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

Browse Source 
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}
This commit is contained in:
kwiberg
2016-05-11 05:07:26 -07:00
committed by Commit bot
parent 7e3968e46c
commit 6ab3db249b
74 changed files with 133 additions and 26 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
talk/LICENSE_THIRD_PARTY
View File

@ -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

1
webrtc/api/audiotrack.h
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@ -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"

1
webrtc/api/jsepsessiondescription.h
View File

@ -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 {

1
webrtc/api/localaudiosource.h
View File

@ -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.

1
webrtc/api/mediastreamprovider.h
View File

@ -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"

1
webrtc/api/peerconnection.h
View File

@ -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 {

1
webrtc/api/peerconnectionfactory.h
View File

@ -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"

18
webrtc/api/peerconnectioninterface_unittest.cc
View File

@ -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()));

1
webrtc/api/rtpsender.h
View File

@ -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 {

1
webrtc/api/statstypes.h
View File

@ -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"

1
webrtc/api/test/fakeaudiocapturemodule.h
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@ -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"

1
webrtc/api/videocapturertracksource.h
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@ -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"

1
webrtc/audio_send_stream.h
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@ -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"

1
webrtc/base/BUILD.gn
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@ -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",

1
webrtc/base/base.gyp
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@ -75,6 +75,7 @@
'refcount.h',
'safe_conversions.h',
'safe_conversions_impl.h',
'scoped_ptr.h',
'scoped_ref_ptr.h',
'stringencode.cc',
'stringencode.h',

1
webrtc/base/messagehandler.h
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@ -15,6 +15,7 @@
#include <utility>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
namespace rtc {

49
webrtc/base/scoped_ptr.h Normal file
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@ -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__

1
webrtc/common_audio/real_fourier.h
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@ -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

1
webrtc/common_video/bitrate_adjuster.cc
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@ -10,7 +10,6 @@
#include "webrtc/common_video/include/bitrate_adjuster.h"
#include <algorithm>
#include <cmath>
#include "webrtc/base/checks.h"

1
webrtc/examples/peerconnection/client/conductor.h
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@ -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;

1
webrtc/examples/peerconnection/client/peer_connection_client.h
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@ -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"

1
webrtc/libjingle/xmllite/xmlbuilder.h
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@ -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"

1
webrtc/libjingle/xmllite/xmlelement.h
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@ -15,6 +15,7 @@
#include <string>
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/base/scoped_ptr.h"
namespace buzz {

1
webrtc/libjingle/xmllite/xmlnsstack.h
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@ -15,6 +15,7 @@
#include <string>
#include <vector>
#include "webrtc/libjingle/xmllite/qname.h"
#include "webrtc/base/scoped_ptr.h"
namespace buzz {

1
webrtc/libjingle/xmpp/fakexmppclient.h
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@ -13,7 +13,6 @@
#ifndef WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
#define WEBRTC_LIBJINGLE_XMPP_FAKEXMPPCLIENT_H_
#include <algorithm>
#include <string>
#include <vector>

1
webrtc/libjingle/xmpp/hangoutpubsubclient.h
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@ -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"

1
webrtc/libjingle/xmpp/pubsubstateclient.h
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@ -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"

1
webrtc/libjingle/xmpp/xmpplogintask.h
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@ -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 {

1
webrtc/libjingle/xmpp/xmpplogintask_unittest.cc
View File

@ -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;

5
webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
View File

@ -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
}

1
webrtc/modules/congestion_controller/include/congestion_controller.h
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@ -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"

1
webrtc/modules/desktop_capture/cropping_window_capturer.h
View File

@ -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"

1
webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
View File

@ -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"

1
webrtc/modules/desktop_capture/desktop_capturer.h
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@ -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"

1
webrtc/modules/desktop_capture/shared_memory.h
View File

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

1
webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
View File

@ -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"

1
webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
View File

@ -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"

1
webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
View File

@ -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 {

1
webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
View File

@ -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"

6
webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -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 =

1
webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
View File

@ -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"

1
webrtc/modules/rtp_rtcp/source/rtcp_utility.h
View File

@ -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"

1
webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
View File

@ -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"

1
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
View File

@ -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"

1
webrtc/modules/utility/include/jvm_android.h
View File

@ -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"

1
webrtc/modules/utility/include/process_thread.h
View File

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

1
webrtc/modules/video_coding/packet_buffer.h
View File

@ -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"

6
webrtc/modules/video_coding/test/rtp_player.cc
View File

@ -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) <
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);

1
webrtc/modules/video_processing/util/noise_estimation.h
View File

@ -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"

1
webrtc/p2p/base/asyncstuntcpsocket.h
View File

@ -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 {

2
webrtc/p2p/base/dtlstransportchannel_unittest.cc
View File

@ -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);

1
webrtc/p2p/base/stunserver.h
View File

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

1
webrtc/p2p/base/testrelayserver.h
View File

@ -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"

1
webrtc/p2p/base/transportdescription.h
View File

@ -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 {

1
webrtc/p2p/client/basicportallocator.h
View File

@ -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 {

1
webrtc/p2p/client/fakeportallocator.h
View File

@ -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;

1
webrtc/p2p/quic/quictransportchannel.h
View File

@ -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"

1
webrtc/p2p/stunprober/stunprober.h
View File

@ -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"

4
webrtc/pc/mediasession_unittest.cc
View File

@ -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);
}

1
webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h
View File

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

1
webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h
View File

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

1
webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h
View File

@ -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,

1
webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.h
View File

@ -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"

1
webrtc/system_wrappers/include/clock.h
View File

@ -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"

1
webrtc/system_wrappers/include/data_log_impl.h
View File

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

1
webrtc/system_wrappers/include/utf_util_win.h
View File

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

1
webrtc/system_wrappers/source/file_impl.h
View File

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

1
webrtc/system_wrappers/source/trace_impl.h
View File

@ -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"

1
webrtc/test/configurable_frame_size_encoder.h
View File

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

1
webrtc/test/direct_transport.h
View File

@ -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"

1
webrtc/test/fake_audio_device.h
View File

@ -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"

1
webrtc/test/fake_network_pipe.h
View File

@ -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 {

1
webrtc/test/frame_generator_capturer.h
View File

@ -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"

1
webrtc/test/test_suite.h
View File

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