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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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91 using webrtc::StreamCollectionInterface; | 91 using webrtc::StreamCollectionInterface; |
92 | 92 |
93 static const int kMaxWaitMs = 10000; | 93 static const int kMaxWaitMs = 10000; |
94 // Disable for TSan v2, see | 94 // Disable for TSan v2, see |
95 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. | 95 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. |
96 // This declaration is also #ifdef'd as it causes uninitialized-variable | 96 // This declaration is also #ifdef'd as it causes uninitialized-variable |
97 // warnings. | 97 // warnings. |
98 #if !defined(THREAD_SANITIZER) | 98 #if !defined(THREAD_SANITIZER) |
99 static const int kMaxWaitForStatsMs = 3000; | 99 static const int kMaxWaitForStatsMs = 3000; |
100 #endif | 100 #endif |
101 static const int kMaxWaitForActivationMs = 5000; | |
102 static const int kMaxWaitForFramesMs = 10000; | 101 static const int kMaxWaitForFramesMs = 10000; |
103 static const int kEndAudioFrameCount = 3; | 102 static const int kEndAudioFrameCount = 3; |
104 static const int kEndVideoFrameCount = 3; | 103 static const int kEndVideoFrameCount = 3; |
105 | 104 |
106 static const char kStreamLabelBase[] = "stream_label"; | 105 static const char kStreamLabelBase[] = "stream_label"; |
107 static const char kVideoTrackLabelBase[] = "video_track"; | 106 static const char kVideoTrackLabelBase[] = "video_track"; |
108 static const char kAudioTrackLabelBase[] = "audio_track"; | 107 static const char kAudioTrackLabelBase[] = "audio_track"; |
109 static const char kDataChannelLabel[] = "data_channel"; | 108 static const char kDataChannelLabel[] = "data_channel"; |
110 | 109 |
111 // Disable for TSan v2, see | 110 // Disable for TSan v2, see |
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282 } | 281 } |
283 | 282 |
284 void AddMediaStream(bool audio, bool video) { | 283 void AddMediaStream(bool audio, bool video) { |
285 std::string stream_label = | 284 std::string stream_label = |
286 kStreamLabelBase + | 285 kStreamLabelBase + |
287 rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count())); | 286 rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count())); |
288 rtc::scoped_refptr<MediaStreamInterface> stream = | 287 rtc::scoped_refptr<MediaStreamInterface> stream = |
289 peer_connection_factory_->CreateLocalMediaStream(stream_label); | 288 peer_connection_factory_->CreateLocalMediaStream(stream_label); |
290 | 289 |
291 if (audio && can_receive_audio()) { | 290 if (audio && can_receive_audio()) { |
292 stream->AddTrack(CreateLocalAudioTrack(stream_label)); | 291 FakeConstraints constraints; |
| 292 // Disable highpass filter so that we can get all the test audio frames. |
| 293 constraints.AddMandatory( |
| 294 MediaConstraintsInterface::kHighpassFilter, false); |
| 295 rtc::scoped_refptr<webrtc::AudioSourceInterface> source = |
| 296 peer_connection_factory_->CreateAudioSource(&constraints); |
| 297 // TODO(perkj): Test audio source when it is implemented. Currently audio |
| 298 // always use the default input. |
| 299 std::string label = stream_label + kAudioTrackLabelBase; |
| 300 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( |
| 301 peer_connection_factory_->CreateAudioTrack(label, source)); |
| 302 stream->AddTrack(audio_track); |
293 } | 303 } |
294 if (video && can_receive_video()) { | 304 if (video && can_receive_video()) { |
295 stream->AddTrack(CreateLocalVideoTrack(stream_label)); | 305 stream->AddTrack(CreateLocalVideoTrack(stream_label)); |
296 } | 306 } |
297 | 307 |
298 EXPECT_TRUE(pc()->AddStream(stream)); | 308 EXPECT_TRUE(pc()->AddStream(stream)); |
299 } | 309 } |
300 | 310 |
301 size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); } | 311 size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); } |
302 | 312 |
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384 data_channel_ = data_channel; | 394 data_channel_ = data_channel; |
385 data_observer_.reset(new MockDataChannelObserver(data_channel)); | 395 data_observer_.reset(new MockDataChannelObserver(data_channel)); |
386 } | 396 } |
387 | 397 |
388 void CreateDataChannel() { | 398 void CreateDataChannel() { |
389 data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, nullptr); | 399 data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, nullptr); |
390 ASSERT_TRUE(data_channel_.get() != nullptr); | 400 ASSERT_TRUE(data_channel_.get() != nullptr); |
391 data_observer_.reset(new MockDataChannelObserver(data_channel_)); | 401 data_observer_.reset(new MockDataChannelObserver(data_channel_)); |
392 } | 402 } |
393 | 403 |
394 rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack( | |
395 const std::string& stream_label) { | |
396 FakeConstraints constraints; | |
397 // Disable highpass filter so that we can get all the test audio frames. | |
398 constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false); | |
399 rtc::scoped_refptr<webrtc::AudioSourceInterface> source = | |
