| /* |
| * Copyright (c) 2014 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 file contains the WebRTC suppressions for ThreadSanitizer. |
| // Please refer to |
| // http://dev.chromium.org/developers/testing/threadsanitizer-tsan-v2 |
| // for more info. |
| |
| #if defined(THREAD_SANITIZER) |
| |
| // Please make sure the code below declares a single string variable |
| // kTSanDefaultSuppressions contains TSan suppressions delimited by newlines. |
| // See http://dev.chromium.org/developers/testing/threadsanitizer-tsan-v2 |
| // for the instructions on writing suppressions. |
| char kTSanDefaultSuppressions[] = |
| |
| // WebRTC specific suppressions. |
| |
| // Usage of trace callback and trace level is racy in libjingle_media_unittests. |
| // https://code.google.com/p/webrtc/issues/detail?id=3372 |
| "race:webrtc::TraceImpl::WriteToFile\n" |
| "race:webrtc::VideoEngine::SetTraceFilter\n" |
| "race:webrtc::VoiceEngine::SetTraceFilter\n" |
| "race:webrtc::Trace::set_level_filter\n" |
| "race:webrtc::GetStaticInstance<webrtc::TraceImpl>\n" |
| |
| // Audio processing |
| // https://code.google.com/p/webrtc/issues/detail?id=2521 for details. |
| "race:webrtc/modules/audio_processing/aec/aec_core.c\n" |
| "race:webrtc/modules/audio_processing/aec/aec_rdft.c\n" |
| |
| // libjingle_p2p_unittest |
| // https://code.google.com/p/webrtc/issues/detail?id=2079 |
| "race:webrtc/base/messagequeue.cc\n" |
| "race:webrtc/base/testclient.cc\n" |
| "race:webrtc/base/virtualsocketserver.cc\n" |
| "race:talk/base/messagequeue.cc\n" |
| "race:talk/base/testclient.cc\n" |
| "race:talk/base/virtualsocketserver.cc\n" |
| "race:talk/p2p/base/stunserver_unittest.cc\n" |
| |
| // libjingle_unittest |
| // https://code.google.com/p/webrtc/issues/detail?id=2080 |
| "race:webrtc/base/logging.cc\n" |
| "race:webrtc/base/sharedexclusivelock_unittest.cc\n" |
| "race:webrtc/base/signalthread_unittest.cc\n" |
| "race:webrtc/base/thread.cc\n" |
| "race:talk/base/logging.cc\n" |
| "race:talk/base/sharedexclusivelock_unittest.cc\n" |
| "race:talk/base/signalthread_unittest.cc\n" |
| "race:talk/base/thread.cc\n" |
| |
| // third_party/usrsctp |
| // TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492 |
| "race:user_sctp_timer_iterate\n" |
| |
| // Potential deadlocks detected after roll in r6516. |
| // https://code.google.com/p/webrtc/issues/detail?id=3509 |
| "deadlock:cricket::WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame\n" |
| "deadlock:cricket::WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer\n" |
| "deadlock:talk_base::AsyncResolver::~AsyncResolver\n" |
| "deadlock:webrtc::ProcessThreadImpl::RegisterModule\n" |
| "deadlock:webrtc::RTCPReceiver::SetSsrcs\n" |
| "deadlock:webrtc::RTPSenderAudio::RegisterAudioPayload\n" |
| "deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n" |
| "deadlock:webrtc::vcm::VideoReceiver::RegisterPacketRequestCallback\n" |
| "deadlock:webrtc::ViECaptureImpl::ConnectCaptureDevice\n" |
| "deadlock:webrtc::ViEChannel::StartSend\n" |
| "deadlock:webrtc::ViECodecImpl::GetSendSideDelay\n" |
| "deadlock:webrtc::ViEEncoder::OnLocalSsrcChanged\n" |
| "deadlock:webrtc::ViESender::RegisterSendTransport\n" |
| |
| // End of suppressions. |
| ; // Please keep this semicolon. |
| |
| #endif // THREAD_SANITIZER |
