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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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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1772 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | 1772 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), |
1773 kMaxWaitMs); | 1773 kMaxWaitMs); |
1774 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), | 1774 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), |
1775 kMaxWaitMs); | 1775 kMaxWaitMs); |
1776 } | 1776 } |
1777 | 1777 |
1778 // This test sets up a Jsep call with SCTP DataChannel and verifies the | 1778 // This test sets up a Jsep call with SCTP DataChannel and verifies the |
1779 // negotiation is completed without error. | 1779 // negotiation is completed without error. |
1780 #ifdef HAVE_SCTP | 1780 #ifdef HAVE_SCTP |
1781 // Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 | 1781 // Disabled on Win dbg: https://bugs.chromium.org/p/webrtc/issues/detail?id=5659 |
1782 #if defined(WIN) && defined(_DEBUG) | 1782 #if defined(WEBRTC_WIN) && defined(_DEBUG) |
1783 #define MAYBE_CreateOfferWithSctpDataChannel \ | 1783 #define MAYBE_CreateOfferWithSctpDataChannel \ |
1784 DISABLED_CreateOfferWithSctpDataChannel | 1784 DISABLED_CreateOfferWithSctpDataChannel |
1785 #else | 1785 #else |
1786 #define MAYBE_CreateOfferWithSctpDataChannel CreateOfferWithSctpDataChannel | 1786 #define MAYBE_CreateOfferWithSctpDataChannel CreateOfferWithSctpDataChannel |
1787 #endif | 1787 #endif |
1788 TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) { | 1788 TEST_F(P2PTestConductor, MAYBE_CreateOfferWithSctpDataChannel) { |
1789 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 1789 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
1790 FakeConstraints constraints; | 1790 FakeConstraints constraints; |
1791 constraints.SetMandatory( | 1791 constraints.SetMandatory( |
1792 MediaConstraintsInterface::kEnableDtlsSrtp, true); | 1792 MediaConstraintsInterface::kEnableDtlsSrtp, true); |
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2094 server.urls.push_back("turn:hostname2"); | 2094 server.urls.push_back("turn:hostname2"); |
2095 servers.push_back(server); | 2095 servers.push_back(server); |
2096 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); | 2096 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); |
2097 EXPECT_EQ(2U, turn_servers_.size()); | 2097 EXPECT_EQ(2U, turn_servers_.size()); |
2098 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); | 2098 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); |
2099 } | 2099 } |
2100 | 2100 |
2101 #endif // if !defined(THREAD_SANITIZER) | 2101 #endif // if !defined(THREAD_SANITIZER) |
2102 | 2102 |
2103 } // namespace | 2103 } // namespace |
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