deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 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 |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | // Disable for TSan v2, see |
| 12 | // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. |
| 13 | #if !defined(THREAD_SANITIZER) |
| 14 | |
| 15 | #include <stdio.h> |
| 16 | |
| 17 | #include <algorithm> |
| 18 | #include <functional> |
| 19 | #include <list> |
| 20 | #include <map> |
| 21 | #include <memory> |
| 22 | #include <utility> |
| 23 | #include <vector> |
| 24 | |
Karl Wiberg | 1b0eae3 | 2017-10-17 14:48:54 +0200 | [diff] [blame] | 25 | #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| 26 | #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 27 | #include "api/fakemetricsobserver.h" |
| 28 | #include "api/mediastreaminterface.h" |
| 29 | #include "api/peerconnectioninterface.h" |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 30 | #include "api/peerconnectionproxy.h" |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 31 | #include "api/rtpreceiverinterface.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 32 | #include "api/test/fakeconstraints.h" |
| 33 | #include "media/engine/fakewebrtcvideoengine.h" |
| 34 | #include "p2p/base/p2pconstants.h" |
| 35 | #include "p2p/base/portinterface.h" |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 36 | #include "p2p/base/teststunserver.h" |
Jonas Oreland | bdcee28 | 2017-10-10 14:01:40 +0200 | [diff] [blame] | 37 | #include "p2p/base/testturncustomizer.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 38 | #include "p2p/base/testturnserver.h" |
| 39 | #include "p2p/client/basicportallocator.h" |
| 40 | #include "pc/dtmfsender.h" |
| 41 | #include "pc/localaudiosource.h" |
| 42 | #include "pc/mediasession.h" |
| 43 | #include "pc/peerconnection.h" |
| 44 | #include "pc/peerconnectionfactory.h" |
Steve Anton | 4ab68ee | 2017-12-19 14:26:11 -0800 | [diff] [blame] | 45 | #include "pc/sessiondescription.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 46 | #include "pc/test/fakeaudiocapturemodule.h" |
| 47 | #include "pc/test/fakeperiodicvideocapturer.h" |
| 48 | #include "pc/test/fakertccertificategenerator.h" |
| 49 | #include "pc/test/fakevideotrackrenderer.h" |
| 50 | #include "pc/test/mockpeerconnectionobservers.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 51 | #include "rtc_base/fakenetwork.h" |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 52 | #include "rtc_base/firewallsocketserver.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 53 | #include "rtc_base/gunit.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 54 | #include "rtc_base/virtualsocketserver.h" |
Elad Alon | 99c3fe5 | 2017-10-13 16:29:40 +0200 | [diff] [blame] | 55 | #include "test/gmock.h" |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 56 | |
| 57 | using cricket::ContentInfo; |
| 58 | using cricket::FakeWebRtcVideoDecoder; |
| 59 | using cricket::FakeWebRtcVideoDecoderFactory; |
| 60 | using cricket::FakeWebRtcVideoEncoder; |
| 61 | using cricket::FakeWebRtcVideoEncoderFactory; |
| 62 | using cricket::MediaContentDescription; |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 63 | using rtc::SocketAddress; |
| 64 | using ::testing::ElementsAre; |
| 65 | using ::testing::Values; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 66 | using webrtc::DataBuffer; |
| 67 | using webrtc::DataChannelInterface; |
| 68 | using webrtc::DtmfSender; |
| 69 | using webrtc::DtmfSenderInterface; |
| 70 | using webrtc::DtmfSenderObserverInterface; |
| 71 | using webrtc::FakeConstraints; |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 72 | using webrtc::FakeVideoTrackRenderer; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 73 | using webrtc::MediaConstraintsInterface; |
| 74 | using webrtc::MediaStreamInterface; |
| 75 | using webrtc::MediaStreamTrackInterface; |
| 76 | using webrtc::MockCreateSessionDescriptionObserver; |
| 77 | using webrtc::MockDataChannelObserver; |
| 78 | using webrtc::MockSetSessionDescriptionObserver; |
| 79 | using webrtc::MockStatsObserver; |
| 80 | using webrtc::ObserverInterface; |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 81 | using webrtc::PeerConnection; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 82 | using webrtc::PeerConnectionInterface; |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 83 | using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 84 | using webrtc::PeerConnectionFactory; |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 85 | using webrtc::PeerConnectionProxy; |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 86 | using webrtc::RTCErrorType; |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 87 | using webrtc::RtpSenderInterface; |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 88 | using webrtc::RtpReceiverInterface; |
Steve Anton | d367921 | 2018-01-17 17:41:02 -0800 | [diff] [blame] | 89 | using webrtc::SdpSemantics; |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 90 | using webrtc::SdpType; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 91 | using webrtc::SessionDescriptionInterface; |
| 92 | using webrtc::StreamCollectionInterface; |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 93 | using webrtc::VideoTrackInterface; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 94 | |
| 95 | namespace { |
| 96 | |
| 97 | static const int kDefaultTimeout = 10000; |
| 98 | static const int kMaxWaitForStatsMs = 3000; |
| 99 | static const int kMaxWaitForActivationMs = 5000; |
| 100 | static const int kMaxWaitForFramesMs = 10000; |
| 101 | // Default number of audio/video frames to wait for before considering a test |
| 102 | // successful. |
| 103 | static const int kDefaultExpectedAudioFrameCount = 3; |
| 104 | static const int kDefaultExpectedVideoFrameCount = 3; |
| 105 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 106 | static const char kDataChannelLabel[] = "data_channel"; |
| 107 | |
| 108 | // SRTP cipher name negotiated by the tests. This must be updated if the |
| 109 | // default changes. |
Tommi | 8e545ee | 2018-02-08 16:25:20 +0000 | [diff] [blame] | 110 | static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 111 | static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM; |
| 112 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 113 | static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0); |
| 114 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 115 | // Helper function for constructing offer/answer options to initiate an ICE |
| 116 | // restart. |
| 117 | PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() { |
| 118 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 119 | options.ice_restart = true; |
| 120 | return options; |
| 121 | } |
| 122 | |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 123 | // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic" |
| 124 | // attribute from received SDP, simulating a legacy endpoint. |
| 125 | void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) { |
| 126 | for (ContentInfo& content : desc->contents()) { |
Steve Anton | b1c1de1 | 2017-12-21 15:14:30 -0800 | [diff] [blame] | 127 | content.media_description()->mutable_streams().clear(); |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 128 | } |
| 129 | desc->set_msid_supported(false); |
| 130 | } |
| 131 | |
zhihuang | f816493 | 2017-05-19 13:09:47 -0700 | [diff] [blame] | 132 | int FindFirstMediaStatsIndexByKind( |
| 133 | const std::string& kind, |
| 134 | const std::vector<const webrtc::RTCMediaStreamTrackStats*>& |
| 135 | media_stats_vec) { |
| 136 | for (size_t i = 0; i < media_stats_vec.size(); i++) { |
| 137 | if (media_stats_vec[i]->kind.ValueToString() == kind) { |
| 138 | return i; |
| 139 | } |
| 140 | } |
| 141 | return -1; |
| 142 | } |
| 143 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 144 | class SignalingMessageReceiver { |
| 145 | public: |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 146 | virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 147 | virtual void ReceiveIceMessage(const std::string& sdp_mid, |
| 148 | int sdp_mline_index, |
| 149 | const std::string& msg) = 0; |
| 150 | |
| 151 | protected: |
| 152 | SignalingMessageReceiver() {} |
| 153 | virtual ~SignalingMessageReceiver() {} |
| 154 | }; |
| 155 | |
| 156 | class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { |
| 157 | public: |
| 158 | explicit MockRtpReceiverObserver(cricket::MediaType media_type) |
| 159 | : expected_media_type_(media_type) {} |
| 160 | |
| 161 | void OnFirstPacketReceived(cricket::MediaType media_type) override { |
| 162 | ASSERT_EQ(expected_media_type_, media_type); |
| 163 | first_packet_received_ = true; |
| 164 | } |
| 165 | |
| 166 | bool first_packet_received() const { return first_packet_received_; } |
| 167 | |
| 168 | virtual ~MockRtpReceiverObserver() {} |
| 169 | |
| 170 | private: |
| 171 | bool first_packet_received_ = false; |
| 172 | cricket::MediaType expected_media_type_; |
| 173 | }; |
| 174 | |
| 175 | // Helper class that wraps a peer connection, observes it, and can accept |
| 176 | // signaling messages from another wrapper. |
| 177 | // |
| 178 | // Uses a fake network, fake A/V capture, and optionally fake |
| 179 | // encoders/decoders, though they aren't used by default since they don't |
| 180 | // advertise support of any codecs. |
Steve Anton | 94286cb | 2017-09-26 16:20:19 -0700 | [diff] [blame] | 181 | // TODO(steveanton): See how this could become a subclass of |
| 182 | // PeerConnectionWrapper defined in peerconnectionwrapper.h . |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 183 | class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 184 | public SignalingMessageReceiver { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 185 | public: |
| 186 | // Different factory methods for convenience. |
| 187 | // TODO(deadbeef): Could use the pattern of: |
| 188 | // |
| 189 | // PeerConnectionWrapper = |
| 190 | // WrapperBuilder.WithConfig(...).WithOptions(...).build(); |
| 191 | // |
| 192 | // To reduce some code duplication. |
| 193 | static PeerConnectionWrapper* CreateWithDtlsIdentityStore( |
| 194 | const std::string& debug_name, |
| 195 | std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
| 196 | rtc::Thread* network_thread, |
| 197 | rtc::Thread* worker_thread) { |
| 198 | PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); |
| 199 | if (!client->Init(nullptr, nullptr, nullptr, std::move(cert_generator), |
| 200 | network_thread, worker_thread)) { |
| 201 | delete client; |
| 202 | return nullptr; |
| 203 | } |
| 204 | return client; |
| 205 | } |
| 206 | |
| 207 | static PeerConnectionWrapper* CreateWithConfig( |
| 208 | const std::string& debug_name, |
| 209 | const PeerConnectionInterface::RTCConfiguration& config, |
| 210 | rtc::Thread* network_thread, |
| 211 | rtc::Thread* worker_thread) { |
| 212 | std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| 213 | new FakeRTCCertificateGenerator()); |
| 214 | PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); |
| 215 | if (!client->Init(nullptr, nullptr, &config, std::move(cert_generator), |
| 216 | network_thread, worker_thread)) { |
| 217 | delete client; |
| 218 | return nullptr; |
| 219 | } |
| 220 | return client; |
| 221 | } |
| 222 | |
| 223 | static PeerConnectionWrapper* CreateWithOptions( |
| 224 | const std::string& debug_name, |
| 225 | const PeerConnectionFactory::Options& options, |
| 226 | rtc::Thread* network_thread, |
| 227 | rtc::Thread* worker_thread) { |
| 228 | std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| 229 | new FakeRTCCertificateGenerator()); |
| 230 | PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); |
| 231 | if (!client->Init(nullptr, &options, nullptr, std::move(cert_generator), |
| 232 | network_thread, worker_thread)) { |
| 233 | delete client; |
| 234 | return nullptr; |
| 235 | } |
| 236 | return client; |
| 237 | } |
| 238 | |
| 239 | static PeerConnectionWrapper* CreateWithConstraints( |
| 240 | const std::string& debug_name, |
| 241 | const MediaConstraintsInterface* constraints, |
| 242 | rtc::Thread* network_thread, |
| 243 | rtc::Thread* worker_thread) { |
| 244 | std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| 245 | new FakeRTCCertificateGenerator()); |
| 246 | PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); |
| 247 | if (!client->Init(constraints, nullptr, nullptr, std::move(cert_generator), |
| 248 | network_thread, worker_thread)) { |
| 249 | delete client; |
| 250 | return nullptr; |
| 251 | } |
| 252 | return client; |
| 253 | } |
| 254 | |
deadbeef | 2f425aa | 2017-04-14 10:41:32 -0700 | [diff] [blame] | 255 | webrtc::PeerConnectionFactoryInterface* pc_factory() const { |
| 256 | return peer_connection_factory_.get(); |
| 257 | } |
| 258 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 259 | webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); } |
| 260 | |
| 261 | // If a signaling message receiver is set (via ConnectFakeSignaling), this |
| 262 | // will set the whole offer/answer exchange in motion. Just need to wait for |
| 263 | // the signaling state to reach "stable". |
| 264 | void CreateAndSetAndSignalOffer() { |
| 265 | auto offer = CreateOffer(); |
| 266 | ASSERT_NE(nullptr, offer); |
| 267 | EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer))); |
| 268 | } |
| 269 | |
| 270 | // Sets the options to be used when CreateAndSetAndSignalOffer is called, or |
| 271 | // when a remote offer is received (via fake signaling) and an answer is |
| 272 | // generated. By default, uses default options. |
| 273 | void SetOfferAnswerOptions( |
| 274 | const PeerConnectionInterface::RTCOfferAnswerOptions& options) { |
| 275 | offer_answer_options_ = options; |
| 276 | } |
| 277 | |
| 278 | // Set a callback to be invoked when SDP is received via the fake signaling |
| 279 | // channel, which provides an opportunity to munge (modify) the SDP. This is |
| 280 | // used to test SDP being applied that a PeerConnection would normally not |
| 281 | // generate, but a non-JSEP endpoint might. |
| 282 | void SetReceivedSdpMunger( |
| 283 | std::function<void(cricket::SessionDescription*)> munger) { |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 284 | received_sdp_munger_ = std::move(munger); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 285 | } |
| 286 | |
deadbeef | c964d0b | 2017-04-03 10:03:35 -0700 | [diff] [blame] | 287 | // Similar to the above, but this is run on SDP immediately after it's |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 288 | // generated. |
| 289 | void SetGeneratedSdpMunger( |
| 290 | std::function<void(cricket::SessionDescription*)> munger) { |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 291 | generated_sdp_munger_ = std::move(munger); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 292 | } |
| 293 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 294 | // Every ICE connection state in order that has been seen by the observer. |
| 295 | std::vector<PeerConnectionInterface::IceConnectionState> |
| 296 | ice_connection_state_history() const { |
| 297 | return ice_connection_state_history_; |
| 298 | } |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 299 | void clear_ice_connection_state_history() { |
| 300 | ice_connection_state_history_.clear(); |
| 301 | } |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 302 | |
| 303 | // Every ICE gathering state in order that has been seen by the observer. |
| 304 | std::vector<PeerConnectionInterface::IceGatheringState> |
| 305 | ice_gathering_state_history() const { |
| 306 | return ice_gathering_state_history_; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 307 | } |
| 308 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 309 | void AddAudioVideoTracks() { |
| 310 | AddAudioTrack(); |
| 311 | AddVideoTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 312 | } |
| 313 | |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 314 | rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() { |
| 315 | return AddTrack(CreateLocalAudioTrack()); |
| 316 | } |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 317 | |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 318 | rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() { |
| 319 | return AddTrack(CreateLocalVideoTrack()); |
| 320 | } |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 321 | |
| 322 | rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() { |
| 323 | FakeConstraints constraints; |
| 324 | // Disable highpass filter so that we can get all the test audio frames. |
| 325 | constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false); |
| 326 | rtc::scoped_refptr<webrtc::AudioSourceInterface> source = |
| 327 | peer_connection_factory_->CreateAudioSource(&constraints); |
| 328 | // TODO(perkj): Test audio source when it is implemented. Currently audio |
| 329 | // always use the default input. |
deadbeef | b1a15d7 | 2017-09-07 14:12:05 -0700 | [diff] [blame] | 330 | return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(), |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 331 | source); |
| 332 | } |
| 333 | |
| 334 | rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() { |
deadbeef | b1a15d7 | 2017-09-07 14:12:05 -0700 | [diff] [blame] | 335 | return CreateLocalVideoTrackInternal(FakeConstraints(), |
| 336 | webrtc::kVideoRotation_0); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 337 | } |
| 338 | |
| 339 | rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| 340 | CreateLocalVideoTrackWithConstraints(const FakeConstraints& constraints) { |
deadbeef | b1a15d7 | 2017-09-07 14:12:05 -0700 | [diff] [blame] | 341 | return CreateLocalVideoTrackInternal(constraints, webrtc::kVideoRotation_0); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 342 | } |
| 343 | |
| 344 | rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| 345 | CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) { |
deadbeef | b1a15d7 | 2017-09-07 14:12:05 -0700 | [diff] [blame] | 346 | return CreateLocalVideoTrackInternal(FakeConstraints(), rotation); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 347 | } |
| 348 | |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 349 | rtc::scoped_refptr<RtpSenderInterface> AddTrack( |
| 350 | rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| 351 | const std::vector<std::string>& stream_labels = {}) { |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 352 | auto result = pc()->AddTrack(track, stream_labels); |
| 353 | EXPECT_EQ(RTCErrorType::NONE, result.error().type()); |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 354 | return result.MoveValue(); |
| 355 | } |
| 356 | |
| 357 | std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType( |
| 358 | cricket::MediaType media_type) { |
| 359 | std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers; |
| 360 | for (auto receiver : pc()->GetReceivers()) { |
| 361 | if (receiver->media_type() == media_type) { |
| 362 | receivers.push_back(receiver); |
| 363 | } |
| 364 | } |
| 365 | return receivers; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 366 | } |
| 367 | |
| 368 | bool SignalingStateStable() { |
| 369 | return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable; |
| 370 | } |
| 371 | |
| 372 | void CreateDataChannel() { CreateDataChannel(nullptr); } |
| 373 | |
| 374 | void CreateDataChannel(const webrtc::DataChannelInit* init) { |
Steve Anton | da6c095 | 2017-10-23 11:41:54 -0700 | [diff] [blame] | 375 | CreateDataChannel(kDataChannelLabel, init); |
| 376 | } |
| 377 | |
| 378 | void CreateDataChannel(const std::string& label, |
| 379 | const webrtc::DataChannelInit* init) { |
| 380 | data_channel_ = pc()->CreateDataChannel(label, init); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 381 | ASSERT_TRUE(data_channel_.get() != nullptr); |
| 382 | data_observer_.reset(new MockDataChannelObserver(data_channel_)); |
| 383 | } |
| 384 | |
| 385 | DataChannelInterface* data_channel() { return data_channel_; } |
| 386 | const MockDataChannelObserver* data_observer() const { |
| 387 | return data_observer_.get(); |
| 388 | } |
| 389 | |
| 390 | int audio_frames_received() const { |
| 391 | return fake_audio_capture_module_->frames_received(); |
| 392 | } |
| 393 | |
| 394 | // Takes minimum of video frames received for each track. |
| 395 | // |
| 396 | // Can be used like: |
| 397 | // EXPECT_GE(expected_frames, min_video_frames_received_per_track()); |
| 398 | // |
| 399 | // To ensure that all video tracks received at least a certain number of |
| 400 | // frames. |
| 401 | int min_video_frames_received_per_track() const { |
| 402 | int min_frames = INT_MAX; |
| 403 | if (video_decoder_factory_enabled_) { |
| 404 | const std::vector<FakeWebRtcVideoDecoder*>& decoders = |
| 405 | fake_video_decoder_factory_->decoders(); |
| 406 | if (decoders.empty()) { |
| 407 | return 0; |
| 408 | } |
| 409 | for (FakeWebRtcVideoDecoder* decoder : decoders) { |
| 410 | min_frames = std::min(min_frames, decoder->GetNumFramesReceived()); |
| 411 | } |
| 412 | return min_frames; |
| 413 | } else { |
| 414 | if (fake_video_renderers_.empty()) { |
| 415 | return 0; |
| 416 | } |
| 417 | |
| 418 | for (const auto& pair : fake_video_renderers_) { |
| 419 | min_frames = std::min(min_frames, pair.second->num_rendered_frames()); |
| 420 | } |
| 421 | return min_frames; |
| 422 | } |
| 423 | } |
| 424 | |
| 425 | // In contrast to the above, sums the video frames received for all tracks. |
| 426 | // Can be used to verify that no video frames were received, or that the |
| 427 | // counts didn't increase. |
| 428 | int total_video_frames_received() const { |
| 429 | int total = 0; |
| 430 | if (video_decoder_factory_enabled_) { |
| 431 | const std::vector<FakeWebRtcVideoDecoder*>& decoders = |
| 432 | fake_video_decoder_factory_->decoders(); |
| 433 | for (const FakeWebRtcVideoDecoder* decoder : decoders) { |
| 434 | total += decoder->GetNumFramesReceived(); |
| 435 | } |
| 436 | } else { |
| 437 | for (const auto& pair : fake_video_renderers_) { |
| 438 | total += pair.second->num_rendered_frames(); |
| 439 | } |
| 440 | for (const auto& renderer : removed_fake_video_renderers_) { |
| 441 | total += renderer->num_rendered_frames(); |
| 442 | } |
| 443 | } |
| 444 | return total; |
| 445 | } |
| 446 | |
| 447 | // Returns a MockStatsObserver in a state after stats gathering finished, |
