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Side by Side Diff: webrtc/api/peerconnection_unittest.cc

Issue 2398963003: Move usage of QualityScaler to ViEEncoder. (Closed)
Patch Set: rebase Created 4 years ago
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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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1458 // Check that the expected number of frames have arrived. 1458 // Check that the expected number of frames have arrived.
1459 EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count), 1459 EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count),
1460 kMaxWaitForFramesMs); 1460 kMaxWaitForFramesMs);
1461 } 1461 }
1462 1462
1463 void SetupAndVerifyDtlsCall() { 1463 void SetupAndVerifyDtlsCall() {
1464 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); 1464 MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
1465 FakeConstraints setup_constraints; 1465 FakeConstraints setup_constraints;
1466 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, 1466 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
1467 true); 1467 true);
1468 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); 1468 // Disable resolution adaptation, we don't want it interfering with the
1469 // test results.
1470 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
1471 rtc_config.set_cpu_adaptation(false);
1472
1473 ASSERT_TRUE(CreateTestClients(&setup_constraints, nullptr, &rtc_config,
1474 &setup_constraints, nullptr, &rtc_config));
1469 LocalP2PTest(); 1475 LocalP2PTest();
1470 VerifyRenderedSize(640, 480); 1476 VerifyRenderedSize(640, 480);
1471 } 1477 }
1472 1478
1473 PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() { 1479 PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
1474 FakeConstraints setup_constraints; 1480 FakeConstraints setup_constraints;
1475 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, 1481 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
1476 true); 1482 true);
1483 // Disable resolution adaptation, we don't want it interfering with the
1484 // test results.
1485 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
1486 rtc_config.set_cpu_adaptation(false);
1477 1487
1478 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( 1488 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
1479 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? 1489 rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
1480 new FakeRTCCertificateGenerator() : nullptr); 1490 new FakeRTCCertificateGenerator() : nullptr);
1481 cert_generator->use_alternate_key(); 1491 cert_generator->use_alternate_key();
1482 1492
1483 // Make sure the new client is using a different certificate. 1493 // Make sure the new client is using a different certificate.
1484 return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore( 1494 return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
1485 "New Peer: ", &setup_constraints, nullptr, nullptr, 1495 "New Peer: ", &setup_constraints, nullptr, &rtc_config,
1486 std::move(cert_generator), prefer_constraint_apis_, 1496 std::move(cert_generator), prefer_constraint_apis_,
1487 network_thread_.get(), worker_thread_.get()); 1497 network_thread_.get(), worker_thread_.get());
1488 } 1498 }
1489 1499
1490 void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) { 1500 void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
1491 // Messages may get lost on the unreliable DataChannel, so we send multiple 1501 // Messages may get lost on the unreliable DataChannel, so we send multiple
1492 // times to avoid test flakiness. 1502 // times to avoid test flakiness.
1493 static const size_t kSendAttempts = 5; 1503 static const size_t kSendAttempts = 5;
1494 1504
1495 for (size_t i = 0; i < kSendAttempts; ++i) { 1505 for (size_t i = 0; i < kSendAttempts; ++i) {
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2774 server.urls.push_back("turn:hostname2"); 2784 server.urls.push_back("turn:hostname2");
2775 servers.push_back(server); 2785 servers.push_back(server);
2776 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); 2786 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
2777 EXPECT_EQ(2U, turn_servers_.size()); 2787 EXPECT_EQ(2U, turn_servers_.size());
2778 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); 2788 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
2779 } 2789 }
2780 2790
2781 #endif // if !defined(THREAD_SANITIZER) 2791 #endif // if !defined(THREAD_SANITIZER)
2782 2792
2783 } // namespace 2793 } // namespace
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