400 peer_connection_factory_->CreateAudioSource(&constraints); | |
401 // TODO(perkj): Test audio source when it is implemented. Currently audio | |
402 // always use the default input. | |
403 std::string label = stream_label + kAudioTrackLabelBase; | |
404 return peer_connection_factory_->CreateAudioTrack(label, source); | |
405 } | |
406 | |
407 rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack( | |
408 const std::string& stream_label) { | |
409 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky. | |
410 FakeConstraints source_constraints = video_constraints_; | |
411 source_constraints.SetMandatoryMaxFrameRate(10); | |
412 | |
413 cricket::FakeVideoCapturer* fake_capturer = | |
414 new webrtc::FakePeriodicVideoCapturer(); | |
415 video_capturers_.push_back(fake_capturer); | |
416 rtc::scoped_refptr<webrtc::VideoSourceInterface> source = | |
417 peer_connection_factory_->CreateVideoSource(fake_capturer, | |
418 &source_constraints); | |
419 std::string label = stream_label + kVideoTrackLabelBase; | |
420 return peer_connection_factory_->CreateVideoTrack(label, source); | |
421 } | |
422 | |
423 DataChannelInterface* data_channel() { return data_channel_; } | 404 DataChannelInterface* data_channel() { return data_channel_; } |
424 const MockDataChannelObserver* data_observer() const { | 405 const MockDataChannelObserver* data_observer() const { |
425 return data_observer_.get(); | 406 return data_observer_.get(); |
426 } | 407 } |
427 | 408 |
428 webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } | 409 webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } |
429 | 410 |
430 void StopVideoCapturers() { | 411 void StopVideoCapturers() { |
431 for (std::vector<cricket::VideoCapturer*>::iterator it = | 412 for (std::vector<cricket::VideoCapturer*>::iterator it = |
432 video_capturers_.begin(); | 413 video_capturers_.begin(); |
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727 return false; | 708 return false; |
728 } | 709 } |
729 if (options) { | 710 if (options) { |
730 peer_connection_factory_->SetOptions(*options); | 711 peer_connection_factory_->SetOptions(*options); |
731 } | 712 } |
732 peer_connection_ = CreatePeerConnection(allocator_factory_.get(), | 713 peer_connection_ = CreatePeerConnection(allocator_factory_.get(), |
733 constraints); | 714 constraints); |
734 return peer_connection_.get() != nullptr; | 715 return peer_connection_.get() != nullptr; |
735 } | 716 } |
736 | 717 |
| 718 rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| 719 CreateLocalVideoTrack(const std::string stream_label) { |
| 720 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky. |
| 721 FakeConstraints source_constraints = video_constraints_; |
| 722 source_constraints.SetMandatoryMaxFrameRate(10); |
| 723 |
| 724 cricket::FakeVideoCapturer* fake_capturer = |
| 725 new webrtc::FakePeriodicVideoCapturer(); |
| 726 video_capturers_.push_back(fake_capturer); |
| 727 rtc::scoped_refptr<webrtc::VideoSourceInterface> source = |
| 728 peer_connection_factory_->CreateVideoSource( |
| 729 fake_capturer, &source_constraints); |
| 730 std::string label = stream_label + kVideoTrackLabelBase; |
| 731 return peer_connection_factory_->CreateVideoTrack(label, source); |
| 732 } |
| 733 |
737 rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( | 734 rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( |
738 webrtc::PortAllocatorFactoryInterface* factory, | 735 webrtc::PortAllocatorFactoryInterface* factory, |
739 const MediaConstraintsInterface* constraints) { | 736 const MediaConstraintsInterface* constraints) { |
740 // CreatePeerConnection with IceServers. | 737 // CreatePeerConnection with IceServers. |
741 webrtc::PeerConnectionInterface::IceServers ice_servers; | 738 webrtc::PeerConnectionInterface::IceServers ice_servers; |
742 webrtc::PeerConnectionInterface::IceServer ice_server; | 739 webrtc::PeerConnectionInterface::IceServer ice_server; |
743 ice_server.uri = "stun:stun.l.google.com:19302"; | 740 ice_server.uri = "stun:stun.l.google.com:19302"; |
744 ice_servers.push_back(ice_server); | 741 ice_servers.push_back(ice_server); |
745 | 742 |
746 rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store( | 743 rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store( |
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1007 } | 1004 } |
1008 | 1005 |
1009 // This test sets up a call between two parties. Both parties send static | 1006 // This test sets up a call between two parties. Both parties send static |
1010 // frames to each other. Once the test is finished the number of sent frames | 1007 // frames to each other. Once the test is finished the number of sent frames |
1011 // is compared to the number of received frames. | 1008 // is compared to the number of received frames. |
1012 void LocalP2PTest() { | 1009 void LocalP2PTest() { |
1013 if (initiating_client_->NumberOfLocalMediaStreams() == 0) { | 1010 if (initiating_client_->NumberOfLocalMediaStreams() == 0) { |
1014 initiating_client_->AddMediaStream(true, true); | 1011 initiating_client_->AddMediaStream(true, true); |