| 448 | // which can be used to access the gathered stats. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 449 | rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 450 | webrtc::MediaStreamTrackInterface* track) { |
| 451 | rtc::scoped_refptr<MockStatsObserver> observer( |
| 452 | new rtc::RefCountedObject<MockStatsObserver>()); |
| 453 | EXPECT_TRUE(peer_connection_->GetStats( |
| 454 | observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); |
| 455 | EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| 456 | return observer; |
| 457 | } |
| 458 | |
| 459 | // Version that doesn't take a track "filter", and gathers all stats. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 460 | rtc::scoped_refptr<MockStatsObserver> OldGetStats() { |
| 461 | return OldGetStatsForTrack(nullptr); |
| 462 | } |
| 463 | |
| 464 | // Synchronously gets stats and returns them. If it times out, fails the test |
| 465 | // and returns null. |
| 466 | rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() { |
| 467 | rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback( |
| 468 | new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>()); |
| 469 | peer_connection_->GetStats(callback); |
| 470 | EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout); |
| 471 | return callback->report(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 472 | } |
| 473 | |
| 474 | int rendered_width() { |
| 475 | EXPECT_FALSE(fake_video_renderers_.empty()); |
| 476 | return fake_video_renderers_.empty() |
| 477 | ? 0 |
| 478 | : fake_video_renderers_.begin()->second->width(); |
| 479 | } |
| 480 | |
| 481 | int rendered_height() { |
| 482 | EXPECT_FALSE(fake_video_renderers_.empty()); |
| 483 | return fake_video_renderers_.empty() |
| 484 | ? 0 |
| 485 | : fake_video_renderers_.begin()->second->height(); |
| 486 | } |
| 487 | |
| 488 | double rendered_aspect_ratio() { |
| 489 | if (rendered_height() == 0) { |
| 490 | return 0.0; |
| 491 | } |
| 492 | return static_cast<double>(rendered_width()) / rendered_height(); |
| 493 | } |
| 494 | |
| 495 | webrtc::VideoRotation rendered_rotation() { |
| 496 | EXPECT_FALSE(fake_video_renderers_.empty()); |
| 497 | return fake_video_renderers_.empty() |
| 498 | ? webrtc::kVideoRotation_0 |
| 499 | : fake_video_renderers_.begin()->second->rotation(); |
| 500 | } |
| 501 | |
| 502 | int local_rendered_width() { |
| 503 | return local_video_renderer_ ? local_video_renderer_->width() : 0; |
| 504 | } |
| 505 | |
| 506 | int local_rendered_height() { |
| 507 | return local_video_renderer_ ? local_video_renderer_->height() : 0; |
| 508 | } |
| 509 | |
| 510 | double local_rendered_aspect_ratio() { |
| 511 | if (local_rendered_height() == 0) { |
| 512 | return 0.0; |
| 513 | } |
| 514 | return static_cast<double>(local_rendered_width()) / |
| 515 | local_rendered_height(); |
| 516 | } |
| 517 | |
| 518 | size_t number_of_remote_streams() { |
| 519 | if (!pc()) { |
| 520 | return 0; |
| 521 | } |
| 522 | return pc()->remote_streams()->count(); |
| 523 | } |
| 524 | |
| 525 | StreamCollectionInterface* remote_streams() const { |
| 526 | if (!pc()) { |
| 527 | ADD_FAILURE(); |
| 528 | return nullptr; |
| 529 | } |
| 530 | return pc()->remote_streams(); |
| 531 | } |
| 532 | |
| 533 | StreamCollectionInterface* local_streams() { |
| 534 | if (!pc()) { |
| 535 | ADD_FAILURE(); |
| 536 | return nullptr; |
| 537 | } |
| 538 | return pc()->local_streams(); |
| 539 | } |
| 540 | |
| 541 | webrtc::PeerConnectionInterface::SignalingState signaling_state() { |
| 542 | return pc()->signaling_state(); |
| 543 | } |
| 544 | |
| 545 | webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() { |
| 546 | return pc()->ice_connection_state(); |
| 547 | } |
| 548 | |
| 549 | webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { |
| 550 | return pc()->ice_gathering_state(); |
| 551 | } |
| 552 | |
| 553 | // Returns a MockRtpReceiverObserver for each RtpReceiver returned by |
| 554 | // GetReceivers. They're updated automatically when a remote offer/answer |
| 555 | // from the fake signaling channel is applied, or when |
| 556 | // ResetRtpReceiverObservers below is called. |
| 557 | const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& |
| 558 | rtp_receiver_observers() { |
| 559 | return rtp_receiver_observers_; |
| 560 | } |
| 561 | |
| 562 | void ResetRtpReceiverObservers() { |
| 563 | rtp_receiver_observers_.clear(); |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 564 | for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver : |
| 565 | pc()->GetReceivers()) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 566 | std::unique_ptr<MockRtpReceiverObserver> observer( |
| 567 | new MockRtpReceiverObserver(receiver->media_type())); |
| 568 | receiver->SetObserver(observer.get()); |
| 569 | rtp_receiver_observers_.push_back(std::move(observer)); |
| 570 | } |
| 571 | } |
| 572 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 573 | rtc::FakeNetworkManager* network() const { |
| 574 | return fake_network_manager_.get(); |
| 575 | } |
| 576 | cricket::PortAllocator* port_allocator() const { return port_allocator_; } |
| 577 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 578 | private: |
| 579 | explicit PeerConnectionWrapper(const std::string& debug_name) |
| 580 | : debug_name_(debug_name) {} |
| 581 | |
| 582 | bool Init( |
| 583 | const MediaConstraintsInterface* constraints, |
| 584 | const PeerConnectionFactory::Options* options, |
| 585 | const PeerConnectionInterface::RTCConfiguration* config, |
| 586 | std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
| 587 | rtc::Thread* network_thread, |
| 588 | rtc::Thread* worker_thread) { |
| 589 | // There's an error in this test code if Init ends up being called twice. |
| 590 | RTC_DCHECK(!peer_connection_); |
| 591 | RTC_DCHECK(!peer_connection_factory_); |
| 592 | |
| 593 | fake_network_manager_.reset(new rtc::FakeNetworkManager()); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 594 | fake_network_manager_->AddInterface(kDefaultLocalAddress); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 595 | |
| 596 | std::unique_ptr<cricket::PortAllocator> port_allocator( |
| 597 | new cricket::BasicPortAllocator(fake_network_manager_.get())); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 598 | port_allocator_ = port_allocator.get(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 599 | fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); |
| 600 | if (!fake_audio_capture_module_) { |
| 601 | return false; |
| 602 | } |
| 603 | // Note that these factories don't end up getting used unless supported |
| 604 | // codecs are added to them. |
| 605 | fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory(); |
| 606 | fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory(); |
| 607 | rtc::Thread* const signaling_thread = rtc::Thread::Current(); |
| 608 | peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( |
| 609 | network_thread, worker_thread, signaling_thread, |
Karl Wiberg | 1b0eae3 | 2017-10-17 14:48:54 +0200 | [diff] [blame] | 610 | fake_audio_capture_module_, webrtc::CreateBuiltinAudioEncoderFactory(), |
| 611 | webrtc::CreateBuiltinAudioDecoderFactory(), fake_video_encoder_factory_, |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 612 | fake_video_decoder_factory_); |
| 613 | if (!peer_connection_factory_) { |
| 614 | return false; |
| 615 | } |
| 616 | if (options) { |
| 617 | peer_connection_factory_->SetOptions(*options); |
| 618 | } |
| 619 | peer_connection_ = |
| 620 | CreatePeerConnection(std::move(port_allocator), constraints, config, |
| 621 | std::move(cert_generator)); |
| 622 | return peer_connection_.get() != nullptr; |
| 623 | } |
| 624 | |
| 625 | rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( |
| 626 | std::unique_ptr<cricket::PortAllocator> port_allocator, |
| 627 | const MediaConstraintsInterface* constraints, |
| 628 | const PeerConnectionInterface::RTCConfiguration* config, |
| 629 | std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) { |
| 630 | PeerConnectionInterface::RTCConfiguration modified_config; |
| 631 | // If |config| is null, this will result in a default configuration being |
| 632 | // used. |
| 633 | if (config) { |
| 634 | modified_config = *config; |
| 635 | } |
| 636 | // Disable resolution adaptation; we don't want it interfering with the |
| 637 | // test results. |
| 638 | // TODO(deadbeef): Do something more robust. Since we're testing for aspect |
| 639 | // ratios and not specific resolutions, is this even necessary? |
| 640 | modified_config.set_cpu_adaptation(false); |
| 641 | |
| 642 | return peer_connection_factory_->CreatePeerConnection( |
| 643 | modified_config, constraints, std::move(port_allocator), |
| 644 | std::move(cert_generator), this); |
| 645 | } |
| 646 | |
| 647 | void set_signaling_message_receiver( |
| 648 | SignalingMessageReceiver* signaling_message_receiver) { |
| 649 | signaling_message_receiver_ = signaling_message_receiver; |
| 650 | } |
| 651 | |
| 652 | void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; } |
| 653 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 654 | void set_signal_ice_candidates(bool signal) { |
| 655 | signal_ice_candidates_ = signal; |
| 656 | } |
| 657 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 658 | void EnableVideoDecoderFactory() { |
| 659 | video_decoder_factory_enabled_ = true; |
| 660 | fake_video_decoder_factory_->AddSupportedVideoCodecType( |
| 661 | webrtc::kVideoCodecVP8); |
| 662 | } |
| 663 | |
| 664 | rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 665 | const FakeConstraints& constraints, |
| 666 | webrtc::VideoRotation rotation) { |
| 667 | // Set max frame rate to 10fps to reduce the risk of test flakiness. |
| 668 | // TODO(deadbeef): Do something more robust. |
| 669 | FakeConstraints source_constraints = constraints; |
| 670 | source_constraints.SetMandatoryMaxFrameRate(10); |
| 671 | |
| 672 | cricket::FakeVideoCapturer* fake_capturer = |
| 673 | new webrtc::FakePeriodicVideoCapturer(); |
| 674 | fake_capturer->SetRotation(rotation); |
| 675 | video_capturers_.push_back(fake_capturer); |
| 676 | rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source = |
| 677 | peer_connection_factory_->CreateVideoSource(fake_capturer, |
| 678 | &source_constraints); |
| 679 | rtc::scoped_refptr<webrtc::VideoTrackInterface> track( |
deadbeef | b1a15d7 | 2017-09-07 14:12:05 -0700 | [diff] [blame] | 680 | peer_connection_factory_->CreateVideoTrack(rtc::CreateRandomUuid(), |
| 681 | source)); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 682 | if (!local_video_renderer_) { |
| 683 | local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track)); |
| 684 | } |
| 685 | return track; |
| 686 | } |
| 687 | |
| 688 | void HandleIncomingOffer(const std::string& msg) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 689 | RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer"; |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 690 | std::unique_ptr<SessionDescriptionInterface> desc = |
| 691 | webrtc::CreateSessionDescription(SdpType::kOffer, msg); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 692 | if (received_sdp_munger_) { |
| 693 | received_sdp_munger_(desc->description()); |
| 694 | } |
| 695 | |
| 696 | EXPECT_TRUE(SetRemoteDescription(std::move(desc))); |
| 697 | // Setting a remote description may have changed the number of receivers, |
| 698 | // so reset the receiver observers. |
| 699 | ResetRtpReceiverObservers(); |
| 700 | auto answer = CreateAnswer(); |
| 701 | ASSERT_NE(nullptr, answer); |
| 702 | EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer))); |
| 703 | } |
| 704 | |
| 705 | void HandleIncomingAnswer(const std::string& msg) { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 706 | RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer"; |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 707 | std::unique_ptr<SessionDescriptionInterface> desc = |
| 708 | webrtc::CreateSessionDescription(SdpType::kAnswer, msg); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 709 | if (received_sdp_munger_) { |
| 710 | received_sdp_munger_(desc->description()); |
| 711 | } |
| 712 | |
| 713 | EXPECT_TRUE(SetRemoteDescription(std::move(desc))); |
| 714 | // Set the RtpReceiverObserver after receivers are created. |
| 715 | ResetRtpReceiverObservers(); |
| 716 | } |
| 717 | |
| 718 | // Returns null on failure. |
| 719 | std::unique_ptr<SessionDescriptionInterface> CreateOffer() { |
| 720 | rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| 721 | new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| 722 | pc()->CreateOffer(observer, offer_answer_options_); |
| 723 | return WaitForDescriptionFromObserver(observer); |
| 724 | } |
| 725 | |
| 726 | // Returns null on failure. |
| 727 | std::unique_ptr<SessionDescriptionInterface> CreateAnswer() { |
| 728 | rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| 729 | new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| 730 | pc()->CreateAnswer(observer, offer_answer_options_); |
| 731 | return WaitForDescriptionFromObserver(observer); |
| 732 | } |
| 733 | |
| 734 | std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver( |
Mirko Bonadei | c61ce0d | 2017-11-21 17:04:20 +0100 | [diff] [blame] | 735 | MockCreateSessionDescriptionObserver* observer) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 736 | EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout); |
| 737 | if (!observer->result()) { |
| 738 | return nullptr; |
| 739 | } |
| 740 | auto description = observer->MoveDescription(); |
| 741 | if (generated_sdp_munger_) { |
| 742 | generated_sdp_munger_(description->description()); |
| 743 | } |
| 744 | return description; |
| 745 | } |
| 746 | |
| 747 | // Setting the local description and sending the SDP message over the fake |
| 748 | // signaling channel are combined into the same method because the SDP |
| 749 | // message needs to be sent as soon as SetLocalDescription finishes, without |
| 750 | // waiting for the observer to be called. This ensures that ICE candidates |
| 751 | // don't outrace the description. |
| 752 | bool SetLocalDescriptionAndSendSdpMessage( |
| 753 | std::unique_ptr<SessionDescriptionInterface> desc) { |
| 754 | rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( |
| 755 | new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 756 | RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage"; |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 757 | SdpType type = desc->GetType(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 758 | std::string sdp; |
| 759 | EXPECT_TRUE(desc->ToString(&sdp)); |
| 760 | pc()->SetLocalDescription(observer, desc.release()); |
| 761 | // As mentioned above, we need to send the message immediately after |
| 762 | // SetLocalDescription. |
| 763 | SendSdpMessage(type, sdp); |
| 764 | EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| 765 | return true; |
| 766 | } |
| 767 | |
| 768 | bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) { |
| 769 | rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( |
| 770 | new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 771 | RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription"; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 772 | pc()->SetRemoteDescription(observer, desc.release()); |
| 773 | EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| 774 | return observer->result(); |
| 775 | } |
| 776 | |
| 777 | // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by |
| 778 | // default). |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 779 | void SendSdpMessage(SdpType type, const std::string& msg) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 780 | if (signaling_delay_ms_ == 0) { |
| 781 | RelaySdpMessageIfReceiverExists(type, msg); |
| 782 | } else { |
| 783 | invoker_.AsyncInvokeDelayed<void>( |
| 784 | RTC_FROM_HERE, rtc::Thread::Current(), |
| 785 | rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists, |
| 786 | this, type, msg), |
| 787 | signaling_delay_ms_); |
| 788 | } |
| 789 | } |
| 790 | |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 791 | void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 792 | if (signaling_message_receiver_) { |
| 793 | signaling_message_receiver_->ReceiveSdpMessage(type, msg); |
| 794 | } |
| 795 | } |
| 796 | |
| 797 | // Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by |
| 798 | // default). |
| 799 | void SendIceMessage(const std::string& sdp_mid, |
| 800 | int sdp_mline_index, |
| 801 | const std::string& msg) { |
| 802 | if (signaling_delay_ms_ == 0) { |
| 803 | RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg); |
| 804 | } else { |
| 805 | invoker_.AsyncInvokeDelayed<void>( |
| 806 | RTC_FROM_HERE, rtc::Thread::Current(), |
| 807 | rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists, |
| 808 | this, sdp_mid, sdp_mline_index, msg), |
| 809 | signaling_delay_ms_); |
| 810 | } |
| 811 | } |
| 812 | |
| 813 | void RelayIceMessageIfReceiverExists(const std::string& sdp_mid, |
| 814 | int sdp_mline_index, |
| 815 | const std::string& msg) { |
| 816 | if (signaling_message_receiver_) { |
| 817 | signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index, |
| 818 | msg); |
| 819 | } |
| 820 | } |
| 821 | |
| 822 | // SignalingMessageReceiver callbacks. |
Steve Anton | a3a92c2 | 2017-12-07 10:27:41 -0800 | [diff] [blame] | 823 | void ReceiveSdpMessage(SdpType type, const std::string& msg) override { |
| 824 | if (type == SdpType::kOffer) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 825 | HandleIncomingOffer(msg); |
| 826 | } else { |
| 827 | HandleIncomingAnswer(msg); |
| 828 | } |
| 829 | } |
| 830 | |
| 831 | void ReceiveIceMessage(const std::string& sdp_mid, |
| 832 | int sdp_mline_index, |
| 833 | const std::string& msg) override { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 834 | RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage"; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 835 | std::unique_ptr<webrtc::IceCandidateInterface> candidate( |
| 836 | webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr)); |
| 837 | EXPECT_TRUE(pc()->AddIceCandidate(candidate.get())); |
| 838 | } |
| 839 | |
| 840 | // PeerConnectionObserver callbacks. |
| 841 | void OnSignalingChange( |
| 842 | webrtc::PeerConnectionInterface::SignalingState new_state) override { |
| 843 | EXPECT_EQ(pc()->signaling_state(), new_state); |
| 844 | } |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 845 | void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver, |
| 846 | const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& |
| 847 | streams) override { |
| 848 | if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| 849 | rtc::scoped_refptr<VideoTrackInterface> video_track( |
| 850 | static_cast<VideoTrackInterface*>(receiver->track().get())); |
| 851 | ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) == |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 852 | fake_video_renderers_.end()); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 853 | fake_video_renderers_[video_track->id()] = |
| 854 | rtc::MakeUnique<FakeVideoTrackRenderer>(video_track); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 855 | } |
| 856 | } |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 857 | void OnRemoveTrack( |
| 858 | rtc::scoped_refptr<RtpReceiverInterface> receiver) override { |
| 859 | if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| 860 | auto it = fake_video_renderers_.find(receiver->track()->id()); |
| 861 | RTC_DCHECK(it != fake_video_renderers_.end()); |
| 862 | fake_video_renderers_.erase(it); |
| 863 | } |
| 864 | } |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 865 | void OnRenegotiationNeeded() override {} |
| 866 | void OnIceConnectionChange( |
| 867 | webrtc::PeerConnectionInterface::IceConnectionState new_state) override { |
| 868 | EXPECT_EQ(pc()->ice_connection_state(), new_state); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 869 | ice_connection_state_history_.push_back(new_state); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 870 | } |
| 871 | void OnIceGatheringChange( |
| 872 | webrtc::PeerConnectionInterface::IceGatheringState new_state) override { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 873 | EXPECT_EQ(pc()->ice_gathering_state(), new_state); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 874 | ice_gathering_state_history_.push_back(new_state); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 875 | } |
| 876 | void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 877 | RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate"; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 878 | |
| 879 | std::string ice_sdp; |
| 880 | EXPECT_TRUE(candidate->ToString(&ice_sdp)); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 881 | if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 882 | // Remote party may be deleted. |
| 883 | return; |
| 884 | } |
| 885 | SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp); |
| 886 | } |
| 887 | void OnDataChannel( |
| 888 | rtc::scoped_refptr<DataChannelInterface> data_channel) override { |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 889 | RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel"; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 890 | data_channel_ = data_channel; |
| 891 | data_observer_.reset(new MockDataChannelObserver(data_channel)); |
| 892 | } |
| 893 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 894 | std::string debug_name_; |
| 895 | |
| 896 | std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_; |
| 897 | |
| 898 | rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| 899 | rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| 900 | peer_connection_factory_; |
| 901 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 902 | cricket::PortAllocator* port_allocator_; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 903 | // Needed to keep track of number of frames sent. |
| 904 | rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| 905 | // Needed to keep track of number of frames received. |
| 906 | std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>> |
| 907 | fake_video_renderers_; |
| 908 | // Needed to ensure frames aren't received for removed tracks. |
