/system_wrappers/source/trace_impl.h index 182f5809a5..5e459765cd 100644 --- a/webrtc/system_wrappers/source/trace_impl.h +++ b/webrtc/system_wrappers/source/trace_impl.h @@ -14,6 +14,7 @@ #include #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" diff --git a/webrtc/test/configurable_frame_size_encoder.h b/webrtc/test/configurable_frame_size_encoder.h index d269441667..0da5d20c15 100644 --- a/webrtc/test/configurable_frame_size_encoder.h +++ b/webrtc/test/configurable_frame_size_encoder.h @@ -14,6 +14,7 @@ #include #include +#include "webrtc/base/scoped_ptr.h" #include "webrtc/video_encoder.h" namespace webrtc { diff --git a/webrtc/test/direct_transport.h b/webrtc/test/direct_transport.h index e1844763d9..d68bc7184e 100644 --- a/webrtc/test/direct_transport.h +++ b/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" diff --git a/webrtc/test/fake_audio_device.h b/webrtc/test/fake_audio_device.h index 77a74bac8f..39f9310cd2 100644 --- a/webrtc/test/fake_audio_device.h +++ b/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" diff --git a/webrtc/test/fake_network_pipe.h b/webrtc/test/fake_network_pipe.h index 608ff008c9..3c2db86214 100644 --- a/webrtc/test/fake_network_pipe.h +++ b/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 { diff --git a/webrtc/test/frame_generator_capturer.h b/webrtc/test/frame_generator_capturer.h index 1d6fb62664..2b34dd2c59 100644 --- a/webrtc/test/frame_generator_capturer.h +++ b/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" diff --git a/webrtc/test/test_suite.h b/webrtc/test/test_suite.h index 4b27f9f2c3..94b11cfbeb 100644 --- a/webrtc/test/test_suite.h +++ b/webrtc/test/test_suite.h @@ -20,6 +20,7 @@ #include #include "webrtc/base/constructormagic.h" +#include "webrtc/base/scoped_ptr.h" namespace webrtc { namespace test {