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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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1755 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); | 1755 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
1756 FakeConstraints setup_constraints; | 1756 FakeConstraints setup_constraints; |
1757 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | 1757 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, |
1758 true); | 1758 true); |
1759 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | 1759 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); |
1760 receiving_client()->RemoveSdesCryptoFromReceivedSdp(true); | 1760 receiving_client()->RemoveSdesCryptoFromReceivedSdp(true); |
1761 LocalP2PTest(); | 1761 LocalP2PTest(); |
1762 VerifyRenderedSize(640, 480); | 1762 VerifyRenderedSize(640, 480); |
1763 } | 1763 } |
1764 | 1764 |
| 1765 // This test verifies that the negotiation will succeed with data channel only |
| 1766 // in max-bundle mode. |
| 1767 TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) { |
| 1768 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; |
| 1769 rtc_config.bundle_policy = |
| 1770 webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle; |
| 1771 ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config)); |
| 1772 initializing_client()->CreateDataChannel(); |
| 1773 initializing_client()->Negotiate(); |
| 1774 } |
| 1775 |
1765 // This test sets up a Jsep call between two parties, and the callee only | 1776 // This test sets up a Jsep call between two parties, and the callee only |
1766 // accept to receive video. | 1777 // accept to receive video. |
1767 TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { | 1778 TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { |
1768 ASSERT_TRUE(CreateTestClients()); | 1779 ASSERT_TRUE(CreateTestClients()); |
1769 receiving_client()->SetReceiveAudioVideo(false, true); | 1780 receiving_client()->SetReceiveAudioVideo(false, true); |
1770 LocalP2PTest(); | 1781 LocalP2PTest(); |
1771 } | 1782 } |
1772 | 1783 |
1773 // This test sets up a Jsep call between two parties, and the callee only | 1784 // This test sets up a Jsep call between two parties, and the callee only |
1774 // accept to receive audio. | 1785 // accept to receive audio. |
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2760 server.urls.push_back("turn:hostname2"); | 2771 server.urls.push_back("turn:hostname2"); |
2761 servers.push_back(server); | 2772 servers.push_back(server); |
2762 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); | 2773 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); |
2763 EXPECT_EQ(2U, turn_servers_.size()); | 2774 EXPECT_EQ(2U, turn_servers_.size()); |
2764 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); | 2775 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); |
2765 } | 2776 } |
2766 | 2777 |
2767 #endif // if !defined(THREAD_SANITIZER) | 2778 #endif // if !defined(THREAD_SANITIZER) |
2768 | 2779 |
2769 } // namespace | 2780 } // namespace |
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