| 909 | std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>> |
| 910 | removed_fake_video_renderers_; |
| 911 | // Needed to keep track of number of frames received when external decoder |
| 912 | // used. |
| 913 | FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr; |
| 914 | FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr; |
| 915 | bool video_decoder_factory_enabled_ = false; |
| 916 | |
| 917 | // For remote peer communication. |
| 918 | SignalingMessageReceiver* signaling_message_receiver_ = nullptr; |
| 919 | int signaling_delay_ms_ = 0; |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 920 | bool signal_ice_candidates_ = true; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 921 | |
| 922 | // Store references to the video capturers we've created, so that we can stop |
| 923 | // them, if required. |
| 924 | std::vector<cricket::FakeVideoCapturer*> video_capturers_; |
| 925 | // |local_video_renderer_| attached to the first created local video track. |
| 926 | std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_; |
| 927 | |
| 928 | PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_; |
| 929 | std::function<void(cricket::SessionDescription*)> received_sdp_munger_; |
| 930 | std::function<void(cricket::SessionDescription*)> generated_sdp_munger_; |
| 931 | |
| 932 | rtc::scoped_refptr<DataChannelInterface> data_channel_; |
| 933 | std::unique_ptr<MockDataChannelObserver> data_observer_; |
| 934 | |
| 935 | std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_; |
| 936 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 937 | std::vector<PeerConnectionInterface::IceConnectionState> |
| 938 | ice_connection_state_history_; |
| 939 | std::vector<PeerConnectionInterface::IceGatheringState> |
| 940 | ice_gathering_state_history_; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 941 | |
| 942 | rtc::AsyncInvoker invoker_; |
| 943 | |
| 944 | friend class PeerConnectionIntegrationTest; |
| 945 | }; |
| 946 | |
Elad Alon | 99c3fe5 | 2017-10-13 16:29:40 +0200 | [diff] [blame] | 947 | class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput { |
| 948 | public: |
| 949 | virtual ~MockRtcEventLogOutput() = default; |
| 950 | MOCK_CONST_METHOD0(IsActive, bool()); |
| 951 | MOCK_METHOD1(Write, bool(const std::string&)); |
| 952 | }; |
| 953 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 954 | // Tests two PeerConnections connecting to each other end-to-end, using a |
| 955 | // virtual network, fake A/V capture and fake encoder/decoders. The |
| 956 | // PeerConnections share the threads/socket servers, but use separate versions |
| 957 | // of everything else (including "PeerConnectionFactory"s). |
| 958 | class PeerConnectionIntegrationTest : public testing::Test { |
| 959 | public: |
| 960 | PeerConnectionIntegrationTest() |
deadbeef | 98e186c | 2017-05-16 18:00:06 -0700 | [diff] [blame] | 961 | : ss_(new rtc::VirtualSocketServer()), |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 962 | fss_(new rtc::FirewallSocketServer(ss_.get())), |
| 963 | network_thread_(new rtc::Thread(fss_.get())), |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 964 | worker_thread_(rtc::Thread::Create()) { |
| 965 | RTC_CHECK(network_thread_->Start()); |
| 966 | RTC_CHECK(worker_thread_->Start()); |
| 967 | } |
| 968 | |
| 969 | ~PeerConnectionIntegrationTest() { |
| 970 | if (caller_) { |
| 971 | caller_->set_signaling_message_receiver(nullptr); |
| 972 | } |
| 973 | if (callee_) { |
| 974 | callee_->set_signaling_message_receiver(nullptr); |
| 975 | } |
| 976 | } |
| 977 | |
| 978 | bool SignalingStateStable() { |
| 979 | return caller_->SignalingStateStable() && callee_->SignalingStateStable(); |
| 980 | } |
| 981 | |
deadbeef | 7145280 | 2017-05-07 17:21:01 -0700 | [diff] [blame] | 982 | bool DtlsConnected() { |
| 983 | // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS |
| 984 | // are connected. This is an important distinction. Once we have separate |
| 985 | // ICE and DTLS state, this check needs to use the DTLS state. |
| 986 | return (callee()->ice_connection_state() == |
| 987 | webrtc::PeerConnectionInterface::kIceConnectionConnected || |
| 988 | callee()->ice_connection_state() == |
| 989 | webrtc::PeerConnectionInterface::kIceConnectionCompleted) && |
| 990 | (caller()->ice_connection_state() == |
| 991 | webrtc::PeerConnectionInterface::kIceConnectionConnected || |
| 992 | caller()->ice_connection_state() == |
| 993 | webrtc::PeerConnectionInterface::kIceConnectionCompleted); |
| 994 | } |
| 995 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 996 | bool CreatePeerConnectionWrappers() { |
| 997 | return CreatePeerConnectionWrappersWithConfig( |
| 998 | PeerConnectionInterface::RTCConfiguration(), |
| 999 | PeerConnectionInterface::RTCConfiguration()); |
| 1000 | } |
| 1001 | |
| 1002 | bool CreatePeerConnectionWrappersWithConstraints( |
| 1003 | MediaConstraintsInterface* caller_constraints, |
| 1004 | MediaConstraintsInterface* callee_constraints) { |
| 1005 | caller_.reset(PeerConnectionWrapper::CreateWithConstraints( |
| 1006 | "Caller", caller_constraints, network_thread_.get(), |
| 1007 | worker_thread_.get())); |
| 1008 | callee_.reset(PeerConnectionWrapper::CreateWithConstraints( |
| 1009 | "Callee", callee_constraints, network_thread_.get(), |
| 1010 | worker_thread_.get())); |
| 1011 | return caller_ && callee_; |
| 1012 | } |
| 1013 | |
| 1014 | bool CreatePeerConnectionWrappersWithConfig( |
| 1015 | const PeerConnectionInterface::RTCConfiguration& caller_config, |
| 1016 | const PeerConnectionInterface::RTCConfiguration& callee_config) { |
| 1017 | caller_.reset(PeerConnectionWrapper::CreateWithConfig( |
| 1018 | "Caller", caller_config, network_thread_.get(), worker_thread_.get())); |
| 1019 | callee_.reset(PeerConnectionWrapper::CreateWithConfig( |
| 1020 | "Callee", callee_config, network_thread_.get(), worker_thread_.get())); |
| 1021 | return caller_ && callee_; |
| 1022 | } |
| 1023 | |
| 1024 | bool CreatePeerConnectionWrappersWithOptions( |
| 1025 | const PeerConnectionFactory::Options& caller_options, |
| 1026 | const PeerConnectionFactory::Options& callee_options) { |
| 1027 | caller_.reset(PeerConnectionWrapper::CreateWithOptions( |
| 1028 | "Caller", caller_options, network_thread_.get(), worker_thread_.get())); |
| 1029 | callee_.reset(PeerConnectionWrapper::CreateWithOptions( |
| 1030 | "Callee", callee_options, network_thread_.get(), worker_thread_.get())); |
| 1031 | return caller_ && callee_; |
| 1032 | } |
| 1033 | |
| 1034 | PeerConnectionWrapper* CreatePeerConnectionWrapperWithAlternateKey() { |
| 1035 | std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| 1036 | new FakeRTCCertificateGenerator()); |
| 1037 | cert_generator->use_alternate_key(); |
| 1038 | |
| 1039 | // Make sure the new client is using a different certificate. |
| 1040 | return PeerConnectionWrapper::CreateWithDtlsIdentityStore( |
| 1041 | "New Peer", std::move(cert_generator), network_thread_.get(), |
| 1042 | worker_thread_.get()); |
| 1043 | } |
| 1044 | |
| 1045 | // Once called, SDP blobs and ICE candidates will be automatically signaled |
| 1046 | // between PeerConnections. |
| 1047 | void ConnectFakeSignaling() { |
| 1048 | caller_->set_signaling_message_receiver(callee_.get()); |
| 1049 | callee_->set_signaling_message_receiver(caller_.get()); |
| 1050 | } |
| 1051 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 1052 | // Once called, SDP blobs will be automatically signaled between |
| 1053 | // PeerConnections. Note that ICE candidates will not be signaled unless they |
| 1054 | // are in the exchanged SDP blobs. |
| 1055 | void ConnectFakeSignalingForSdpOnly() { |
| 1056 | ConnectFakeSignaling(); |
| 1057 | SetSignalIceCandidates(false); |
| 1058 | } |
| 1059 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1060 | void SetSignalingDelayMs(int delay_ms) { |
| 1061 | caller_->set_signaling_delay_ms(delay_ms); |
| 1062 | callee_->set_signaling_delay_ms(delay_ms); |
| 1063 | } |
| 1064 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 1065 | void SetSignalIceCandidates(bool signal) { |
| 1066 | caller_->set_signal_ice_candidates(signal); |
| 1067 | callee_->set_signal_ice_candidates(signal); |
| 1068 | } |
| 1069 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1070 | void EnableVideoDecoderFactory() { |
| 1071 | caller_->EnableVideoDecoderFactory(); |
| 1072 | callee_->EnableVideoDecoderFactory(); |
| 1073 | } |
| 1074 | |
| 1075 | // Messages may get lost on the unreliable DataChannel, so we send multiple |
| 1076 | // times to avoid test flakiness. |
| 1077 | void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc, |
| 1078 | const std::string& data, |
| 1079 | int retries) { |
| 1080 | for (int i = 0; i < retries; ++i) { |
| 1081 | dc->Send(DataBuffer(data)); |
| 1082 | } |
| 1083 | } |
| 1084 | |
| 1085 | rtc::Thread* network_thread() { return network_thread_.get(); } |
| 1086 | |
| 1087 | rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); } |
| 1088 | |
| 1089 | PeerConnectionWrapper* caller() { return caller_.get(); } |
| 1090 | |
| 1091 | // Set the |caller_| to the |wrapper| passed in and return the |
| 1092 | // original |caller_|. |
| 1093 | PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent( |
| 1094 | PeerConnectionWrapper* wrapper) { |
| 1095 | PeerConnectionWrapper* old = caller_.release(); |
| 1096 | caller_.reset(wrapper); |
| 1097 | return old; |
| 1098 | } |
| 1099 | |
| 1100 | PeerConnectionWrapper* callee() { return callee_.get(); } |
| 1101 | |
| 1102 | // Set the |callee_| to the |wrapper| passed in and return the |
| 1103 | // original |callee_|. |
| 1104 | PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent( |
| 1105 | PeerConnectionWrapper* wrapper) { |
| 1106 | PeerConnectionWrapper* old = callee_.release(); |
| 1107 | callee_.reset(wrapper); |
| 1108 | return old; |
| 1109 | } |
| 1110 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 1111 | rtc::FirewallSocketServer* firewall() const { return fss_.get(); } |
| 1112 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1113 | // Expects the provided number of new frames to be received within |wait_ms|. |
| 1114 | // "New frames" meaning that it waits for the current frame counts to |
| 1115 | // *increase* by the provided values. For video, uses |
| 1116 | // RecievedVideoFramesForEachTrack for the case of multiple video tracks |
| 1117 | // being received. |
| 1118 | void ExpectNewFramesReceivedWithWait( |
| 1119 | int expected_caller_received_audio_frames, |
| 1120 | int expected_caller_received_video_frames, |
| 1121 | int expected_callee_received_audio_frames, |
| 1122 | int expected_callee_received_video_frames, |
| 1123 | int wait_ms) { |
| 1124 | // Add current frame counts to the provided values, in order to wait for |
| 1125 | // the frame count to increase. |
| 1126 | expected_caller_received_audio_frames += caller()->audio_frames_received(); |
| 1127 | expected_caller_received_video_frames += |
| 1128 | caller()->min_video_frames_received_per_track(); |
| 1129 | expected_callee_received_audio_frames += callee()->audio_frames_received(); |
| 1130 | expected_callee_received_video_frames += |
| 1131 | callee()->min_video_frames_received_per_track(); |
| 1132 | |
| 1133 | EXPECT_TRUE_WAIT(caller()->audio_frames_received() >= |
| 1134 | expected_caller_received_audio_frames && |
| 1135 | caller()->min_video_frames_received_per_track() >= |
| 1136 | expected_caller_received_video_frames && |
| 1137 | callee()->audio_frames_received() >= |
| 1138 | expected_callee_received_audio_frames && |
| 1139 | callee()->min_video_frames_received_per_track() >= |
| 1140 | expected_callee_received_video_frames, |
| 1141 | wait_ms); |
| 1142 | |
| 1143 | // After the combined wait, do an "expect" for each individual count, to |
| 1144 | // print out a more detailed message upon failure. |
| 1145 | EXPECT_GE(caller()->audio_frames_received(), |
| 1146 | expected_caller_received_audio_frames); |
| 1147 | EXPECT_GE(caller()->min_video_frames_received_per_track(), |
| 1148 | expected_caller_received_video_frames); |
| 1149 | EXPECT_GE(callee()->audio_frames_received(), |
| 1150 | expected_callee_received_audio_frames); |
| 1151 | EXPECT_GE(callee()->min_video_frames_received_per_track(), |
| 1152 | expected_callee_received_video_frames); |
| 1153 | } |
| 1154 | |
Tommi | 8e545ee | 2018-02-08 16:25:20 +0000 | [diff] [blame] | 1155 | void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled, |
| 1156 | bool remote_gcm_enabled, |
| 1157 | int expected_cipher_suite) { |
| 1158 | PeerConnectionFactory::Options caller_options; |
| 1159 | caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled; |
| 1160 | PeerConnectionFactory::Options callee_options; |
| 1161 | callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1162 | ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options, |
| 1163 | callee_options)); |
| 1164 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = |
| 1165 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); |
| 1166 | caller()->pc()->RegisterUMAObserver(caller_observer); |
| 1167 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1168 | caller()->AddAudioVideoTracks(); |
| 1169 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1170 | caller()->CreateAndSetAndSignalOffer(); |
| 1171 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1172 | EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite), |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1173 | caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1174 | EXPECT_EQ( |
| 1175 | 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
| 1176 | expected_cipher_suite)); |
| 1177 | caller()->pc()->RegisterUMAObserver(nullptr); |
| 1178 | } |
| 1179 | |
| 1180 | private: |
| 1181 | // |ss_| is used by |network_thread_| so it must be destroyed later. |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1182 | std::unique_ptr<rtc::VirtualSocketServer> ss_; |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 1183 | std::unique_ptr<rtc::FirewallSocketServer> fss_; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1184 | // |network_thread_| and |worker_thread_| are used by both |
| 1185 | // |caller_| and |callee_| so they must be destroyed |
| 1186 | // later. |
| 1187 | std::unique_ptr<rtc::Thread> network_thread_; |
| 1188 | std::unique_ptr<rtc::Thread> worker_thread_; |
| 1189 | std::unique_ptr<PeerConnectionWrapper> caller_; |
| 1190 | std::unique_ptr<PeerConnectionWrapper> callee_; |
| 1191 | }; |
| 1192 | |
| 1193 | // Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This |
| 1194 | // includes testing that the callback is invoked if an observer is connected |
| 1195 | // after the first packet has already been received. |
| 1196 | TEST_F(PeerConnectionIntegrationTest, |
| 1197 | RtpReceiverObserverOnFirstPacketReceived) { |
| 1198 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1199 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1200 | caller()->AddAudioVideoTracks(); |
| 1201 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1202 | // Start offer/answer exchange and wait for it to complete. |
| 1203 | caller()->CreateAndSetAndSignalOffer(); |
| 1204 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1205 | // Should be one receiver each for audio/video. |
| 1206 | EXPECT_EQ(2, caller()->rtp_receiver_observers().size()); |
| 1207 | EXPECT_EQ(2, callee()->rtp_receiver_observers().size()); |
| 1208 | // Wait for all "first packet received" callbacks to be fired. |
| 1209 | EXPECT_TRUE_WAIT( |
| 1210 | std::all_of(caller()->rtp_receiver_observers().begin(), |
| 1211 | caller()->rtp_receiver_observers().end(), |
| 1212 | [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| 1213 | return o->first_packet_received(); |
| 1214 | }), |
| 1215 | kMaxWaitForFramesMs); |
| 1216 | EXPECT_TRUE_WAIT( |
| 1217 | std::all_of(callee()->rtp_receiver_observers().begin(), |
| 1218 | callee()->rtp_receiver_observers().end(), |
| 1219 | [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| 1220 | return o->first_packet_received(); |
| 1221 | }), |
| 1222 | kMaxWaitForFramesMs); |
| 1223 | // If new observers are set after the first packet was already received, the |
| 1224 | // callback should still be invoked. |
| 1225 | caller()->ResetRtpReceiverObservers(); |
| 1226 | callee()->ResetRtpReceiverObservers(); |
| 1227 | EXPECT_EQ(2, caller()->rtp_receiver_observers().size()); |
| 1228 | EXPECT_EQ(2, callee()->rtp_receiver_observers().size()); |
| 1229 | EXPECT_TRUE( |
| 1230 | std::all_of(caller()->rtp_receiver_observers().begin(), |
| 1231 | caller()->rtp_receiver_observers().end(), |
| 1232 | [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| 1233 | return o->first_packet_received(); |
| 1234 | })); |
| 1235 | EXPECT_TRUE( |
| 1236 | std::all_of(callee()->rtp_receiver_observers().begin(), |
| 1237 | callee()->rtp_receiver_observers().end(), |
| 1238 | [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| 1239 | return o->first_packet_received(); |
| 1240 | })); |
| 1241 | } |
| 1242 | |
| 1243 | class DummyDtmfObserver : public DtmfSenderObserverInterface { |
| 1244 | public: |
| 1245 | DummyDtmfObserver() : completed_(false) {} |
| 1246 | |
| 1247 | // Implements DtmfSenderObserverInterface. |
| 1248 | void OnToneChange(const std::string& tone) override { |
| 1249 | tones_.push_back(tone); |
| 1250 | if (tone.empty()) { |
| 1251 | completed_ = true; |
| 1252 | } |
| 1253 | } |
| 1254 | |
| 1255 | const std::vector<std::string>& tones() const { return tones_; } |
| 1256 | bool completed() const { return completed_; } |
| 1257 | |
| 1258 | private: |
| 1259 | bool completed_; |
| 1260 | std::vector<std::string> tones_; |
| 1261 | }; |
| 1262 | |
| 1263 | // Assumes |sender| already has an audio track added and the offer/answer |
| 1264 | // exchange is done. |
| 1265 | void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender, |
| 1266 | PeerConnectionWrapper* receiver) { |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1267 | // We should be able to get a DTMF sender from the local sender. |
| 1268 | rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender = |
| 1269 | sender->pc()->GetSenders().at(0)->GetDtmfSender(); |
| 1270 | ASSERT_TRUE(dtmf_sender); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1271 | DummyDtmfObserver observer; |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1272 | dtmf_sender->RegisterObserver(&observer); |
| 1273 | |
| 1274 | // Test the DtmfSender object just created. |
| 1275 | EXPECT_TRUE(dtmf_sender->CanInsertDtmf()); |
| 1276 | EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50)); |
| 1277 | |
| 1278 | EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout); |
| 1279 | std::vector<std::string> tones = {"1", "a", ""}; |
| 1280 | EXPECT_EQ(tones, observer.tones()); |
| 1281 | dtmf_sender->UnregisterObserver(); |
| 1282 | // TODO(deadbeef): Verify the tones were actually received end-to-end. |
| 1283 | } |
| 1284 | |
| 1285 | // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each |
| 1286 | // direction). |
| 1287 | TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) { |
| 1288 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1289 | ConnectFakeSignaling(); |
| 1290 | // Only need audio for DTMF. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1291 | caller()->AddAudioTrack(); |
| 1292 | callee()->AddAudioTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1293 | caller()->CreateAndSetAndSignalOffer(); |
| 1294 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
deadbeef | 7145280 | 2017-05-07 17:21:01 -0700 | [diff] [blame] | 1295 | // DTLS must finish before the DTMF sender can be used reliably. |
| 1296 | ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1297 | TestDtmfFromSenderToReceiver(caller(), callee()); |
| 1298 | TestDtmfFromSenderToReceiver(callee(), caller()); |
| 1299 | } |
| 1300 | |
| 1301 | // Basic end-to-end test, verifying media can be encoded/transmitted/decoded |
| 1302 | // between two connections, using DTLS-SRTP. |
| 1303 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls) { |
| 1304 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1305 | ConnectFakeSignaling(); |
Harald Alvestrand | 194939b | 2018-01-24 16:04:13 +0100 | [diff] [blame] | 1306 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = |
| 1307 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); |
| 1308 | caller()->pc()->RegisterUMAObserver(caller_observer); |
| 1309 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1310 | // Do normal offer/answer and wait for some frames to be received in each |
| 1311 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1312 | caller()->AddAudioVideoTracks(); |
| 1313 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1314 | caller()->CreateAndSetAndSignalOffer(); |
| 1315 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1316 | ExpectNewFramesReceivedWithWait( |
| 1317 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1318 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1319 | kMaxWaitForFramesMs); |
Harald Alvestrand | 194939b | 2018-01-24 16:04:13 +0100 | [diff] [blame] | 1320 | EXPECT_LE( |
| 1321 | 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol, |
| 1322 | webrtc::kEnumCounterKeyProtocolDtls)); |
| 1323 | EXPECT_EQ( |
| 1324 | 0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol, |
| 1325 | webrtc::kEnumCounterKeyProtocolSdes)); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1326 | } |
| 1327 | |
| 1328 | // Uses SDES instead of DTLS for key agreement. |
| 1329 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSdes) { |
| 1330 | PeerConnectionInterface::RTCConfiguration sdes_config; |
| 1331 | sdes_config.enable_dtls_srtp.emplace(false); |
| 1332 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config)); |
| 1333 | ConnectFakeSignaling(); |
Harald Alvestrand | 194939b | 2018-01-24 16:04:13 +0100 | [diff] [blame] | 1334 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = |
| 1335 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); |
| 1336 | caller()->pc()->RegisterUMAObserver(caller_observer); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1337 | |
| 1338 | // Do normal offer/answer and wait for some frames to be received in each |
| 1339 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1340 | caller()->AddAudioVideoTracks(); |
