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Issue 2534783003: Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots. (Closed)
Patch Set: Created 4 years ago
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1 /* 1 /*
2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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175 // https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details. 175 // https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
176 #if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC) 176 #if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
177 TEST_F(PeerConnectionEndToEndTest, Call) { 177 TEST_F(PeerConnectionEndToEndTest, Call) {
178 CreatePcs(); 178 CreatePcs();
179 GetAndAddUserMedia(); 179 GetAndAddUserMedia();
180 Negotiate(); 180 Negotiate();
181 WaitForCallEstablished(); 181 WaitForCallEstablished();
182 } 182 }
183 #endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC) 183 #endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
184 184
185 #if !defined(ADDRESS_SANITIZER)
185 TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) { 186 TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
186 FakeConstraints pc_constraints; 187 FakeConstraints pc_constraints;
187 pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, 188 pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
188 false); 189 false);
189 CreatePcs(&pc_constraints); 190 CreatePcs(&pc_constraints);
190 GetAndAddUserMedia(); 191 GetAndAddUserMedia();
191 Negotiate(); 192 Negotiate();
192 WaitForCallEstablished(); 193 WaitForCallEstablished();
193 } 194 }
195 #endif // !defined(ADDRESS_SANITIZER)
194 196
195 // Verifies that a DataChannel created before the negotiation can transition to 197 // Verifies that a DataChannel created before the negotiation can transition to
196 // "OPEN" and transfer data. 198 // "OPEN" and transfer data.
197 TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { 199 TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
198 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); 200 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
199 201
200 CreatePcs(); 202 CreatePcs();
201 203
202 webrtc::DataChannelInit init; 204 webrtc::DataChannelInit init;
203 rtc::scoped_refptr<DataChannelInterface> caller_dc( 205 rtc::scoped_refptr<DataChannelInterface> caller_dc(
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446 WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0); 448 WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
447 // This removes the reference to the remote data channel that we hold. 449 // This removes the reference to the remote data channel that we hold.
448 callee_signaled_data_channels_.clear(); 450 callee_signaled_data_channels_.clear();
449 caller_dc->Close(); 451 caller_dc->Close();
450 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait); 452 EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
451 453
452 // Wait for a bit longer so the remote data channel will receive the 454 // Wait for a bit longer so the remote data channel will receive the
453 // close message and be destroyed. 455 // close message and be destroyed.
454 rtc::Thread::Current()->ProcessMessages(100); 456 rtc::Thread::Current()->ProcessMessages(100);
455 } 457 }
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