2015-02-18 10:37:51 +00:00
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains the WebRTC suppressions for LeakSanitizer.
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// You can also pass additional suppressions via LSAN_OPTIONS:
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// LSAN_OPTIONS=suppressions=/path/to/suppressions. Please refer to
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// http://dev.chromium.org/developers/testing/leaksanitizer for more info.
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#if defined(LEAK_SANITIZER)
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// Please make sure the code below declares a single string variable
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// kLSanDefaultSuppressions which contains LSan suppressions delimited by
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// newlines. See http://dev.chromium.org/developers/testing/leaksanitizer
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// for the instructions on writing suppressions.
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char kLSanDefaultSuppressions[] =
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// ============ Leaks in third-party code shared with Chromium =============
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// These entries are copied from build/sanitizers/lsan_suppressions.cc in
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// Chromium. Please don't add new entries here unless they're present in
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// there.
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// False positives in libfontconfig. http://crbug.com/39050
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"leak:libfontconfig\n"
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// Leaks in Nvidia's libGL.
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"leak:libGL.so\n"
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// XRandR has several one time leaks.
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"leak:libxrandr\n"
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// xrandr leak. http://crbug.com/119677
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"leak:XRRFindDisplay\n"
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// ========== Leaks in third-party code not shared with Chromium ===========
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// None known so far.
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// ================ Leaks in WebRTC code ================
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// PLEASE DO NOT ADD SUPPRESSIONS FOR NEW LEAKS.
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// Instead, commits that introduce memory leaks should be reverted.
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// Suppressing the leak is acceptable in some cases when reverting is
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// impossible, i.e. when enabling leak detection for the first time for a
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// test target with pre-existing leaks.
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// rtc_unittest
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2017-03-23 15:45:49 -07:00
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// https://code.google.com/p/webrtc/issues/detail?id=3827 for details.
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"leak:rtc::unstarted_task_test_DoNotDeleteTask2_Test::TestBody\n"
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"leak:rtc::HttpServer::HandleConnection\n"
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"leak:rtc::HttpServer::Connection::onHttpHeaderComplete\n"
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"leak:rtc::HttpResponseData::set_success\n"
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"leak:rtc::HttpData::changeHeader\n"
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2015-02-18 10:37:51 +00:00
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// https://code.google.com/p/webrtc/issues/detail?id=4149 for details.
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"leak:StartDNSLookup\n"
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2018-06-19 15:03:05 +02:00
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Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
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// rtc_media_unittests
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2015-02-18 10:37:51 +00:00
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"leak:cricket::FakeNetworkInterface::SetOption\n"
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"leak:CodecTest_TestCodecOperators_Test::TestBody\n"
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"leak:VideoEngineTest*::ConstrainNewCodecBody\n"
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"leak:VideoMediaChannelTest*::AddRemoveRecvStreams\n"
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"leak:WebRtcVideoCapturerTest_TestCapture_Test::TestBody\n"
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"leak:WebRtcVideoEngineTestFake_MultipleSendStreamsWithOneCapturer_Test::"
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"TestBody\n"
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"leak:WebRtcVideoEngineTestFake_SetBandwidthInConference_Test::TestBody\n"
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"leak:WebRtcVideoEngineTestFake_SetSendCodecsRejectBadFormat_Test::"
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"TestBody\n"
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2018-06-19 15:03:05 +02:00
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2016-01-19 06:20:00 -08:00
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// peerconnection_unittests
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2015-02-18 10:37:51 +00:00
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// https://code.google.com/p/webrtc/issues/detail?id=2528
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"leak:cricket::FakeVideoMediaChannel::~FakeVideoMediaChannel\n"
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"leak:cricket::MediaSessionDescriptionFactory::CreateAnswer\n"
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"leak:cricket::MediaSessionDescriptionFactory::CreateOffer\n"
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"leak:DtmfSenderTest_InsertEmptyTonesToCancelPreviousTask_Test::TestBody\n"
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"leak:sigslot::_signal_base2*::~_signal_base2\n"
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"leak:testing::internal::CmpHelperEQ\n"
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"leak:webrtc::AudioDeviceLinuxALSA::InitMicrophone\n"
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"leak:webrtc::AudioDeviceLinuxALSA::InitSpeaker\n"
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"leak:webrtc::CreateIceCandidate\n"
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"leak:webrtc::WebRtcIdentityRequestObserver::OnSuccess\n"
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"leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateAnswer\n"
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"leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateOffer\n"
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"leak:PeerConnectionInterfaceTest_SsrcInOfferAnswer_Test::TestBody\n"
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"leak:PeerConnectionInterfaceTest_CloseAndTestMethods_Test::TestBody\n"
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"leak:WebRtcSdpTest::TestDeserializeRtcpFb\n"
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"leak:WebRtcSdpTest::TestSerialize\n"
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"leak:WebRtcSdpTest_SerializeSessionDescriptionWithDataChannelAndBandwidth_"
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"Test::TestBody\n"
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"leak:WebRtcSdpTest_SerializeSessionDescriptionWithBandwidth_Test::"
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"TestBody\n"
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"leak:WebRtcSessionTest::SetLocalDescriptionExpectError\n"
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"leak:WebRtcSessionTest_TestAVOfferWithAudioOnlyAnswer_Test::TestBody\n"
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// PLEASE READ ABOVE BEFORE ADDING NEW SUPPRESSIONS.
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2018-06-19 15:03:05 +02:00
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2015-02-18 10:37:51 +00:00
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// End of suppressions.
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; // Please keep this semicolon.
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#endif // LEAK_SANITIZER
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