| 1341 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1342 | caller()->CreateAndSetAndSignalOffer(); |
| 1343 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1344 | ExpectNewFramesReceivedWithWait( |
| 1345 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1346 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1347 | kMaxWaitForFramesMs); |
Harald Alvestrand | 194939b | 2018-01-24 16:04:13 +0100 | [diff] [blame] | 1348 | EXPECT_LE( |
| 1349 | 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol, |
| 1350 | webrtc::kEnumCounterKeyProtocolSdes)); |
| 1351 | EXPECT_EQ( |
| 1352 | 0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol, |
| 1353 | webrtc::kEnumCounterKeyProtocolDtls)); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1354 | } |
| 1355 | |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 1356 | // Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS |
| 1357 | // certificate once the DTLS handshake has finished. |
| 1358 | TEST_F(PeerConnectionIntegrationTest, |
| 1359 | GetRemoteAudioSSLCertificateReturnsExchangedCertificate) { |
| 1360 | auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) { |
| 1361 | auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc()); |
| 1362 | auto pc = reinterpret_cast<PeerConnection*>(pci->internal()); |
| 1363 | return pc->GetRemoteAudioSSLCertificate(); |
| 1364 | }; |
Zhi Huang | 70b820f | 2018-01-27 14:16:15 -0800 | [diff] [blame] | 1365 | auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) { |
| 1366 | auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc()); |
| 1367 | auto pc = reinterpret_cast<PeerConnection*>(pci->internal()); |
| 1368 | return pc->GetRemoteAudioSSLCertChain(); |
| 1369 | }; |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 1370 | |
| 1371 | auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]); |
| 1372 | auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]); |
| 1373 | |
| 1374 | // Configure each side with a known certificate so they can be compared later. |
| 1375 | PeerConnectionInterface::RTCConfiguration caller_config; |
| 1376 | caller_config.enable_dtls_srtp.emplace(true); |
| 1377 | caller_config.certificates.push_back(caller_cert); |
| 1378 | PeerConnectionInterface::RTCConfiguration callee_config; |
| 1379 | callee_config.enable_dtls_srtp.emplace(true); |
| 1380 | callee_config.certificates.push_back(callee_cert); |
| 1381 | ASSERT_TRUE( |
| 1382 | CreatePeerConnectionWrappersWithConfig(caller_config, callee_config)); |
| 1383 | ConnectFakeSignaling(); |
| 1384 | |
| 1385 | // When first initialized, there should not be a remote SSL certificate (and |
| 1386 | // calling this method should not crash). |
| 1387 | EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller())); |
| 1388 | EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee())); |
Zhi Huang | 70b820f | 2018-01-27 14:16:15 -0800 | [diff] [blame] | 1389 | EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller())); |
| 1390 | EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee())); |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 1391 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1392 | caller()->AddAudioTrack(); |
| 1393 | callee()->AddAudioTrack(); |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 1394 | caller()->CreateAndSetAndSignalOffer(); |
| 1395 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1396 | ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| 1397 | |
| 1398 | // Once DTLS has been connected, each side should return the other's SSL |
| 1399 | // certificate when calling GetRemoteAudioSSLCertificate. |
| 1400 | |
| 1401 | auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller()); |
| 1402 | ASSERT_TRUE(caller_remote_cert); |
| 1403 | EXPECT_EQ(callee_cert->ssl_certificate().ToPEMString(), |
| 1404 | caller_remote_cert->ToPEMString()); |
| 1405 | |
| 1406 | auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee()); |
| 1407 | ASSERT_TRUE(callee_remote_cert); |
| 1408 | EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(), |
| 1409 | callee_remote_cert->ToPEMString()); |
Zhi Huang | 70b820f | 2018-01-27 14:16:15 -0800 | [diff] [blame] | 1410 | |
| 1411 | auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller()); |
| 1412 | ASSERT_TRUE(caller_remote_cert_chain); |
| 1413 | ASSERT_EQ(1U, caller_remote_cert_chain->GetSize()); |
| 1414 | auto remote_cert = &caller_remote_cert_chain->Get(0); |
| 1415 | EXPECT_EQ(callee_cert->ssl_certificate().ToPEMString(), |
| 1416 | remote_cert->ToPEMString()); |
| 1417 | |
| 1418 | auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee()); |
| 1419 | ASSERT_TRUE(callee_remote_cert_chain); |
| 1420 | ASSERT_EQ(1U, callee_remote_cert_chain->GetSize()); |
| 1421 | remote_cert = &callee_remote_cert_chain->Get(0); |
| 1422 | EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(), |
| 1423 | remote_cert->ToPEMString()); |
Steve Anton | 8c0f7a7 | 2017-10-03 10:03:10 -0700 | [diff] [blame] | 1424 | } |
| 1425 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1426 | // This test sets up a call between two parties (using DTLS) and tests that we |
| 1427 | // can get a video aspect ratio of 16:9. |
| 1428 | TEST_F(PeerConnectionIntegrationTest, SendAndReceive16To9AspectRatio) { |
| 1429 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1430 | ConnectFakeSignaling(); |
| 1431 | |
| 1432 | // Add video tracks with 16:9 constraint. |
| 1433 | FakeConstraints constraints; |
| 1434 | double requested_ratio = 16.0 / 9; |
| 1435 | constraints.SetMandatoryMinAspectRatio(requested_ratio); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1436 | caller()->AddTrack( |
| 1437 | caller()->CreateLocalVideoTrackWithConstraints(constraints)); |
| 1438 | callee()->AddTrack( |
| 1439 | callee()->CreateLocalVideoTrackWithConstraints(constraints)); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1440 | |
| 1441 | // Do normal offer/answer and wait for at least one frame to be received in |
| 1442 | // each direction. |
| 1443 | caller()->CreateAndSetAndSignalOffer(); |
| 1444 | ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| 1445 | callee()->min_video_frames_received_per_track() > 0, |
| 1446 | kMaxWaitForFramesMs); |
| 1447 | |
| 1448 | // Check rendered aspect ratio. |
| 1449 | EXPECT_EQ(requested_ratio, caller()->local_rendered_aspect_ratio()); |
| 1450 | EXPECT_EQ(requested_ratio, caller()->rendered_aspect_ratio()); |
| 1451 | EXPECT_EQ(requested_ratio, callee()->local_rendered_aspect_ratio()); |
| 1452 | EXPECT_EQ(requested_ratio, callee()->rendered_aspect_ratio()); |
| 1453 | } |
| 1454 | |
| 1455 | // This test sets up a call between two parties with a source resolution of |
| 1456 | // 1280x720 and verifies that a 16:9 aspect ratio is received. |
| 1457 | TEST_F(PeerConnectionIntegrationTest, |
| 1458 | Send1280By720ResolutionAndReceive16To9AspectRatio) { |
| 1459 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1460 | ConnectFakeSignaling(); |
| 1461 | |
| 1462 | // Similar to above test, but uses MandatoryMin[Width/Height] constraint |
| 1463 | // instead of aspect ratio constraint. |
| 1464 | FakeConstraints constraints; |
| 1465 | constraints.SetMandatoryMinWidth(1280); |
| 1466 | constraints.SetMandatoryMinHeight(720); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1467 | caller()->AddTrack( |
| 1468 | caller()->CreateLocalVideoTrackWithConstraints(constraints)); |
| 1469 | callee()->AddTrack( |
| 1470 | callee()->CreateLocalVideoTrackWithConstraints(constraints)); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1471 | |
| 1472 | // Do normal offer/answer and wait for at least one frame to be received in |
| 1473 | // each direction. |
| 1474 | caller()->CreateAndSetAndSignalOffer(); |
| 1475 | ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| 1476 | callee()->min_video_frames_received_per_track() > 0, |
| 1477 | kMaxWaitForFramesMs); |
| 1478 | |
| 1479 | // Check rendered aspect ratio. |
| 1480 | EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio()); |
| 1481 | EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio()); |
| 1482 | EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio()); |
| 1483 | EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio()); |
| 1484 | } |
| 1485 | |
| 1486 | // This test sets up an one-way call, with media only from caller to |
| 1487 | // callee. |
| 1488 | TEST_F(PeerConnectionIntegrationTest, OneWayMediaCall) { |
| 1489 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1490 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1491 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1492 | caller()->CreateAndSetAndSignalOffer(); |
| 1493 | int caller_received_frames = 0; |
| 1494 | ExpectNewFramesReceivedWithWait( |
| 1495 | caller_received_frames, caller_received_frames, |
| 1496 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1497 | kMaxWaitForFramesMs); |
| 1498 | } |
| 1499 | |
| 1500 | // This test sets up a audio call initially, with the callee rejecting video |
| 1501 | // initially. Then later the callee decides to upgrade to audio/video, and |
| 1502 | // initiates a new offer/answer exchange. |
| 1503 | TEST_F(PeerConnectionIntegrationTest, AudioToVideoUpgrade) { |
| 1504 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1505 | ConnectFakeSignaling(); |
| 1506 | // Initially, offer an audio/video stream from the caller, but refuse to |
| 1507 | // send/receive video on the callee side. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1508 | caller()->AddAudioVideoTracks(); |
| 1509 | callee()->AddAudioTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1510 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 1511 | options.offer_to_receive_video = 0; |
| 1512 | callee()->SetOfferAnswerOptions(options); |
| 1513 | // Do offer/answer and make sure audio is still received end-to-end. |
| 1514 | caller()->CreateAndSetAndSignalOffer(); |
| 1515 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1516 | ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0, |
| 1517 | kDefaultExpectedAudioFrameCount, 0, |
| 1518 | kMaxWaitForFramesMs); |
| 1519 | // Sanity check that the callee's description has a rejected video section. |
| 1520 | ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| 1521 | const ContentInfo* callee_video_content = |
| 1522 | GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| 1523 | ASSERT_NE(nullptr, callee_video_content); |
| 1524 | EXPECT_TRUE(callee_video_content->rejected); |
| 1525 | // Now negotiate with video and ensure negotiation succeeds, with video |
| 1526 | // frames and additional audio frames being received. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1527 | callee()->AddVideoTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1528 | options.offer_to_receive_video = 1; |
| 1529 | callee()->SetOfferAnswerOptions(options); |
| 1530 | callee()->CreateAndSetAndSignalOffer(); |
| 1531 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1532 | // Expect additional audio frames to be received after the upgrade. |
| 1533 | ExpectNewFramesReceivedWithWait( |
| 1534 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1535 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1536 | kMaxWaitForFramesMs); |
| 1537 | } |
| 1538 | |
deadbeef | 4389b4d | 2017-09-07 09:07:36 -0700 | [diff] [blame] | 1539 | // Simpler than the above test; just add an audio track to an established |
| 1540 | // video-only connection. |
| 1541 | TEST_F(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) { |
| 1542 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1543 | ConnectFakeSignaling(); |
| 1544 | // Do initial offer/answer with just a video track. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1545 | caller()->AddVideoTrack(); |
| 1546 | callee()->AddVideoTrack(); |
deadbeef | 4389b4d | 2017-09-07 09:07:36 -0700 | [diff] [blame] | 1547 | caller()->CreateAndSetAndSignalOffer(); |
| 1548 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1549 | // Now add an audio track and do another offer/answer. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1550 | caller()->AddAudioTrack(); |
| 1551 | callee()->AddAudioTrack(); |
deadbeef | 4389b4d | 2017-09-07 09:07:36 -0700 | [diff] [blame] | 1552 | caller()->CreateAndSetAndSignalOffer(); |
| 1553 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1554 | // Ensure both audio and video frames are received end-to-end. |
| 1555 | ExpectNewFramesReceivedWithWait( |
| 1556 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1557 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1558 | kMaxWaitForFramesMs); |
| 1559 | } |
| 1560 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1561 | // This test sets up a call that's transferred to a new caller with a different |
| 1562 | // DTLS fingerprint. |
| 1563 | TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) { |
| 1564 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1565 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1566 | caller()->AddAudioVideoTracks(); |
| 1567 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1568 | caller()->CreateAndSetAndSignalOffer(); |
| 1569 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1570 | |
| 1571 | // Keep the original peer around which will still send packets to the |
| 1572 | // receiving client. These SRTP packets will be dropped. |
| 1573 | std::unique_ptr<PeerConnectionWrapper> original_peer( |
| 1574 | SetCallerPcWrapperAndReturnCurrent( |
| 1575 | CreatePeerConnectionWrapperWithAlternateKey())); |
| 1576 | // TODO(deadbeef): Why do we call Close here? That goes against the comment |
| 1577 | // directly above. |
| 1578 | original_peer->pc()->Close(); |
| 1579 | |
| 1580 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1581 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1582 | caller()->CreateAndSetAndSignalOffer(); |
| 1583 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1584 | // Wait for some additional frames to be transmitted end-to-end. |
| 1585 | ExpectNewFramesReceivedWithWait( |
| 1586 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1587 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1588 | kMaxWaitForFramesMs); |
| 1589 | } |
| 1590 | |
| 1591 | // This test sets up a call that's transferred to a new callee with a different |
| 1592 | // DTLS fingerprint. |
| 1593 | TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) { |
| 1594 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1595 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1596 | caller()->AddAudioVideoTracks(); |
| 1597 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1598 | caller()->CreateAndSetAndSignalOffer(); |
| 1599 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1600 | |
| 1601 | // Keep the original peer around which will still send packets to the |
| 1602 | // receiving client. These SRTP packets will be dropped. |
| 1603 | std::unique_ptr<PeerConnectionWrapper> original_peer( |
| 1604 | SetCalleePcWrapperAndReturnCurrent( |
| 1605 | CreatePeerConnectionWrapperWithAlternateKey())); |
| 1606 | // TODO(deadbeef): Why do we call Close here? That goes against the comment |
| 1607 | // directly above. |
| 1608 | original_peer->pc()->Close(); |
| 1609 | |
| 1610 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1611 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1612 | caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); |
| 1613 | caller()->CreateAndSetAndSignalOffer(); |
| 1614 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1615 | // Wait for some additional frames to be transmitted end-to-end. |
| 1616 | ExpectNewFramesReceivedWithWait( |
| 1617 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1618 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1619 | kMaxWaitForFramesMs); |
| 1620 | } |
| 1621 | |
| 1622 | // This test sets up a non-bundled call and negotiates bundling at the same |
| 1623 | // time as starting an ICE restart. When bundling is in effect in the restart, |
| 1624 | // the DTLS-SRTP context should be successfully reset. |
| 1625 | TEST_F(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) { |
| 1626 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1627 | ConnectFakeSignaling(); |
| 1628 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1629 | caller()->AddAudioVideoTracks(); |
| 1630 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1631 | // Remove the bundle group from the SDP received by the callee. |
| 1632 | callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) { |
| 1633 | desc->RemoveGroupByName("BUNDLE"); |
| 1634 | }); |
| 1635 | caller()->CreateAndSetAndSignalOffer(); |
| 1636 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1637 | ExpectNewFramesReceivedWithWait( |
| 1638 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1639 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1640 | kMaxWaitForFramesMs); |
| 1641 | |
| 1642 | // Now stop removing the BUNDLE group, and trigger an ICE restart. |
| 1643 | callee()->SetReceivedSdpMunger(nullptr); |
| 1644 | caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); |
| 1645 | caller()->CreateAndSetAndSignalOffer(); |
| 1646 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1647 | |
| 1648 | // Expect additional frames to be received after the ICE restart. |
| 1649 | ExpectNewFramesReceivedWithWait( |
| 1650 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1651 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1652 | kMaxWaitForFramesMs); |
| 1653 | } |
| 1654 | |
| 1655 | // Test CVO (Coordination of Video Orientation). If a video source is rotated |
| 1656 | // and both peers support the CVO RTP header extension, the actual video frames |
| 1657 | // don't need to be encoded in different resolutions, since the rotation is |
| 1658 | // communicated through the RTP header extension. |
| 1659 | TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) { |
| 1660 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1661 | ConnectFakeSignaling(); |
| 1662 | // Add rotated video tracks. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1663 | caller()->AddTrack( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1664 | caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1665 | callee()->AddTrack( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1666 | callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); |
| 1667 | |
| 1668 | // Wait for video frames to be received by both sides. |
| 1669 | caller()->CreateAndSetAndSignalOffer(); |
| 1670 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1671 | ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| 1672 | callee()->min_video_frames_received_per_track() > 0, |
| 1673 | kMaxWaitForFramesMs); |
| 1674 | |
| 1675 | // Ensure that the aspect ratio is unmodified. |
| 1676 | // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, |
| 1677 | // not just assumed. |
| 1678 | EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio()); |
| 1679 | EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio()); |
| 1680 | EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio()); |
| 1681 | EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio()); |
| 1682 | // Ensure that the CVO bits were surfaced to the renderer. |
| 1683 | EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation()); |
| 1684 | EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation()); |
| 1685 | } |
| 1686 | |
| 1687 | // Test that when the CVO extension isn't supported, video is rotated the |
| 1688 | // old-fashioned way, by encoding rotated frames. |
| 1689 | TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) { |
| 1690 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1691 | ConnectFakeSignaling(); |
| 1692 | // Add rotated video tracks. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1693 | caller()->AddTrack( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1694 | caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1695 | callee()->AddTrack( |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1696 | callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); |
| 1697 | |
| 1698 | // Remove the CVO extension from the offered SDP. |
| 1699 | callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) { |
| 1700 | cricket::VideoContentDescription* video = |
| 1701 | GetFirstVideoContentDescription(desc); |
| 1702 | video->ClearRtpHeaderExtensions(); |
| 1703 | }); |
| 1704 | // Wait for video frames to be received by both sides. |
| 1705 | caller()->CreateAndSetAndSignalOffer(); |
| 1706 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1707 | ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| 1708 | callee()->min_video_frames_received_per_track() > 0, |
| 1709 | kMaxWaitForFramesMs); |
| 1710 | |
| 1711 | // Expect that the aspect ratio is inversed to account for the 90/270 degree |
| 1712 | // rotation. |
| 1713 | // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, |
| 1714 | // not just assumed. |
| 1715 | EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio()); |
| 1716 | EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio()); |
| 1717 | EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio()); |
| 1718 | EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio()); |
| 1719 | // Expect that each endpoint is unaware of the rotation of the other endpoint. |
| 1720 | EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation()); |
| 1721 | EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation()); |
| 1722 | } |
| 1723 | |
| 1724 | // TODO(deadbeef): The tests below rely on RTCOfferAnswerOptions to reject an |
| 1725 | // m= section. When we implement Unified Plan SDP, the right way to do this |
| 1726 | // would be by stopping an RtpTransceiver. |
| 1727 | |
| 1728 | // Test that if the answerer rejects the audio m= section, no audio is sent or |
| 1729 | // received, but video still can be. |
| 1730 | TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) { |
| 1731 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1732 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1733 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1734 | // Only add video track for callee, and set offer_to_receive_audio to 0, so |
| 1735 | // it will reject the audio m= section completely. |
| 1736 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 1737 | options.offer_to_receive_audio = 0; |
