2014-06-27 09:18:51 +00:00
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/*
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* Copyright (c) 2014 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 ThreadSanitizer.
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// Please refer to
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// http://dev.chromium.org/developers/testing/threadsanitizer-tsan-v2
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// for more info.
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#if defined(THREAD_SANITIZER)
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// Please make sure the code below declares a single string variable
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// kTSanDefaultSuppressions contains TSan suppressions delimited by newlines.
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// See http://dev.chromium.org/developers/testing/threadsanitizer-tsan-v2
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// for the instructions on writing suppressions.
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char kTSanDefaultSuppressions[] =
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// WebRTC specific suppressions.
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2014-09-11 14:59:06 +00:00
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// Split up suppressions covered previously by thread.cc and
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// messagequeue.cc.
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"race:vp8cx_remove_encoder_threads\n"
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2016-03-04 14:39:28 -08:00
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"race:third_party/libvpx/source/libvpx/vp9/common/vp9_scan.h\n"
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2014-09-11 14:59:06 +00:00
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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_unittests
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2015-03-02 09:40:44 +00:00
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// https://code.google.com/p/webrtc/issues/detail?id=2080
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2017-09-15 06:47:31 +02:00
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"race:rtc_base/logging.cc\n"
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2014-10-09 15:41:40 +00:00
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2014-06-27 09:18:51 +00:00
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// Potential deadlocks detected after roll in r6516.
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// https://code.google.com/p/webrtc/issues/detail?id=3509
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2014-07-02 14:19:05 +00:00
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"deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
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2014-06-27 09:18:51 +00:00
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2015-01-08 12:38:42 +00:00
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// TODO(pbos): Trace events are racy due to lack of proper POD atomics.
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// https://code.google.com/p/webrtc/issues/detail?id=2497
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"race:*trace_event_unique_catstatic*\n"
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2015-10-20 13:37:30 +02:00
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// Race between InitCpuFlags and TestCpuFlag in libyuv.
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// https://code.google.com/p/libyuv/issues/detail?id=508
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2015-11-02 13:04:23 +01:00
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"race:InitCpuFlags\n"
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2015-10-20 13:37:30 +02:00
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2017-02-23 19:05:39 -08:00
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// Test-only race due to PeerConnection::session() being virtual for
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// testing. The stats collector may call session() before or after the
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// destructor begins executing, which modifies the vtable.
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"race:*RTCStatsIntegrationTest_GetsStatsWhileDestroyingPeerConnections_"
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"Test::TestBody\n"
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2019-07-22 19:46:02 +02:00
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// http://crbug.com/244856
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"race:libpulsecommon*.so\n"
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2022-03-15 16:01:52 +01:00
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// https://crbug.com/1158622
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"race:absl::synchronization_internal::Waiter::Post\n"
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2014-06-27 09:18:51 +00:00
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// End of suppressions.
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; // Please keep this semicolon.
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#endif // THREAD_SANITIZER
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