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1 /* | 1 /* |
2 * Copyright 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2014 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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22 #include "webrtc/api/peerconnectionfactory.h" | 22 #include "webrtc/api/peerconnectionfactory.h" |
23 #include "webrtc/api/test/fakedatachannelprovider.h" | 23 #include "webrtc/api/test/fakedatachannelprovider.h" |
24 #include "webrtc/api/test/fakevideotracksource.h" | 24 #include "webrtc/api/test/fakevideotracksource.h" |
25 #include "webrtc/api/test/mock_peerconnection.h" | 25 #include "webrtc/api/test/mock_peerconnection.h" |
26 #include "webrtc/api/test/mock_webrtcsession.h" | 26 #include "webrtc/api/test/mock_webrtcsession.h" |
27 #include "webrtc/api/videotrack.h" | 27 #include "webrtc/api/videotrack.h" |
28 #include "webrtc/base/base64.h" | 28 #include "webrtc/base/base64.h" |
29 #include "webrtc/base/fakesslidentity.h" | 29 #include "webrtc/base/fakesslidentity.h" |
30 #include "webrtc/base/gunit.h" | 30 #include "webrtc/base/gunit.h" |
31 #include "webrtc/base/network.h" | 31 #include "webrtc/base/network.h" |
| 32 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
32 #include "webrtc/media/base/fakemediaengine.h" | 33 #include "webrtc/media/base/fakemediaengine.h" |
33 #include "webrtc/media/base/test/mock_mediachannel.h" | 34 #include "webrtc/media/base/test/mock_mediachannel.h" |
34 #include "webrtc/p2p/base/faketransportcontroller.h" | 35 #include "webrtc/p2p/base/faketransportcontroller.h" |
35 #include "webrtc/pc/channelmanager.h" | 36 #include "webrtc/pc/channelmanager.h" |
36 #include "webrtc/test/gmock.h" | 37 #include "webrtc/test/gmock.h" |
37 #include "webrtc/test/gtest.h" | 38 #include "webrtc/test/gtest.h" |
38 | 39 |
39 using testing::_; | 40 using testing::_; |
40 using testing::DoAll; | 41 using testing::DoAll; |
41 using testing::Field; | 42 using testing::Field; |
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486 StatsCollectorTest() | 487 StatsCollectorTest() |
487 : worker_thread_(rtc::Thread::Current()), | 488 : worker_thread_(rtc::Thread::Current()), |
488 network_thread_(rtc::Thread::Current()), | 489 network_thread_(rtc::Thread::Current()), |
489 media_engine_(new cricket::FakeMediaEngine()), | 490 media_engine_(new cricket::FakeMediaEngine()), |
490 channel_manager_(new cricket::ChannelManager(media_engine_, | 491 channel_manager_(new cricket::ChannelManager(media_engine_, |
491 worker_thread_, | 492 worker_thread_, |
492 network_thread_)), | 493 network_thread_)), |
493 media_controller_( | 494 media_controller_( |
494 webrtc::MediaControllerInterface::Create(cricket::MediaConfig(), | 495 webrtc::MediaControllerInterface::Create(cricket::MediaConfig(), |
495 worker_thread_, | 496 worker_thread_, |
496 channel_manager_.get())), | 497 channel_manager_.get(), |
| 498 &event_log_)), |
497 session_(media_controller_.get()) { | 499 session_(media_controller_.get()) { |
498 // By default, we ignore session GetStats calls. | 500 // By default, we ignore session GetStats calls. |
499 EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false)); | 501 EXPECT_CALL(session_, GetTransportStats(_)).WillRepeatedly(Return(false)); |
500 // Add default returns for mock classes. | 502 // Add default returns for mock classes. |
501 EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull()); | 503 EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull()); |
502 EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull()); | 504 EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull()); |
503 EXPECT_CALL(pc_, session()).WillRepeatedly(Return(&session_)); | 505 EXPECT_CALL(pc_, session()).WillRepeatedly(Return(&session_)); |
504 EXPECT_CALL(pc_, sctp_data_channels()) | 506 EXPECT_CALL(pc_, sctp_data_channels()) |
505 .WillRepeatedly(ReturnRef(data_channels_)); | 507 .WillRepeatedly(ReturnRef(data_channels_)); |
506 } | 508 } |
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744 EXPECT_EQ(rtc::SSLStreamAdapter::SslCipherSuiteToName( | 746 EXPECT_EQ(rtc::SSLStreamAdapter::SslCipherSuiteToName( |
745 internal::TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA), | 747 internal::TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA), |
746 dtls_cipher_suite); | 748 dtls_cipher_suite); |
747 std::string srtp_crypto_suite = | 749 std::string srtp_crypto_suite = |
748 ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports, | 750 ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports, |
749 StatsReport::kStatsValueNameSrtpCipher); | 751 StatsReport::kStatsValueNameSrtpCipher); |
750 EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80), | 752 EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80), |
751 srtp_crypto_suite); | 753 srtp_crypto_suite); |
752 } | 754 } |
753 | 755 |
| 756 webrtc::RtcEventLogNullImpl event_log_; |
754 rtc::Thread* const worker_thread_; | 757 rtc::Thread* const worker_thread_; |
755 rtc::Thread* const network_thread_; | 758 rtc::Thread* const network_thread_; |
756 cricket::FakeMediaEngine* media_engine_; | 759 cricket::FakeMediaEngine* media_engine_; |
757 std::unique_ptr<cricket::ChannelManager> channel_manager_; | 760 std::unique_ptr<cricket::ChannelManager> channel_manager_; |
758 std::unique_ptr<webrtc::MediaControllerInterface> media_controller_; | 761 std::unique_ptr<webrtc::MediaControllerInterface> media_controller_; |
759 MockWebRtcSession session_; | 762 MockWebRtcSession session_; |
760 MockPeerConnection pc_; | 763 MockPeerConnection pc_; |
761 FakeDataChannelProvider data_channel_provider_; | 764 FakeDataChannelProvider data_channel_provider_; |
762 SessionStats session_stats_; | 765 SessionStats session_stats_; |
763 rtc::scoped_refptr<webrtc::MediaStream> stream_; | 766 rtc::scoped_refptr<webrtc::MediaStream> stream_; |
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1882 cricket::VoiceSenderInfo new_voice_sender_info; | 1885 cricket::VoiceSenderInfo new_voice_sender_info; |
1883 InitVoiceSenderInfo(&new_voice_sender_info); | 1886 InitVoiceSenderInfo(&new_voice_sender_info); |
1884 cricket::VoiceMediaInfo new_stats_read; | 1887 cricket::VoiceMediaInfo new_stats_read; |
1885 reports.clear(); | 1888 reports.clear(); |
1886 SetupAndVerifyAudioTrackStats( | 1889 SetupAndVerifyAudioTrackStats( |
1887 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName, | 1890 new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName, |
1888 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports); | 1891 media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports); |
1889 } | 1892 } |
1890 | 1893 |
1891 } // namespace webrtc | 1894 } // namespace webrtc |
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