| 1738 | callee()->SetOfferAnswerOptions(options); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1739 | callee()->AddTrack(callee()->CreateLocalVideoTrack()); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1740 | // Do offer/answer and wait for successful end-to-end video frames. |
| 1741 | caller()->CreateAndSetAndSignalOffer(); |
| 1742 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1743 | ExpectNewFramesReceivedWithWait(0, kDefaultExpectedVideoFrameCount, 0, |
| 1744 | kDefaultExpectedVideoFrameCount, |
| 1745 | kMaxWaitForFramesMs); |
| 1746 | // Shouldn't have received audio frames at any point. |
| 1747 | EXPECT_EQ(0, caller()->audio_frames_received()); |
| 1748 | EXPECT_EQ(0, callee()->audio_frames_received()); |
| 1749 | // Sanity check that the callee's description has a rejected audio section. |
| 1750 | ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| 1751 | const ContentInfo* callee_audio_content = |
| 1752 | GetFirstAudioContent(callee()->pc()->local_description()->description()); |
| 1753 | ASSERT_NE(nullptr, callee_audio_content); |
| 1754 | EXPECT_TRUE(callee_audio_content->rejected); |
| 1755 | } |
| 1756 | |
| 1757 | // Test that if the answerer rejects the video m= section, no video is sent or |
| 1758 | // received, but audio still can be. |
| 1759 | TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) { |
| 1760 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1761 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1762 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1763 | // Only add audio track for callee, and set offer_to_receive_video to 0, so |
| 1764 | // it will reject the video m= section completely. |
| 1765 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 1766 | options.offer_to_receive_video = 0; |
| 1767 | callee()->SetOfferAnswerOptions(options); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1768 | callee()->AddTrack(callee()->CreateLocalAudioTrack()); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1769 | // Do offer/answer and wait for successful end-to-end audio frames. |
| 1770 | caller()->CreateAndSetAndSignalOffer(); |
| 1771 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1772 | ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0, |
| 1773 | kDefaultExpectedAudioFrameCount, 0, |
| 1774 | kMaxWaitForFramesMs); |
| 1775 | // Shouldn't have received video frames at any point. |
| 1776 | EXPECT_EQ(0, caller()->total_video_frames_received()); |
| 1777 | EXPECT_EQ(0, callee()->total_video_frames_received()); |
| 1778 | // Sanity check that the callee's description has a rejected video section. |
| 1779 | ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| 1780 | const ContentInfo* callee_video_content = |
| 1781 | GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| 1782 | ASSERT_NE(nullptr, callee_video_content); |
| 1783 | EXPECT_TRUE(callee_video_content->rejected); |
| 1784 | } |
| 1785 | |
| 1786 | // Test that if the answerer rejects both audio and video m= sections, nothing |
| 1787 | // bad happens. |
| 1788 | // TODO(deadbeef): Test that a data channel still works. Currently this doesn't |
| 1789 | // test anything but the fact that negotiation succeeds, which doesn't mean |
| 1790 | // much. |
| 1791 | TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) { |
| 1792 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1793 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1794 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1795 | // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it |
| 1796 | // will reject both audio and video m= sections. |
| 1797 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 1798 | options.offer_to_receive_audio = 0; |
| 1799 | options.offer_to_receive_video = 0; |
| 1800 | callee()->SetOfferAnswerOptions(options); |
| 1801 | // Do offer/answer and wait for stable signaling state. |
| 1802 | caller()->CreateAndSetAndSignalOffer(); |
| 1803 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1804 | // Sanity check that the callee's description has rejected m= sections. |
| 1805 | ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| 1806 | const ContentInfo* callee_audio_content = |
| 1807 | GetFirstAudioContent(callee()->pc()->local_description()->description()); |
| 1808 | ASSERT_NE(nullptr, callee_audio_content); |
| 1809 | EXPECT_TRUE(callee_audio_content->rejected); |
| 1810 | const ContentInfo* callee_video_content = |
| 1811 | GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| 1812 | ASSERT_NE(nullptr, callee_video_content); |
| 1813 | EXPECT_TRUE(callee_video_content->rejected); |
| 1814 | } |
| 1815 | |
| 1816 | // This test sets up an audio and video call between two parties. After the |
| 1817 | // call runs for a while, the caller sends an updated offer with video being |
| 1818 | // rejected. Once the re-negotiation is done, the video flow should stop and |
| 1819 | // the audio flow should continue. |
| 1820 | TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) { |
| 1821 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1822 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1823 | caller()->AddAudioVideoTracks(); |
| 1824 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1825 | caller()->CreateAndSetAndSignalOffer(); |
| 1826 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1827 | ExpectNewFramesReceivedWithWait( |
| 1828 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1829 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1830 | kMaxWaitForFramesMs); |
| 1831 | |
| 1832 | // Renegotiate, rejecting the video m= section. |
| 1833 | // TODO(deadbeef): When an RtpTransceiver API is available, use that to |
| 1834 | // reject the video m= section. |
| 1835 | caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) { |
| 1836 | for (cricket::ContentInfo& content : description->contents()) { |
| 1837 | if (cricket::IsVideoContent(&content)) { |
| 1838 | content.rejected = true; |
| 1839 | } |
| 1840 | } |
| 1841 | }); |
| 1842 | caller()->CreateAndSetAndSignalOffer(); |
| 1843 | ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); |
| 1844 | |
| 1845 | // Sanity check that the caller's description has a rejected video section. |
| 1846 | ASSERT_NE(nullptr, caller()->pc()->local_description()); |
| 1847 | const ContentInfo* caller_video_content = |
| 1848 | GetFirstVideoContent(caller()->pc()->local_description()->description()); |
| 1849 | ASSERT_NE(nullptr, caller_video_content); |
| 1850 | EXPECT_TRUE(caller_video_content->rejected); |
| 1851 | |
| 1852 | int caller_video_received = caller()->total_video_frames_received(); |
| 1853 | int callee_video_received = callee()->total_video_frames_received(); |
| 1854 | |
| 1855 | // Wait for some additional audio frames to be received. |
| 1856 | ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0, |
| 1857 | kDefaultExpectedAudioFrameCount, 0, |
| 1858 | kMaxWaitForFramesMs); |
| 1859 | |
| 1860 | // During this time, we shouldn't have received any additional video frames |
| 1861 | // for the rejected video tracks. |
| 1862 | EXPECT_EQ(caller_video_received, caller()->total_video_frames_received()); |
| 1863 | EXPECT_EQ(callee_video_received, callee()->total_video_frames_received()); |
| 1864 | } |
| 1865 | |
| 1866 | // Basic end-to-end test, but without SSRC/MSID signaling. This functionality |
| 1867 | // is needed to support legacy endpoints. |
| 1868 | // TODO(deadbeef): When we support the MID extension and demuxing on MID, also |
| 1869 | // add a test for an end-to-end test without MID signaling either (basically, |
| 1870 | // the minimum acceptable SDP). |
| 1871 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) { |
| 1872 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1873 | ConnectFakeSignaling(); |
| 1874 | // Add audio and video, testing that packets can be demuxed on payload type. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1875 | caller()->AddAudioVideoTracks(); |
| 1876 | callee()->AddAudioVideoTracks(); |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1877 | // Remove SSRCs and MSIDs from the received offer SDP. |
| 1878 | callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1879 | caller()->CreateAndSetAndSignalOffer(); |
| 1880 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1881 | ExpectNewFramesReceivedWithWait( |
| 1882 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1883 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1884 | kMaxWaitForFramesMs); |
| 1885 | } |
| 1886 | |
| 1887 | // Test that if two video tracks are sent (from caller to callee, in this test), |
| 1888 | // they're transmitted correctly end-to-end. |
| 1889 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) { |
| 1890 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1891 | ConnectFakeSignaling(); |
| 1892 | // Add one audio/video stream, and one video-only stream. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1893 | caller()->AddAudioVideoTracks(); |
| 1894 | caller()->AddVideoTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1895 | caller()->CreateAndSetAndSignalOffer(); |
| 1896 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1897 | ASSERT_EQ(3u, callee()->pc()->GetReceivers().size()); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1898 | int expected_callee_received_frames = kDefaultExpectedVideoFrameCount; |
| 1899 | ExpectNewFramesReceivedWithWait(0, 0, 0, expected_callee_received_frames, |
| 1900 | kMaxWaitForFramesMs); |
| 1901 | } |
| 1902 | |
| 1903 | static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) { |
| 1904 | bool first = true; |
| 1905 | for (cricket::ContentInfo& content : desc->contents()) { |
| 1906 | if (first) { |
| 1907 | first = false; |
| 1908 | continue; |
| 1909 | } |
| 1910 | content.bundle_only = true; |
| 1911 | } |
| 1912 | first = true; |
| 1913 | for (cricket::TransportInfo& transport : desc->transport_infos()) { |
| 1914 | if (first) { |
| 1915 | first = false; |
| 1916 | continue; |
| 1917 | } |
| 1918 | transport.description.ice_ufrag.clear(); |
| 1919 | transport.description.ice_pwd.clear(); |
| 1920 | transport.description.connection_role = cricket::CONNECTIONROLE_NONE; |
| 1921 | transport.description.identity_fingerprint.reset(nullptr); |
| 1922 | } |
| 1923 | } |
| 1924 | |
| 1925 | // Test that if applying a true "max bundle" offer, which uses ports of 0, |
| 1926 | // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and |
| 1927 | // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes |
| 1928 | // successfully and media flows. |
| 1929 | // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works. |
| 1930 | // TODO(deadbeef): Won't need this test once we start generating actual |
| 1931 | // standards-compliant SDP. |
| 1932 | TEST_F(PeerConnectionIntegrationTest, |
| 1933 | EndToEndCallWithSpecCompliantMaxBundleOffer) { |
| 1934 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1935 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1936 | caller()->AddAudioVideoTracks(); |
| 1937 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1938 | // Do the equivalent of setting the port to 0, adding a=bundle-only, and |
| 1939 | // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all |
| 1940 | // but the first m= section. |
| 1941 | callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer); |
| 1942 | caller()->CreateAndSetAndSignalOffer(); |
| 1943 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1944 | ExpectNewFramesReceivedWithWait( |
| 1945 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1946 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1947 | kMaxWaitForFramesMs); |
| 1948 | } |
| 1949 | |
| 1950 | // Test that we can receive the audio output level from a remote audio track. |
| 1951 | // TODO(deadbeef): Use a fake audio source and verify that the output level is |
| 1952 | // exactly what the source on the other side was configured with. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1953 | TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1954 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1955 | ConnectFakeSignaling(); |
| 1956 | // Just add an audio track. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1957 | caller()->AddAudioTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1958 | caller()->CreateAndSetAndSignalOffer(); |
| 1959 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1960 | |
| 1961 | // Get the audio output level stats. Note that the level is not available |
| 1962 | // until an RTCP packet has been received. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1963 | EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0, |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1964 | kMaxWaitForFramesMs); |
| 1965 | } |
| 1966 | |
| 1967 | // Test that an audio input level is reported. |
| 1968 | // TODO(deadbeef): Use a fake audio source and verify that the input level is |
| 1969 | // exactly what the source was configured with. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1970 | TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1971 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1972 | ConnectFakeSignaling(); |
| 1973 | // Just add an audio track. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1974 | caller()->AddAudioTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1975 | caller()->CreateAndSetAndSignalOffer(); |
| 1976 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1977 | |
| 1978 | // Get the audio input level stats. The level should be available very |
| 1979 | // soon after the test starts. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1980 | EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0, |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1981 | kMaxWaitForStatsMs); |
| 1982 | } |
| 1983 | |
| 1984 | // Test that we can get incoming byte counts from both audio and video tracks. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 1985 | TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1986 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 1987 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 1988 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 1989 | // Do offer/answer, wait for the callee to receive some frames. |
| 1990 | caller()->CreateAndSetAndSignalOffer(); |
| 1991 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 1992 | int expected_caller_received_frames = 0; |
| 1993 | ExpectNewFramesReceivedWithWait( |
| 1994 | expected_caller_received_frames, expected_caller_received_frames, |
| 1995 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 1996 | kMaxWaitForFramesMs); |
| 1997 | |
| 1998 | // Get a handle to the remote tracks created, so they can be used as GetStats |
| 1999 | // filters. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2000 | for (auto receiver : callee()->pc()->GetReceivers()) { |
| 2001 | // We received frames, so we definitely should have nonzero "received bytes" |
| 2002 | // stats at this point. |
| 2003 | EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(), |
| 2004 | 0); |
| 2005 | } |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2006 | } |
| 2007 | |
| 2008 | // Test that we can get outgoing byte counts from both audio and video tracks. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2009 | TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) { |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2010 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2011 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2012 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2013 | auto audio_track = caller()->CreateLocalAudioTrack(); |
| 2014 | auto video_track = caller()->CreateLocalVideoTrack(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2015 | caller()->AddTrack(audio_track); |
| 2016 | caller()->AddTrack(video_track); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2017 | // Do offer/answer, wait for the callee to receive some frames. |
| 2018 | caller()->CreateAndSetAndSignalOffer(); |
| 2019 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2020 | int expected_caller_received_frames = 0; |
| 2021 | ExpectNewFramesReceivedWithWait( |
| 2022 | expected_caller_received_frames, expected_caller_received_frames, |
| 2023 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2024 | kMaxWaitForFramesMs); |
| 2025 | |
| 2026 | // The callee received frames, so we definitely should have nonzero "sent |
| 2027 | // bytes" stats at this point. |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2028 | EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0); |
| 2029 | EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0); |
| 2030 | } |
| 2031 | |
Fredrik Solenberg | 73276ad | 2017-09-14 14:46:47 +0200 | [diff] [blame] | 2032 | // Test that we can get capture start ntp time. |
| 2033 | TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) { |
| 2034 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2035 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2036 | caller()->AddAudioTrack(); |
Fredrik Solenberg | 73276ad | 2017-09-14 14:46:47 +0200 | [diff] [blame] | 2037 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2038 | callee()->AddAudioTrack(); |
Fredrik Solenberg | 73276ad | 2017-09-14 14:46:47 +0200 | [diff] [blame] | 2039 | |
| 2040 | // Do offer/answer, wait for the callee to receive some frames. |
| 2041 | caller()->CreateAndSetAndSignalOffer(); |
| 2042 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2043 | |
| 2044 | // Get the remote audio track created on the receiver, so they can be used as |
| 2045 | // GetStats filters. |
| 2046 | StreamCollectionInterface* remote_streams = callee()->remote_streams(); |
| 2047 | ASSERT_EQ(1u, remote_streams->count()); |
| 2048 | ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size()); |
| 2049 | MediaStreamTrackInterface* remote_audio_track = |
| 2050 | remote_streams->at(0)->GetAudioTracks()[0]; |
| 2051 | |
| 2052 | // Get the audio output level stats. Note that the level is not available |
| 2053 | // until an RTCP packet has been received. |
| 2054 | EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)-> |
| 2055 | CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs); |
| 2056 | } |
| 2057 | |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2058 | // Test that we can get stats (using the new stats implemnetation) for |
| 2059 | // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in |
| 2060 | // SDP. |
| 2061 | TEST_F(PeerConnectionIntegrationTest, |
| 2062 | GetStatsForUnsignaledStreamWithNewStatsApi) { |
| 2063 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2064 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2065 | caller()->AddAudioTrack(); |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2066 | // Remove SSRCs and MSIDs from the received offer SDP. |
| 2067 | callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| 2068 | caller()->CreateAndSetAndSignalOffer(); |
| 2069 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2070 | // Wait for one audio frame to be received by the callee. |
| 2071 | ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs); |
| 2072 | |
| 2073 | // We received a frame, so we should have nonzero "bytes received" stats for |
| 2074 | // the unsignaled stream, if stats are working for it. |
| 2075 | rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| 2076 | callee()->NewGetStats(); |
| 2077 | ASSERT_NE(nullptr, report); |
| 2078 | auto inbound_stream_stats = |
| 2079 | report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>(); |
| 2080 | ASSERT_EQ(1U, inbound_stream_stats.size()); |
| 2081 | ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined()); |
| 2082 | ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U); |
zhihuang | f816493 | 2017-05-19 13:09:47 -0700 | [diff] [blame] | 2083 | ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined()); |
| 2084 | } |
| 2085 | |
| 2086 | // Test that we can successfully get the media related stats (audio level |
| 2087 | // etc.) for the unsignaled stream. |
| 2088 | TEST_F(PeerConnectionIntegrationTest, |
| 2089 | GetMediaStatsForUnsignaledStreamWithNewStatsApi) { |
| 2090 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2091 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2092 | caller()->AddAudioVideoTracks(); |
zhihuang | f816493 | 2017-05-19 13:09:47 -0700 | [diff] [blame] | 2093 | // Remove SSRCs and MSIDs from the received offer SDP. |
| 2094 | callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| 2095 | caller()->CreateAndSetAndSignalOffer(); |
| 2096 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2097 | // Wait for one audio frame to be received by the callee. |
| 2098 | ExpectNewFramesReceivedWithWait(0, 0, 1, 1, kMaxWaitForFramesMs); |
| 2099 | |
| 2100 | rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| 2101 | callee()->NewGetStats(); |
| 2102 | ASSERT_NE(nullptr, report); |
| 2103 | |
| 2104 | auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| 2105 | auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats); |
| 2106 | ASSERT_GE(audio_index, 0); |
| 2107 | EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined()); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2108 | } |
| 2109 | |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2110 | // Helper for test below. |
| 2111 | void ModifySsrcs(cricket::SessionDescription* desc) { |
| 2112 | for (ContentInfo& content : desc->contents()) { |
Steve Anton | b1c1de1 | 2017-12-21 15:14:30 -0800 | [diff] [blame] | 2113 | for (cricket::StreamParams& stream : |
| 2114 | content.media_description()->mutable_streams()) { |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2115 | for (uint32_t& ssrc : stream.ssrcs) { |
| 2116 | ssrc = rtc::CreateRandomId(); |
| 2117 | } |
| 2118 | } |
| 2119 | } |
| 2120 | } |
| 2121 | |
| 2122 | // Test that the "RTCMediaSteamTrackStats" object is updated correctly when |
| 2123 | // SSRCs are unsignaled, and the SSRC of the received (audio) stream changes. |
| 2124 | // This should result in two "RTCInboundRTPStreamStats", but only one |
| 2125 | // "RTCMediaStreamTrackStats", whose counters go up continuously rather than |
| 2126 | // being reset to 0 once the SSRC change occurs. |
| 2127 | // |
| 2128 | // Regression test for this bug: |
| 2129 | // https://bugs.chromium.org/p/webrtc/issues/detail?id=8158 |
| 2130 | // |
| 2131 | // The bug causes the track stats to only represent one of the two streams: |