1015 } | 1012 } |
1016 initiating_client_->Negotiate(); | 1013 initiating_client_->Negotiate(); |
| 1014 const int kMaxWaitForActivationMs = 5000; |
1017 // Assert true is used here since next tests are guaranteed to fail and | 1015 // Assert true is used here since next tests are guaranteed to fail and |
1018 // would eat up 5 seconds. | 1016 // would eat up 5 seconds. |
1019 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | 1017 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); |
1020 VerifySessionDescriptions(); | 1018 VerifySessionDescriptions(); |
1021 | 1019 |
1022 int audio_frame_count = kEndAudioFrameCount; | 1020 int audio_frame_count = kEndAudioFrameCount; |
1023 // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly. | 1021 // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly. |
1024 if (!initiating_client_->can_receive_audio() || | 1022 if (!initiating_client_->can_receive_audio() || |
1025 !receiving_client_->can_receive_audio()) { | 1023 !receiving_client_->can_receive_audio()) { |
1026 audio_frame_count = -1; | 1024 audio_frame_count = -1; |
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1648 // VideoDecoderFactory. | 1646 // VideoDecoderFactory. |
1649 // TODO(holmer): Disabled due to sometimes crashing on buildbots. | 1647 // TODO(holmer): Disabled due to sometimes crashing on buildbots. |
1650 // See issue webrtc/2378. | 1648 // See issue webrtc/2378. |
1651 TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, | 1649 TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, |
1652 DISABLED_LocalP2PTestWithVideoDecoderFactory) { | 1650 DISABLED_LocalP2PTestWithVideoDecoderFactory) { |
1653 ASSERT_TRUE(CreateTestClients()); | 1651 ASSERT_TRUE(CreateTestClients()); |
1654 EnableVideoDecoderFactory(); | 1652 EnableVideoDecoderFactory(); |
1655 LocalP2PTest(); | 1653 LocalP2PTest(); |
1656 } | 1654 } |
1657 | 1655 |
1658 // This tests that if we negotiate after calling CreateSender but before we | |
1659 // have a track, then set a track later, frames from the newly-set track are | |
1660 // received end-to-end. | |
1661 TEST_F(MAYBE_JsepPeerConnectionP2PTestClient, EarlyWarmupTest) { | |
1662 ASSERT_TRUE(CreateTestClients()); | |
1663 auto audio_sender = initializing_client()->pc()->CreateSender("audio"); | |
1664 auto video_sender = initializing_client()->pc()->CreateSender("video"); | |
1665 initializing_client()->Negotiate(); | |
1666 // Wait for ICE connection to complete, without any tracks. | |
1667 // Note that the receiving client WILL (in HandleIncomingOffer) create | |
1668 // tracks, so it's only the initiator here that's doing early warmup. | |
1669 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | |
1670 VerifySessionDescriptions(); | |
1671 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
1672 initializing_client()->ice_connection_state(), | |
1673 kMaxWaitForFramesMs); | |
1674 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
1675 receiving_client()->ice_connection_state(), | |
1676 kMaxWaitForFramesMs); | |
1677 // Now set the tracks, and expect frames to immediately start flowing. | |
1678 EXPECT_TRUE( | |
1679 audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack(""))); | |
1680 EXPECT_TRUE( | |
1681 video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack(""))); | |
1682 EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount), | |
1683 kMaxWaitForFramesMs); | |
1684 } | |
1685 | |
1686 class IceServerParsingTest : public testing::Test { | 1656 class IceServerParsingTest : public testing::Test { |
1687 public: | 1657 public: |
1688 // Convenience for parsing a single URL. | 1658 // Convenience for parsing a single URL. |
1689 bool ParseUrl(const std::string& url) { | 1659 bool ParseUrl(const std::string& url) { |
1690 return ParseUrl(url, std::string(), std::string()); | 1660 return ParseUrl(url, std::string(), std::string()); |
1691 } | 1661 } |
1692 | 1662 |
1693 bool ParseUrl(const std::string& url, | 1663 bool ParseUrl(const std::string& url, |
1694 const std::string& username, | 1664 const std::string& username, |
1695 const std::string& password) { | 1665 const std::string& password) { |
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1853 server.urls.push_back("stun:hostname"); | 1823 server.urls.push_back("stun:hostname"); |
1854 server.urls.push_back("turn:hostname"); | 1824 server.urls.push_back("turn:hostname"); |
1855 servers.push_back(server); | 1825 servers.push_back(server); |
1856 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_configurations_, | 1826 EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_configurations_, |
1857 &turn_configurations_)); | 1827 &turn_configurations_)); |
1858 EXPECT_EQ(1U, stun_configurations_.size()); | 1828 EXPECT_EQ(1U, stun_configurations_.size()); |
1859 EXPECT_EQ(1U, turn_configurations_.size()); | 1829 EXPECT_EQ(1U, turn_configurations_.size()); |
1860 } | 1830 } |
1861 | 1831 |
1862 #endif // if !defined(THREAD_SANITIZER) | 1832 #endif // if !defined(THREAD_SANITIZER) |
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