| 2132 | // whichever one has the higher SSRC. So with this bug, there was a 50% chance |
| 2133 | // that the track stat counters would reset to 0 when the new stream is |
| 2134 | // received, and a 50% chance that they'll stop updating (while |
| 2135 | // "concealed_samples" continues increasing, due to silence being generated for |
| 2136 | // the inactive stream). |
| 2137 | TEST_F(PeerConnectionIntegrationTest, |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2138 | TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) { |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2139 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2140 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2141 | caller()->AddAudioTrack(); |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2142 | // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint |
| 2143 | // that doesn't signal SSRCs (from the callee's perspective). |
| 2144 | callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| 2145 | caller()->CreateAndSetAndSignalOffer(); |
| 2146 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2147 | // Wait for 50 audio frames (500ms of audio) to be received by the callee. |
| 2148 | ExpectNewFramesReceivedWithWait(0, 0, 25, 0, kMaxWaitForFramesMs); |
| 2149 | |
| 2150 | // Some audio frames were received, so we should have nonzero "samples |
| 2151 | // received" for the track. |
| 2152 | rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| 2153 | callee()->NewGetStats(); |
| 2154 | ASSERT_NE(nullptr, report); |
| 2155 | auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| 2156 | ASSERT_EQ(1U, track_stats.size()); |
| 2157 | ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined()); |
| 2158 | ASSERT_GT(*track_stats[0]->total_samples_received, 0U); |
| 2159 | // uint64_t prev_samples_received = *track_stats[0]->total_samples_received; |
| 2160 | |
| 2161 | // Create a new offer and munge it to cause the caller to use a new SSRC. |
| 2162 | caller()->SetGeneratedSdpMunger(ModifySsrcs); |
| 2163 | caller()->CreateAndSetAndSignalOffer(); |
| 2164 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2165 | // Wait for 25 more audio frames (250ms of audio) to be received, from the new |
| 2166 | // SSRC. |
| 2167 | ExpectNewFramesReceivedWithWait(0, 0, 25, 0, kMaxWaitForFramesMs); |
| 2168 | |
| 2169 | report = callee()->NewGetStats(); |
| 2170 | ASSERT_NE(nullptr, report); |
| 2171 | track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| 2172 | ASSERT_EQ(1U, track_stats.size()); |
| 2173 | ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined()); |
| 2174 | // The "total samples received" stat should only be greater than it was |
| 2175 | // before. |
| 2176 | // TODO(deadbeef): Uncomment this assertion once the bug is completely fixed. |
| 2177 | // Right now, the new SSRC will cause the counters to reset to 0. |
| 2178 | // EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received); |
| 2179 | |
| 2180 | // Additionally, the percentage of concealed samples (samples generated to |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2181 | // conceal packet loss) should be less than 50%. If it's greater, that's a |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2182 | // good sign that we're seeing stats from the old stream that's no longer |
| 2183 | // receiving packets, and is generating concealed samples of silence. |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2184 | constexpr double kAcceptableConcealedSamplesPercentage = 0.50; |
deadbeef | 4e2deab | 2017-09-20 13:56:21 -0700 | [diff] [blame] | 2185 | ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined()); |
| 2186 | EXPECT_LT(*track_stats[0]->concealed_samples, |
| 2187 | *track_stats[0]->total_samples_received * |
| 2188 | kAcceptableConcealedSamplesPercentage); |
| 2189 | |
| 2190 | // Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a |
| 2191 | // sanity check that the SSRC really changed. |
| 2192 | // TODO(deadbeef): This isn't working right now, because we're not returning |
| 2193 | // *any* stats for the inactive stream. Uncomment when the bug is completely |
| 2194 | // fixed. |
| 2195 | // auto inbound_stream_stats = |
| 2196 | // report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>(); |
| 2197 | // ASSERT_EQ(2U, inbound_stream_stats.size()); |
| 2198 | } |
| 2199 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2200 | // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. |
| 2201 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) { |
| 2202 | PeerConnectionFactory::Options dtls_10_options; |
| 2203 | dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| 2204 | ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, |
| 2205 | dtls_10_options)); |
| 2206 | ConnectFakeSignaling(); |
| 2207 | // Do normal offer/answer and wait for some frames to be received in each |
| 2208 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2209 | caller()->AddAudioVideoTracks(); |
| 2210 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2211 | caller()->CreateAndSetAndSignalOffer(); |
| 2212 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2213 | ExpectNewFramesReceivedWithWait( |
| 2214 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2215 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2216 | kMaxWaitForFramesMs); |
| 2217 | } |
| 2218 | |
| 2219 | // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated. |
| 2220 | TEST_F(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) { |
| 2221 | PeerConnectionFactory::Options dtls_10_options; |
| 2222 | dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| 2223 | ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, |
| 2224 | dtls_10_options)); |
| 2225 | ConnectFakeSignaling(); |
| 2226 | // Register UMA observer before signaling begins. |
| 2227 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = |
| 2228 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); |
| 2229 | caller()->pc()->RegisterUMAObserver(caller_observer); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2230 | caller()->AddAudioVideoTracks(); |
| 2231 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2232 | caller()->CreateAndSetAndSignalOffer(); |
| 2233 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2234 | EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher( |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2235 | caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT), |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2236 | kDefaultTimeout); |
| 2237 | EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2238 | caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2239 | EXPECT_EQ(1, |
| 2240 | caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
| 2241 | kDefaultSrtpCryptoSuite)); |
| 2242 | } |
| 2243 | |
| 2244 | // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated. |
| 2245 | TEST_F(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) { |
| 2246 | PeerConnectionFactory::Options dtls_12_options; |
| 2247 | dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| 2248 | ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options, |
| 2249 | dtls_12_options)); |
| 2250 | ConnectFakeSignaling(); |
| 2251 | // Register UMA observer before signaling begins. |
| 2252 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = |
| 2253 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); |
| 2254 | caller()->pc()->RegisterUMAObserver(caller_observer); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2255 | caller()->AddAudioVideoTracks(); |
| 2256 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2257 | caller()->CreateAndSetAndSignalOffer(); |
| 2258 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2259 | EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher( |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2260 | caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT), |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2261 | kDefaultTimeout); |
| 2262 | EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 2263 | caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2264 | EXPECT_EQ(1, |
| 2265 | caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, |
| 2266 | kDefaultSrtpCryptoSuite)); |
| 2267 | } |
| 2268 | |
| 2269 | // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the |
| 2270 | // callee only supports 1.0. |
| 2271 | TEST_F(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) { |
| 2272 | PeerConnectionFactory::Options caller_options; |
| 2273 | caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| 2274 | PeerConnectionFactory::Options callee_options; |
| 2275 | callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| 2276 | ASSERT_TRUE( |
| 2277 | CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); |
| 2278 | ConnectFakeSignaling(); |
| 2279 | // Do normal offer/answer and wait for some frames to be received in each |
| 2280 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2281 | caller()->AddAudioVideoTracks(); |
| 2282 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2283 | caller()->CreateAndSetAndSignalOffer(); |
| 2284 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2285 | ExpectNewFramesReceivedWithWait( |
| 2286 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2287 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2288 | kMaxWaitForFramesMs); |
| 2289 | } |
| 2290 | |
| 2291 | // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the |
| 2292 | // callee supports 1.2. |
| 2293 | TEST_F(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) { |
| 2294 | PeerConnectionFactory::Options caller_options; |
| 2295 | caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| 2296 | PeerConnectionFactory::Options callee_options; |
| 2297 | callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| 2298 | ASSERT_TRUE( |
| 2299 | CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); |
| 2300 | ConnectFakeSignaling(); |
| 2301 | // Do normal offer/answer and wait for some frames to be received in each |
| 2302 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2303 | caller()->AddAudioVideoTracks(); |
| 2304 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2305 | caller()->CreateAndSetAndSignalOffer(); |
| 2306 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2307 | ExpectNewFramesReceivedWithWait( |
| 2308 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2309 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2310 | kMaxWaitForFramesMs); |
| 2311 | } |
| 2312 | |
| 2313 | // Test that a non-GCM cipher is used if both sides only support non-GCM. |
| 2314 | TEST_F(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) { |
| 2315 | bool local_gcm_enabled = false; |
| 2316 | bool remote_gcm_enabled = false; |
| 2317 | int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| 2318 | TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| 2319 | expected_cipher_suite); |
| 2320 | } |
| 2321 | |
| 2322 | // Test that a GCM cipher is used if both ends support it. |
| 2323 | TEST_F(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) { |
| 2324 | bool local_gcm_enabled = true; |
| 2325 | bool remote_gcm_enabled = true; |
| 2326 | int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm; |
| 2327 | TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| 2328 | expected_cipher_suite); |
| 2329 | } |
| 2330 | |
| 2331 | // Test that GCM isn't used if only the offerer supports it. |
| 2332 | TEST_F(PeerConnectionIntegrationTest, |
| 2333 | NonGcmCipherUsedWhenOnlyCallerSupportsGcm) { |
| 2334 | bool local_gcm_enabled = true; |
| 2335 | bool remote_gcm_enabled = false; |
| 2336 | int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| 2337 | TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| 2338 | expected_cipher_suite); |
| 2339 | } |
| 2340 | |
| 2341 | // Test that GCM isn't used if only the answerer supports it. |
| 2342 | TEST_F(PeerConnectionIntegrationTest, |
| 2343 | NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) { |
| 2344 | bool local_gcm_enabled = false; |
| 2345 | bool remote_gcm_enabled = true; |
| 2346 | int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| 2347 | TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| 2348 | expected_cipher_suite); |
| 2349 | } |
| 2350 | |
deadbeef | 7914b8c | 2017-04-21 03:23:33 -0700 | [diff] [blame] | 2351 | // Verify that media can be transmitted end-to-end when GCM crypto suites are |
| 2352 | // enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported, |
| 2353 | // only verify that a GCM cipher is negotiated, and not necessarily that SRTP |
| 2354 | // works with it. |
| 2355 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) { |
| 2356 | PeerConnectionFactory::Options gcm_options; |
| 2357 | gcm_options.crypto_options.enable_gcm_crypto_suites = true; |
| 2358 | ASSERT_TRUE( |
| 2359 | CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options)); |
| 2360 | ConnectFakeSignaling(); |
| 2361 | // Do normal offer/answer and wait for some frames to be received in each |
| 2362 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2363 | caller()->AddAudioVideoTracks(); |
| 2364 | callee()->AddAudioVideoTracks(); |
deadbeef | 7914b8c | 2017-04-21 03:23:33 -0700 | [diff] [blame] | 2365 | caller()->CreateAndSetAndSignalOffer(); |
| 2366 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2367 | ExpectNewFramesReceivedWithWait( |
| 2368 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2369 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2370 | kMaxWaitForFramesMs); |
| 2371 | } |
| 2372 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2373 | // This test sets up a call between two parties with audio, video and an RTP |
| 2374 | // data channel. |
| 2375 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) { |
| 2376 | FakeConstraints setup_constraints; |
| 2377 | setup_constraints.SetAllowRtpDataChannels(); |
| 2378 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, |
| 2379 | &setup_constraints)); |
| 2380 | ConnectFakeSignaling(); |
| 2381 | // Expect that data channel created on caller side will show up for callee as |
| 2382 | // well. |
| 2383 | caller()->CreateDataChannel(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2384 | caller()->AddAudioVideoTracks(); |
| 2385 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2386 | caller()->CreateAndSetAndSignalOffer(); |
| 2387 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2388 | // Ensure the existence of the RTP data channel didn't impede audio/video. |
| 2389 | ExpectNewFramesReceivedWithWait( |
| 2390 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2391 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2392 | kMaxWaitForFramesMs); |
| 2393 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2394 | ASSERT_NE(nullptr, callee()->data_channel()); |
| 2395 | EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2396 | EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2397 | |
| 2398 | // Ensure data can be sent in both directions. |
| 2399 | std::string data = "hello world"; |
| 2400 | SendRtpDataWithRetries(caller()->data_channel(), data, 5); |
| 2401 | EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(), |
| 2402 | kDefaultTimeout); |
| 2403 | SendRtpDataWithRetries(callee()->data_channel(), data, 5); |
| 2404 | EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(), |
| 2405 | kDefaultTimeout); |
| 2406 | } |
| 2407 | |
| 2408 | // Ensure that an RTP data channel is signaled as closed for the caller when |
| 2409 | // the callee rejects it in a subsequent offer. |
| 2410 | TEST_F(PeerConnectionIntegrationTest, |
| 2411 | RtpDataChannelSignaledClosedInCalleeOffer) { |
| 2412 | // Same procedure as above test. |
| 2413 | FakeConstraints setup_constraints; |
| 2414 | setup_constraints.SetAllowRtpDataChannels(); |
| 2415 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, |
| 2416 | &setup_constraints)); |
| 2417 | ConnectFakeSignaling(); |
| 2418 | caller()->CreateDataChannel(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2419 | caller()->AddAudioVideoTracks(); |
| 2420 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2421 | caller()->CreateAndSetAndSignalOffer(); |
| 2422 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2423 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2424 | ASSERT_NE(nullptr, callee()->data_channel()); |
| 2425 | ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2426 | ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2427 | |
| 2428 | // Close the data channel on the callee, and do an updated offer/answer. |
| 2429 | callee()->data_channel()->Close(); |
| 2430 | callee()->CreateAndSetAndSignalOffer(); |
| 2431 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2432 | EXPECT_FALSE(caller()->data_observer()->IsOpen()); |
| 2433 | EXPECT_FALSE(callee()->data_observer()->IsOpen()); |
| 2434 | } |
| 2435 | |
| 2436 | // Tests that data is buffered in an RTP data channel until an observer is |
| 2437 | // registered for it. |
| 2438 | // |
| 2439 | // NOTE: RTP data channels can receive data before the underlying |
| 2440 | // transport has detected that a channel is writable and thus data can be |
| 2441 | // received before the data channel state changes to open. That is hard to test |
| 2442 | // but the same buffering is expected to be used in that case. |
| 2443 | TEST_F(PeerConnectionIntegrationTest, |
| 2444 | DataBufferedUntilRtpDataChannelObserverRegistered) { |
| 2445 | // Use fake clock and simulated network delay so that we predictably can wait |
| 2446 | // until an SCTP message has been delivered without "sleep()"ing. |
| 2447 | rtc::ScopedFakeClock fake_clock; |
| 2448 | // Some things use a time of "0" as a special value, so we need to start out |
| 2449 | // the fake clock at a nonzero time. |
| 2450 | // TODO(deadbeef): Fix this. |
| 2451 | fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1)); |
| 2452 | virtual_socket_server()->set_delay_mean(5); // 5 ms per hop. |
| 2453 | virtual_socket_server()->UpdateDelayDistribution(); |
| 2454 | |
| 2455 | FakeConstraints constraints; |
| 2456 | constraints.SetAllowRtpDataChannels(); |
| 2457 | ASSERT_TRUE( |
| 2458 | CreatePeerConnectionWrappersWithConstraints(&constraints, &constraints)); |
| 2459 | ConnectFakeSignaling(); |
| 2460 | caller()->CreateDataChannel(); |
| 2461 | caller()->CreateAndSetAndSignalOffer(); |
| 2462 | ASSERT_TRUE(caller()->data_channel() != nullptr); |
| 2463 | ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr, |
| 2464 | kDefaultTimeout, fake_clock); |
| 2465 | ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(), |
| 2466 | kDefaultTimeout, fake_clock); |
| 2467 | ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen, |
| 2468 | callee()->data_channel()->state(), kDefaultTimeout, |
| 2469 | fake_clock); |
| 2470 | |
| 2471 | // Unregister the observer which is normally automatically registered. |
| 2472 | callee()->data_channel()->UnregisterObserver(); |
| 2473 | // Send data and advance fake clock until it should have been received. |
| 2474 | std::string data = "hello world"; |
| 2475 | caller()->data_channel()->Send(DataBuffer(data)); |
| 2476 | SIMULATED_WAIT(false, 50, fake_clock); |
| 2477 | |
| 2478 | // Attach data channel and expect data to be received immediately. Note that |
| 2479 | // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any |
| 2480 | // further, but data can be received even if the callback is asynchronous. |
| 2481 | MockDataChannelObserver new_observer(callee()->data_channel()); |
| 2482 | EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout, |
| 2483 | fake_clock); |
| 2484 | } |
| 2485 | |
| 2486 | // This test sets up a call between two parties with audio, video and but only |
| 2487 | // the caller client supports RTP data channels. |
| 2488 | TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) { |
| 2489 | FakeConstraints setup_constraints_1; |
| 2490 | setup_constraints_1.SetAllowRtpDataChannels(); |
| 2491 | // Must disable DTLS to make negotiation succeed. |
| 2492 | setup_constraints_1.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, |
| 2493 | false); |
| 2494 | FakeConstraints setup_constraints_2; |
| 2495 | setup_constraints_2.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, |
| 2496 | false); |
| 2497 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints( |
| 2498 | &setup_constraints_1, &setup_constraints_2)); |
| 2499 | ConnectFakeSignaling(); |
| 2500 | caller()->CreateDataChannel(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2501 | caller()->AddAudioVideoTracks(); |
| 2502 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2503 | caller()->CreateAndSetAndSignalOffer(); |
| 2504 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2505 | // The caller should still have a data channel, but it should be closed, and |
| 2506 | // one should ever have been created for the callee. |
| 2507 | EXPECT_TRUE(caller()->data_channel() != nullptr); |
| 2508 | EXPECT_FALSE(caller()->data_observer()->IsOpen()); |
| 2509 | EXPECT_EQ(nullptr, callee()->data_channel()); |
| 2510 | } |
| 2511 | |
| 2512 | // This test sets up a call between two parties with audio, and video. When |
| 2513 | // audio and video is setup and flowing, an RTP data channel is negotiated. |
| 2514 | TEST_F(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) { |
| 2515 | FakeConstraints setup_constraints; |
| 2516 | setup_constraints.SetAllowRtpDataChannels(); |
| 2517 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, |
| 2518 | &setup_constraints)); |
| 2519 | ConnectFakeSignaling(); |
| 2520 | // Do initial offer/answer with audio/video. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2521 | caller()->AddAudioVideoTracks(); |
| 2522 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2523 | caller()->CreateAndSetAndSignalOffer(); |
| 2524 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2525 | // Create data channel and do new offer and answer. |
| 2526 | caller()->CreateDataChannel(); |
| 2527 | caller()->CreateAndSetAndSignalOffer(); |
| 2528 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2529 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2530 | ASSERT_NE(nullptr, callee()->data_channel()); |
| 2531 | EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2532 | EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2533 | // Ensure data can be sent in both directions. |
| 2534 | std::string data = "hello world"; |
| 2535 | SendRtpDataWithRetries(caller()->data_channel(), data, 5); |
| 2536 | EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(), |
| 2537 | kDefaultTimeout); |
| 2538 | SendRtpDataWithRetries(callee()->data_channel(), data, 5); |
| 2539 | EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(), |
| 2540 | kDefaultTimeout); |
| 2541 | } |
| 2542 | |
| 2543 | #ifdef HAVE_SCTP |
| 2544 | |
| 2545 | // This test sets up a call between two parties with audio, video and an SCTP |
| 2546 | // data channel. |
| 2547 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) { |
| 2548 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2549 | ConnectFakeSignaling(); |
| 2550 | // Expect that data channel created on caller side will show up for callee as |
| 2551 | // well. |
| 2552 | caller()->CreateDataChannel(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2553 | caller()->AddAudioVideoTracks(); |
| 2554 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2555 | caller()->CreateAndSetAndSignalOffer(); |
| 2556 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2557 | // Ensure the existence of the SCTP data channel didn't impede audio/video. |
| 2558 | ExpectNewFramesReceivedWithWait( |
| 2559 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2560 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2561 | kMaxWaitForFramesMs); |
| 2562 | // Caller data channel should already exist (it created one). Callee data |
| 2563 | // channel may not exist yet, since negotiation happens in-band, not in SDP. |
| 2564 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2565 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2566 | EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2567 | EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2568 | |
| 2569 | // Ensure data can be sent in both directions. |
| 2570 | std::string data = "hello world"; |
| 2571 | caller()->data_channel()->Send(DataBuffer(data)); |
| 2572 | EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(), |
| 2573 | kDefaultTimeout); |
| 2574 | callee()->data_channel()->Send(DataBuffer(data)); |
| 2575 | EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(), |
| 2576 | kDefaultTimeout); |
| 2577 | } |
| 2578 | |
| 2579 | // Ensure that when the callee closes an SCTP data channel, the closing |
| 2580 | // procedure results in the data channel being closed for the caller as well. |
| 2581 | TEST_F(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) { |
| 2582 | // Same procedure as above test. |
| 2583 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2584 | ConnectFakeSignaling(); |
| 2585 | caller()->CreateDataChannel(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2586 | caller()->AddAudioVideoTracks(); |
| 2587 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2588 | caller()->CreateAndSetAndSignalOffer(); |
| 2589 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2590 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2591 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2592 | ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2593 | ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2594 | |
| 2595 | // Close the data channel on the callee side, and wait for it to reach the |
| 2596 | // "closed" state on both sides. |
| 2597 | callee()->data_channel()->Close(); |
| 2598 | EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2599 | EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2600 | } |
| 2601 | |
Steve Anton | da6c095 | 2017-10-23 11:41:54 -0700 | [diff] [blame] | 2602 | TEST_F(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) { |
| 2603 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2604 | ConnectFakeSignaling(); |
| 2605 | webrtc::DataChannelInit init; |
| 2606 | init.id = 53; |
| 2607 | init.maxRetransmits = 52; |
| 2608 | caller()->CreateDataChannel("data-channel", &init); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2609 | caller()->AddAudioVideoTracks(); |
| 2610 | callee()->AddAudioVideoTracks(); |
Steve Anton | da6c095 | 2017-10-23 11:41:54 -0700 | [diff] [blame] | 2611 | caller()->CreateAndSetAndSignalOffer(); |
| 2612 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
Steve Anton | 074dece | 2017-10-24 13:04:12 -0700 | [diff] [blame] | 2613 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2614 | ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
Steve Anton | da6c095 | 2017-10-23 11:41:54 -0700 | [diff] [blame] | 2615 | EXPECT_EQ(init.id, callee()->data_channel()->id()); |
| 2616 | EXPECT_EQ("data-channel", callee()->data_channel()->label()); |
| 2617 | EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits()); |
| 2618 | EXPECT_FALSE(callee()->data_channel()->negotiated()); |
| 2619 | } |
| 2620 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2621 | // Test usrsctp's ability to process unordered data stream, where data actually |
| 2622 | // arrives out of order using simulated delays. Previously there have been some |
| 2623 | // bugs in this area. |
| 2624 | TEST_F(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) { |
| 2625 | // Introduce random network delays. |
| 2626 | // Otherwise it's not a true "unordered" test. |
| 2627 | virtual_socket_server()->set_delay_mean(20); |
| 2628 | virtual_socket_server()->set_delay_stddev(5); |
| 2629 | virtual_socket_server()->UpdateDelayDistribution(); |
| 2630 | // Normal procedure, but with unordered data channel config. |
| 2631 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2632 | ConnectFakeSignaling(); |
| 2633 | webrtc::DataChannelInit init; |
| 2634 | init.ordered = false; |
| 2635 | caller()->CreateDataChannel(&init); |
| 2636 | caller()->CreateAndSetAndSignalOffer(); |
| 2637 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2638 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2639 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2640 | ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2641 | ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2642 | |
| 2643 | static constexpr int kNumMessages = 100; |
| 2644 | // Deliberately chosen to be larger than the MTU so messages get fragmented. |
| 2645 | static constexpr size_t kMaxMessageSize = 4096; |
| 2646 | // Create and send random messages. |
| 2647 | std::vector<std::string> sent_messages; |
| 2648 | for (int i = 0; i < kNumMessages; ++i) { |
| 2649 | size_t length = |
| 2650 | (rand() % kMaxMessageSize) + 1; // NOLINT (rand_r instead of rand) |
| 2651 | std::string message; |
| 2652 | ASSERT_TRUE(rtc::CreateRandomString(length, &message)); |
| 2653 | caller()->data_channel()->Send(DataBuffer(message)); |
| 2654 | callee()->data_channel()->Send(DataBuffer(message)); |
| 2655 | sent_messages.push_back(message); |
| 2656 | } |
| 2657 | |
| 2658 | // Wait for all messages to be received. |
| 2659 | EXPECT_EQ_WAIT(kNumMessages, |
| 2660 | caller()->data_observer()->received_message_count(), |
| 2661 | kDefaultTimeout); |
| 2662 | EXPECT_EQ_WAIT(kNumMessages, |
| 2663 | callee()->data_observer()->received_message_count(), |
| 2664 | kDefaultTimeout); |
| 2665 | |
| 2666 | // Sort and compare to make sure none of the messages were corrupted. |
| 2667 | std::vector<std::string> caller_received_messages = |
| 2668 | caller()->data_observer()->messages(); |
| 2669 | std::vector<std::string> callee_received_messages = |
| 2670 | callee()->data_observer()->messages(); |
| 2671 | std::sort(sent_messages.begin(), sent_messages.end()); |
| 2672 | std::sort(caller_received_messages.begin(), caller_received_messages.end()); |
| 2673 | std::sort(callee_received_messages.begin(), callee_received_messages.end()); |
| 2674 | EXPECT_EQ(sent_messages, caller_received_messages); |
| 2675 | EXPECT_EQ(sent_messages, callee_received_messages); |
| 2676 | } |
| 2677 | |
| 2678 | // This test sets up a call between two parties with audio, and video. When |
| 2679 | // audio and video are setup and flowing, an SCTP data channel is negotiated. |
| 2680 | TEST_F(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) { |
| 2681 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2682 | ConnectFakeSignaling(); |
| 2683 | // Do initial offer/answer with audio/video. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2684 | caller()->AddAudioVideoTracks(); |
| 2685 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2686 | caller()->CreateAndSetAndSignalOffer(); |
| 2687 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2688 | // Create data channel and do new offer and answer. |
| 2689 | caller()->CreateDataChannel(); |
| 2690 | caller()->CreateAndSetAndSignalOffer(); |
| 2691 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2692 | // Caller data channel should already exist (it created one). Callee data |
| 2693 | // channel may not exist yet, since negotiation happens in-band, not in SDP. |
| 2694 | ASSERT_NE(nullptr, caller()->data_channel()); |
| 2695 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2696 | EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2697 | EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2698 | // Ensure data can be sent in both directions. |
| 2699 | std::string data = "hello world"; |
| 2700 | caller()->data_channel()->Send(DataBuffer(data)); |
| 2701 | EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(), |
| 2702 | kDefaultTimeout); |
| 2703 | callee()->data_channel()->Send(DataBuffer(data)); |
| 2704 | EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(), |
| 2705 | kDefaultTimeout); |
| 2706 | } |
| 2707 | |
deadbeef | 7914b8c | 2017-04-21 03:23:33 -0700 | [diff] [blame] | 2708 | // Set up a connection initially just using SCTP data channels, later upgrading |
| 2709 | // to audio/video, ensuring frames are received end-to-end. Effectively the |
| 2710 | // inverse of the test above. |
| 2711 | // This was broken in M57; see https://crbug.com/711243 |
| 2712 | TEST_F(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) { |
| 2713 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2714 | ConnectFakeSignaling(); |
| 2715 | // Do initial offer/answer with just data channel. |
| 2716 | caller()->CreateDataChannel(); |
| 2717 | caller()->CreateAndSetAndSignalOffer(); |
| 2718 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2719 | // Wait until data can be sent over the data channel. |
| 2720 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2721 | ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2722 | ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2723 | |
| 2724 | // Do subsequent offer/answer with two-way audio and video. Audio and video |
| 2725 | // should end up bundled on the DTLS/ICE transport already used for data. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2726 | caller()->AddAudioVideoTracks(); |
| 2727 | callee()->AddAudioVideoTracks(); |
deadbeef | 7914b8c | 2017-04-21 03:23:33 -0700 | [diff] [blame] | 2728 | caller()->CreateAndSetAndSignalOffer(); |
| 2729 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2730 | ExpectNewFramesReceivedWithWait( |
| 2731 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2732 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 2733 | kMaxWaitForFramesMs); |
| 2734 | } |
| 2735 | |
deadbeef | 8b7e9ad | 2017-05-25 09:38:55 -0700 | [diff] [blame] | 2736 | static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) { |
deadbeef | 8b7e9ad | 2017-05-25 09:38:55 -0700 | [diff] [blame] | 2737 | cricket::DataContentDescription* dcd_offer = |
Steve Anton | b1c1de1 | 2017-12-21 15:14:30 -0800 | [diff] [blame] | 2738 | GetFirstDataContentDescription(desc); |
| 2739 | ASSERT_TRUE(dcd_offer); |
deadbeef | 8b7e9ad | 2017-05-25 09:38:55 -0700 | [diff] [blame] | 2740 | dcd_offer->set_use_sctpmap(false); |
| 2741 | dcd_offer->set_protocol("UDP/DTLS/SCTP"); |
| 2742 | } |
| 2743 | |
| 2744 | // Test that the data channel works when a spec-compliant SCTP m= section is |
| 2745 | // offered (using "a=sctp-port" instead of "a=sctpmap", and using |
| 2746 | // "UDP/DTLS/SCTP" as the protocol). |
| 2747 | TEST_F(PeerConnectionIntegrationTest, |
| 2748 | DataChannelWorksWhenSpecCompliantSctpOfferReceived) { |
| 2749 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2750 | ConnectFakeSignaling(); |
| 2751 | caller()->CreateDataChannel(); |
| 2752 | caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer); |
| 2753 | caller()->CreateAndSetAndSignalOffer(); |
| 2754 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2755 | ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout); |
| 2756 | EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2757 | EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout); |
| 2758 | |
| 2759 | // Ensure data can be sent in both directions. |
| 2760 | std::string data = "hello world"; |
| 2761 | caller()->data_channel()->Send(DataBuffer(data)); |
| 2762 | EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(), |
| 2763 | kDefaultTimeout); |
| 2764 | callee()->data_channel()->Send(DataBuffer(data)); |
| 2765 | EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(), |
| 2766 | kDefaultTimeout); |
| 2767 | } |
| 2768 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2769 | #endif // HAVE_SCTP |
| 2770 | |
| 2771 | // Test that the ICE connection and gathering states eventually reach |
| 2772 | // "complete". |
| 2773 | TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) { |
| 2774 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2775 | ConnectFakeSignaling(); |
| 2776 | // Do normal offer/answer. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2777 | caller()->AddAudioVideoTracks(); |
| 2778 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2779 | caller()->CreateAndSetAndSignalOffer(); |
| 2780 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2781 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, |
| 2782 | caller()->ice_gathering_state(), kMaxWaitForFramesMs); |
| 2783 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, |
| 2784 | callee()->ice_gathering_state(), kMaxWaitForFramesMs); |
| 2785 | // After the best candidate pair is selected and all candidates are signaled, |
| 2786 | // the ICE connection state should reach "complete". |
| 2787 | // TODO(deadbeef): Currently, the ICE "controlled" agent (the |
| 2788 | // answerer/"callee" by default) only reaches "connected". When this is |
| 2789 | // fixed, this test should be updated. |
| 2790 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, |
| 2791 | caller()->ice_connection_state(), kDefaultTimeout); |
| 2792 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, |
| 2793 | callee()->ice_connection_state(), kDefaultTimeout); |
| 2794 | } |
| 2795 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2796 | // Test that firewalling the ICE connection causes the clients to identify the |
| 2797 | // disconnected state and then removing the firewall causes them to reconnect. |
| 2798 | class PeerConnectionIntegrationIceStatesTest |
| 2799 | : public PeerConnectionIntegrationTest, |
| 2800 | public ::testing::WithParamInterface<std::tuple<std::string, uint32_t>> { |
| 2801 | protected: |
| 2802 | PeerConnectionIntegrationIceStatesTest() { |
| 2803 | port_allocator_flags_ = std::get<1>(GetParam()); |
| 2804 | } |
| 2805 | |
| 2806 | void StartStunServer(const SocketAddress& server_address) { |
| 2807 | stun_server_.reset( |
| 2808 | cricket::TestStunServer::Create(network_thread(), server_address)); |
| 2809 | } |
| 2810 | |
| 2811 | bool TestIPv6() { |
| 2812 | return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6); |
| 2813 | } |
| 2814 | |
| 2815 | void SetPortAllocatorFlags() { |
| 2816 | caller()->port_allocator()->set_flags(port_allocator_flags_); |
| 2817 | callee()->port_allocator()->set_flags(port_allocator_flags_); |
| 2818 | } |
| 2819 | |
| 2820 | std::vector<SocketAddress> CallerAddresses() { |
| 2821 | std::vector<SocketAddress> addresses; |
| 2822 | addresses.push_back(SocketAddress("1.1.1.1", 0)); |
| 2823 | if (TestIPv6()) { |
| 2824 | addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0)); |
| 2825 | } |
| 2826 | return addresses; |
| 2827 | } |
| 2828 | |
| 2829 | std::vector<SocketAddress> CalleeAddresses() { |
| 2830 | std::vector<SocketAddress> addresses; |
| 2831 | addresses.push_back(SocketAddress("2.2.2.2", 0)); |
| 2832 | if (TestIPv6()) { |
| 2833 | addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0)); |
| 2834 | } |
| 2835 | return addresses; |
| 2836 | } |
| 2837 | |
| 2838 | void SetUpNetworkInterfaces() { |
| 2839 | // Remove the default interfaces added by the test infrastructure. |
| 2840 | caller()->network()->RemoveInterface(kDefaultLocalAddress); |
| 2841 | callee()->network()->RemoveInterface(kDefaultLocalAddress); |
| 2842 | |
| 2843 | // Add network addresses for test. |
| 2844 | for (const auto& caller_address : CallerAddresses()) { |
| 2845 | caller()->network()->AddInterface(caller_address); |
| 2846 | } |
| 2847 | for (const auto& callee_address : CalleeAddresses()) { |
| 2848 | callee()->network()->AddInterface(callee_address); |
| 2849 | } |
| 2850 | } |
| 2851 | |
| 2852 | private: |
| 2853 | uint32_t port_allocator_flags_; |
| 2854 | std::unique_ptr<cricket::TestStunServer> stun_server_; |
| 2855 | }; |
| 2856 | |
| 2857 | // Tests that the PeerConnection goes through all the ICE gathering/connection |
| 2858 | // states over the duration of the call. This includes Disconnected and Failed |
| 2859 | // states, induced by putting a firewall between the peers and waiting for them |
| 2860 | // to time out. |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2861 | TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) { |
| 2862 | // TODO(bugs.webrtc.org/8295): When using a ScopedFakeClock, this test will |
| 2863 | // sometimes hit a DCHECK in platform_thread.cc about the PacerThread being |
| 2864 | // too busy. For now, revert to running without a fake clock. |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2865 | |
| 2866 | const SocketAddress kStunServerAddress = |
| 2867 | SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT); |
| 2868 | StartStunServer(kStunServerAddress); |
| 2869 | |
| 2870 | PeerConnectionInterface::RTCConfiguration config; |
| 2871 | PeerConnectionInterface::IceServer ice_stun_server; |
| 2872 | ice_stun_server.urls.push_back( |
| 2873 | "stun:" + kStunServerAddress.HostAsURIString() + ":" + |
| 2874 | kStunServerAddress.PortAsString()); |
| 2875 | config.servers.push_back(ice_stun_server); |
| 2876 | |
| 2877 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); |
| 2878 | ConnectFakeSignaling(); |
| 2879 | SetPortAllocatorFlags(); |
| 2880 | SetUpNetworkInterfaces(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2881 | caller()->AddAudioVideoTracks(); |
| 2882 | callee()->AddAudioVideoTracks(); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2883 | |
| 2884 | // Initial state before anything happens. |
| 2885 | ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew, |
| 2886 | caller()->ice_gathering_state()); |
| 2887 | ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew, |
| 2888 | caller()->ice_connection_state()); |
| 2889 | |
| 2890 | // Start the call by creating the offer, setting it as the local description, |
| 2891 | // then sending it to the peer who will respond with an answer. This happens |
| 2892 | // asynchronously so that we can watch the states as it runs in the |
| 2893 | // background. |
| 2894 | caller()->CreateAndSetAndSignalOffer(); |
| 2895 | |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2896 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, |
| 2897 | caller()->ice_connection_state(), kDefaultTimeout); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2898 | |
| 2899 | // Verify that the observer was notified of the intermediate transitions. |
| 2900 | EXPECT_THAT(caller()->ice_connection_state_history(), |
| 2901 | ElementsAre(PeerConnectionInterface::kIceConnectionChecking, |
| 2902 | PeerConnectionInterface::kIceConnectionConnected, |
| 2903 | PeerConnectionInterface::kIceConnectionCompleted)); |
| 2904 | EXPECT_THAT(caller()->ice_gathering_state_history(), |
| 2905 | ElementsAre(PeerConnectionInterface::kIceGatheringGathering, |
| 2906 | PeerConnectionInterface::kIceGatheringComplete)); |
| 2907 | |
| 2908 | // Block connections to/from the caller and wait for ICE to become |
| 2909 | // disconnected. |
| 2910 | for (const auto& caller_address : CallerAddresses()) { |
| 2911 | firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address); |
| 2912 | } |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2913 | RTC_LOG(LS_INFO) << "Firewall rules applied"; |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2914 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected, |
| 2915 | caller()->ice_connection_state(), kDefaultTimeout); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2916 | |
| 2917 | // Let ICE re-establish by removing the firewall rules. |
| 2918 | firewall()->ClearRules(); |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2919 | RTC_LOG(LS_INFO) << "Firewall rules cleared"; |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2920 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, |
| 2921 | caller()->ice_connection_state(), kDefaultTimeout); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2922 | |
| 2923 | // According to RFC7675, if there is no response within 30 seconds then the |
| 2924 | // peer should consider the other side to have rejected the connection. This |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2925 | // is signaled by the state transitioning to "failed". |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2926 | constexpr int kConsentTimeout = 30000; |
| 2927 | for (const auto& caller_address : CallerAddresses()) { |
| 2928 | firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address); |
| 2929 | } |
Mirko Bonadei | 675513b | 2017-11-09 11:09:25 +0100 | [diff] [blame] | 2930 | RTC_LOG(LS_INFO) << "Firewall rules applied again"; |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 2931 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed, |
| 2932 | caller()->ice_connection_state(), kConsentTimeout); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2933 | } |
| 2934 | |
| 2935 | // Tests that the best connection is set to the appropriate IPv4/IPv6 connection |
| 2936 | // and that the statistics in the metric observers are updated correctly. |
| 2937 | TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) { |
| 2938 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2939 | ConnectFakeSignaling(); |
| 2940 | SetPortAllocatorFlags(); |
| 2941 | SetUpNetworkInterfaces(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2942 | caller()->AddAudioVideoTracks(); |
| 2943 | callee()->AddAudioVideoTracks(); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 2944 | |
| 2945 | rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer( |
| 2946 | new rtc::RefCountedObject<webrtc::FakeMetricsObserver>()); |
| 2947 | caller()->pc()->RegisterUMAObserver(metrics_observer.get()); |
| 2948 | |
| 2949 | caller()->CreateAndSetAndSignalOffer(); |
| 2950 | |
| 2951 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 2952 | |
| 2953 | const int num_best_ipv4 = metrics_observer->GetEnumCounter( |
| 2954 | webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv4); |
| 2955 | const int num_best_ipv6 = metrics_observer->GetEnumCounter( |
| 2956 | webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv6); |
| 2957 | if (TestIPv6()) { |
| 2958 | // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4 |
| 2959 | // connection. |
| 2960 | EXPECT_EQ(0u, num_best_ipv4); |
| 2961 | EXPECT_EQ(1u, num_best_ipv6); |
| 2962 | } else { |
| 2963 | EXPECT_EQ(1u, num_best_ipv4); |
| 2964 | EXPECT_EQ(0u, num_best_ipv6); |
| 2965 | } |
| 2966 | |
| 2967 | EXPECT_EQ(0u, metrics_observer->GetEnumCounter( |
| 2968 | webrtc::kEnumCounterIceCandidatePairTypeUdp, |
| 2969 | webrtc::kIceCandidatePairHostHost)); |
| 2970 | EXPECT_EQ(1u, metrics_observer->GetEnumCounter( |
| 2971 | webrtc::kEnumCounterIceCandidatePairTypeUdp, |
| 2972 | webrtc::kIceCandidatePairHostPublicHostPublic)); |
| 2973 | } |
| 2974 | |
| 2975 | constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP | |
| 2976 | cricket::PORTALLOCATOR_DISABLE_STUN | |
| 2977 | cricket::PORTALLOCATOR_DISABLE_RELAY; |
| 2978 | constexpr uint32_t kFlagsIPv6NoStun = |
| 2979 | cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN | |
| 2980 | cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY; |
| 2981 | constexpr uint32_t kFlagsIPv4Stun = |
| 2982 | cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY; |
| 2983 | |
| 2984 | INSTANTIATE_TEST_CASE_P(PeerConnectionIntegrationTest, |
| 2985 | PeerConnectionIntegrationIceStatesTest, |
| 2986 | Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun), |
| 2987 | std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun), |
| 2988 | std::make_pair("IPv4 with STUN", |
| 2989 | kFlagsIPv4Stun))); |
| 2990 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 2991 | // This test sets up a call between two parties with audio and video. |
| 2992 | // During the call, the caller restarts ICE and the test verifies that |
| 2993 | // new ICE candidates are generated and audio and video still can flow, and the |
| 2994 | // ICE state reaches completed again. |
| 2995 | TEST_F(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) { |
| 2996 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 2997 | ConnectFakeSignaling(); |
| 2998 | // Do normal offer/answer and wait for ICE to complete. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 2999 | caller()->AddAudioVideoTracks(); |
| 3000 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3001 | caller()->CreateAndSetAndSignalOffer(); |
| 3002 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3003 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, |
| 3004 | caller()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3005 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, |
| 3006 | callee()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3007 | |
| 3008 | // To verify that the ICE restart actually occurs, get |
| 3009 | // ufrag/password/candidates before and after restart. |
| 3010 | // Create an SDP string of the first audio candidate for both clients. |
| 3011 | const webrtc::IceCandidateCollection* audio_candidates_caller = |
| 3012 | caller()->pc()->local_description()->candidates(0); |
| 3013 | const webrtc::IceCandidateCollection* audio_candidates_callee = |
| 3014 | callee()->pc()->local_description()->candidates(0); |
| 3015 | ASSERT_GT(audio_candidates_caller->count(), 0u); |
| 3016 | ASSERT_GT(audio_candidates_callee->count(), 0u); |
| 3017 | std::string caller_candidate_pre_restart; |
| 3018 | ASSERT_TRUE( |
| 3019 | audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart)); |
| 3020 | std::string callee_candidate_pre_restart; |
| 3021 | ASSERT_TRUE( |
| 3022 | audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart)); |
| 3023 | const cricket::SessionDescription* desc = |
| 3024 | caller()->pc()->local_description()->description(); |
| 3025 | std::string caller_ufrag_pre_restart = |
| 3026 | desc->transport_infos()[0].description.ice_ufrag; |
| 3027 | desc = callee()->pc()->local_description()->description(); |
| 3028 | std::string callee_ufrag_pre_restart = |
| 3029 | desc->transport_infos()[0].description.ice_ufrag; |
| 3030 | |
| 3031 | // Have the caller initiate an ICE restart. |
| 3032 | caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); |
| 3033 | caller()->CreateAndSetAndSignalOffer(); |
| 3034 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3035 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, |
| 3036 | caller()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3037 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, |
| 3038 | callee()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3039 | |
| 3040 | // Grab the ufrags/candidates again. |
| 3041 | audio_candidates_caller = caller()->pc()->local_description()->candidates(0); |
| 3042 | audio_candidates_callee = callee()->pc()->local_description()->candidates(0); |
| 3043 | ASSERT_GT(audio_candidates_caller->count(), 0u); |
| 3044 | ASSERT_GT(audio_candidates_callee->count(), 0u); |
| 3045 | std::string caller_candidate_post_restart; |
| 3046 | ASSERT_TRUE( |
| 3047 | audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart)); |
| 3048 | std::string callee_candidate_post_restart; |
| 3049 | ASSERT_TRUE( |
| 3050 | audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart)); |
| 3051 | desc = caller()->pc()->local_description()->description(); |
| 3052 | std::string caller_ufrag_post_restart = |
| 3053 | desc->transport_infos()[0].description.ice_ufrag; |
| 3054 | desc = callee()->pc()->local_description()->description(); |
| 3055 | std::string callee_ufrag_post_restart = |
| 3056 | desc->transport_infos()[0].description.ice_ufrag; |
| 3057 | // Sanity check that an ICE restart was actually negotiated in SDP. |
| 3058 | ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart); |
| 3059 | ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart); |
| 3060 | ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart); |
| 3061 | ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart); |
| 3062 | |
| 3063 | // Ensure that additional frames are received after the ICE restart. |
| 3064 | ExpectNewFramesReceivedWithWait( |
| 3065 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3066 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3067 | kMaxWaitForFramesMs); |
| 3068 | } |
| 3069 | |
| 3070 | // Verify that audio/video can be received end-to-end when ICE renomination is |
| 3071 | // enabled. |
| 3072 | TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) { |
| 3073 | PeerConnectionInterface::RTCConfiguration config; |
| 3074 | config.enable_ice_renomination = true; |
| 3075 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); |
| 3076 | ConnectFakeSignaling(); |
| 3077 | // Do normal offer/answer and wait for some frames to be received in each |
| 3078 | // direction. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3079 | caller()->AddAudioVideoTracks(); |
| 3080 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3081 | caller()->CreateAndSetAndSignalOffer(); |
| 3082 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3083 | // Sanity check that ICE renomination was actually negotiated. |
| 3084 | const cricket::SessionDescription* desc = |
| 3085 | caller()->pc()->local_description()->description(); |
| 3086 | for (const cricket::TransportInfo& info : desc->transport_infos()) { |
deadbeef | 30952b4 | 2017-04-21 02:41:29 -0700 | [diff] [blame] | 3087 | ASSERT_NE( |
| 3088 | info.description.transport_options.end(), |
| 3089 | std::find(info.description.transport_options.begin(), |
| 3090 | info.description.transport_options.end(), "renomination")); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3091 | } |
| 3092 | desc = callee()->pc()->local_description()->description(); |
| 3093 | for (const cricket::TransportInfo& info : desc->transport_infos()) { |
deadbeef | 30952b4 | 2017-04-21 02:41:29 -0700 | [diff] [blame] | 3094 | ASSERT_NE( |
| 3095 | info.description.transport_options.end(), |
| 3096 | std::find(info.description.transport_options.begin(), |
| 3097 | info.description.transport_options.end(), "renomination")); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3098 | } |
| 3099 | ExpectNewFramesReceivedWithWait( |
| 3100 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3101 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3102 | kMaxWaitForFramesMs); |
| 3103 | } |
| 3104 | |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 3105 | // With a max bundle policy and RTCP muxing, adding a new media description to |
| 3106 | // the connection should not affect ICE at all because the new media will use |
| 3107 | // the existing connection. |
| 3108 | TEST_F(PeerConnectionIntegrationTest, |
Steve Anton | 83119dd | 2017-11-10 16:19:52 -0800 | [diff] [blame] | 3109 | AddMediaToConnectedBundleDoesNotRestartIce) { |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 3110 | PeerConnectionInterface::RTCConfiguration config; |
| 3111 | config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle; |
| 3112 | config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire; |
| 3113 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig( |
| 3114 | config, PeerConnectionInterface::RTCConfiguration())); |
| 3115 | ConnectFakeSignaling(); |
| 3116 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3117 | caller()->AddAudioTrack(); |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 3118 | caller()->CreateAndSetAndSignalOffer(); |
| 3119 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
Steve Anton | ff52f1b | 2017-10-26 12:24:50 -0700 | [diff] [blame] | 3120 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted, |
| 3121 | caller()->ice_connection_state(), kDefaultTimeout); |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 3122 | |
| 3123 | caller()->clear_ice_connection_state_history(); |
| 3124 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3125 | caller()->AddVideoTrack(); |
Steve Anton | 6f25b09 | 2017-10-23 09:39:20 -0700 | [diff] [blame] | 3126 | caller()->CreateAndSetAndSignalOffer(); |
| 3127 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3128 | |
| 3129 | EXPECT_EQ(0u, caller()->ice_connection_state_history().size()); |
| 3130 | } |
| 3131 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3132 | // This test sets up a call between two parties with audio and video. It then |
| 3133 | // renegotiates setting the video m-line to "port 0", then later renegotiates |
| 3134 | // again, enabling video. |
| 3135 | TEST_F(PeerConnectionIntegrationTest, |
| 3136 | VideoFlowsAfterMediaSectionIsRejectedAndRecycled) { |
| 3137 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3138 | ConnectFakeSignaling(); |
| 3139 | |
| 3140 | // Do initial negotiation, only sending media from the caller. Will result in |
| 3141 | // video and audio recvonly "m=" sections. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3142 | caller()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3143 | caller()->CreateAndSetAndSignalOffer(); |
| 3144 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3145 | |
| 3146 | // Negotiate again, disabling the video "m=" section (the callee will set the |
| 3147 | // port to 0 due to offer_to_receive_video = 0). |
| 3148 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 3149 | options.offer_to_receive_video = 0; |
| 3150 | callee()->SetOfferAnswerOptions(options); |
| 3151 | caller()->CreateAndSetAndSignalOffer(); |
| 3152 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3153 | // Sanity check that video "m=" section was actually rejected. |
| 3154 | const ContentInfo* answer_video_content = cricket::GetFirstVideoContent( |
| 3155 | callee()->pc()->local_description()->description()); |
| 3156 | ASSERT_NE(nullptr, answer_video_content); |
| 3157 | ASSERT_TRUE(answer_video_content->rejected); |
| 3158 | |
| 3159 | // Enable video and do negotiation again, making sure video is received |
| 3160 | // end-to-end, also adding media stream to callee. |
| 3161 | options.offer_to_receive_video = 1; |
| 3162 | callee()->SetOfferAnswerOptions(options); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3163 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3164 | caller()->CreateAndSetAndSignalOffer(); |
| 3165 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3166 | // Verify the caller receives frames from the newly added stream, and the |
| 3167 | // callee receives additional frames from the re-enabled video m= section. |
| 3168 | ExpectNewFramesReceivedWithWait( |
| 3169 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3170 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3171 | kMaxWaitForFramesMs); |
| 3172 | } |
| 3173 | |
| 3174 | // This test sets up a Jsep call between two parties with external |
| 3175 | // VideoDecoderFactory. |
| 3176 | // TODO(holmer): Disabled due to sometimes crashing on buildbots. |
| 3177 | // See issue webrtc/2378. |
| 3178 | TEST_F(PeerConnectionIntegrationTest, |
| 3179 | DISABLED_EndToEndCallWithVideoDecoderFactory) { |
| 3180 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3181 | EnableVideoDecoderFactory(); |
| 3182 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3183 | caller()->AddAudioVideoTracks(); |
| 3184 | callee()->AddAudioVideoTracks(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3185 | caller()->CreateAndSetAndSignalOffer(); |
| 3186 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3187 | ExpectNewFramesReceivedWithWait( |
| 3188 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3189 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3190 | kMaxWaitForFramesMs); |
| 3191 | } |
| 3192 | |
| 3193 | // This tests that if we negotiate after calling CreateSender but before we |
| 3194 | // have a track, then set a track later, frames from the newly-set track are |
| 3195 | // received end-to-end. |
| 3196 | // TODO(deadbeef): Change this test to use AddTransceiver, once that's |
| 3197 | // implemented. |
| 3198 | TEST_F(PeerConnectionIntegrationTest, |
| 3199 | MediaFlowsAfterEarlyWarmupWithCreateSender) { |
| 3200 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3201 | ConnectFakeSignaling(); |
| 3202 | auto caller_audio_sender = |
| 3203 | caller()->pc()->CreateSender("audio", "caller_stream"); |
| 3204 | auto caller_video_sender = |
| 3205 | caller()->pc()->CreateSender("video", "caller_stream"); |
| 3206 | auto callee_audio_sender = |
| 3207 | callee()->pc()->CreateSender("audio", "callee_stream"); |
| 3208 | auto callee_video_sender = |
| 3209 | callee()->pc()->CreateSender("video", "callee_stream"); |
| 3210 | caller()->CreateAndSetAndSignalOffer(); |
| 3211 | ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); |
| 3212 | // Wait for ICE to complete, without any tracks being set. |
| 3213 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, |
| 3214 | caller()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3215 | EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, |
| 3216 | callee()->ice_connection_state(), kMaxWaitForFramesMs); |
| 3217 | // Now set the tracks, and expect frames to immediately start flowing. |
| 3218 | EXPECT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack())); |
| 3219 | EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack())); |
| 3220 | EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack())); |
| 3221 | EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack())); |
| 3222 | ExpectNewFramesReceivedWithWait( |
| 3223 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3224 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3225 | kMaxWaitForFramesMs); |
| 3226 | } |
| 3227 | |
| 3228 | // This test verifies that a remote video track can be added via AddStream, |
| 3229 | // and sent end-to-end. For this particular test, it's simply echoed back |
| 3230 | // from the caller to the callee, rather than being forwarded to a third |
| 3231 | // PeerConnection. |
| 3232 | TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) { |
| 3233 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3234 | ConnectFakeSignaling(); |
| 3235 | // Just send a video track from the caller. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3236 | caller()->AddVideoTrack(); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3237 | caller()->CreateAndSetAndSignalOffer(); |
| 3238 | ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); |
| 3239 | ASSERT_EQ(1, callee()->remote_streams()->count()); |
| 3240 | |
| 3241 | // Echo the stream back, and do a new offer/anwer (initiated by callee this |
| 3242 | // time). |
| 3243 | callee()->pc()->AddStream(callee()->remote_streams()->at(0)); |
| 3244 | callee()->CreateAndSetAndSignalOffer(); |
| 3245 | ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); |
| 3246 | |
| 3247 | int expected_caller_received_video_frames = kDefaultExpectedVideoFrameCount; |
| 3248 | ExpectNewFramesReceivedWithWait(0, expected_caller_received_video_frames, 0, |
| 3249 | 0, kMaxWaitForFramesMs); |
| 3250 | } |
| 3251 | |
| 3252 | // Test that we achieve the expected end-to-end connection time, using a |
| 3253 | // fake clock and simulated latency on the media and signaling paths. |
| 3254 | // We use a TURN<->TURN connection because this is usually the quickest to |
| 3255 | // set up initially, especially when we're confident the connection will work |
| 3256 | // and can start sending media before we get a STUN response. |
| 3257 | // |
| 3258 | // With various optimizations enabled, here are the network delays we expect to |
| 3259 | // be on the critical path: |
| 3260 | // 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then |
| 3261 | // signaling answer (with DTLS fingerprint). |
| 3262 | // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when |
| 3263 | // using TURN<->TURN pair, and DTLS exchange is 4 packets, |
| 3264 | // the first of which should have arrived before the answer. |
| 3265 | TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) { |
| 3266 | rtc::ScopedFakeClock fake_clock; |
| 3267 | // Some things use a time of "0" as a special value, so we need to start out |
| 3268 | // the fake clock at a nonzero time. |
| 3269 | // TODO(deadbeef): Fix this. |
| 3270 | fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1)); |
| 3271 | |
| 3272 | static constexpr int media_hop_delay_ms = 50; |
| 3273 | static constexpr int signaling_trip_delay_ms = 500; |
| 3274 | // For explanation of these values, see comment above. |
| 3275 | static constexpr int required_media_hops = 9; |
| 3276 | static constexpr int required_signaling_trips = 2; |
| 3277 | // For internal delays (such as posting an event asychronously). |
| 3278 | static constexpr int allowed_internal_delay_ms = 20; |
| 3279 | static constexpr int total_connection_time_ms = |
| 3280 | media_hop_delay_ms * required_media_hops + |
| 3281 | signaling_trip_delay_ms * required_signaling_trips + |
| 3282 | allowed_internal_delay_ms; |
| 3283 | |
| 3284 | static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0", |
| 3285 | 3478}; |
| 3286 | static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1", |
| 3287 | 0}; |
| 3288 | static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0", |
| 3289 | 3478}; |
| 3290 | static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1", |
| 3291 | 0}; |
| 3292 | cricket::TestTurnServer turn_server_1(network_thread(), |
| 3293 | turn_server_1_internal_address, |
| 3294 | turn_server_1_external_address); |
| 3295 | cricket::TestTurnServer turn_server_2(network_thread(), |
| 3296 | turn_server_2_internal_address, |
| 3297 | turn_server_2_external_address); |
Jonas Oreland | bdcee28 | 2017-10-10 14:01:40 +0200 | [diff] [blame] | 3298 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3299 | // Bypass permission check on received packets so media can be sent before |
| 3300 | // the candidate is signaled. |
| 3301 | turn_server_1.set_enable_permission_checks(false); |
| 3302 | turn_server_2.set_enable_permission_checks(false); |
| 3303 | |
| 3304 | PeerConnectionInterface::RTCConfiguration client_1_config; |
| 3305 | webrtc::PeerConnectionInterface::IceServer ice_server_1; |
| 3306 | ice_server_1.urls.push_back("turn:88.88.88.0:3478"); |
| 3307 | ice_server_1.username = "test"; |
| 3308 | ice_server_1.password = "test"; |
| 3309 | client_1_config.servers.push_back(ice_server_1); |
| 3310 | client_1_config.type = webrtc::PeerConnectionInterface::kRelay; |
| 3311 | client_1_config.presume_writable_when_fully_relayed = true; |
| 3312 | |
| 3313 | PeerConnectionInterface::RTCConfiguration client_2_config; |
| 3314 | webrtc::PeerConnectionInterface::IceServer ice_server_2; |
| 3315 | ice_server_2.urls.push_back("turn:99.99.99.0:3478"); |
| 3316 | ice_server_2.username = "test"; |
| 3317 | ice_server_2.password = "test"; |
| 3318 | client_2_config.servers.push_back(ice_server_2); |
| 3319 | client_2_config.type = webrtc::PeerConnectionInterface::kRelay; |
| 3320 | client_2_config.presume_writable_when_fully_relayed = true; |
| 3321 | |
| 3322 | ASSERT_TRUE( |
| 3323 | CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config)); |
| 3324 | // Set up the simulated delays. |
| 3325 | SetSignalingDelayMs(signaling_trip_delay_ms); |
| 3326 | ConnectFakeSignaling(); |
| 3327 | virtual_socket_server()->set_delay_mean(media_hop_delay_ms); |
| 3328 | virtual_socket_server()->UpdateDelayDistribution(); |
| 3329 | |
| 3330 | // Set "offer to receive audio/video" without adding any tracks, so we just |
| 3331 | // set up ICE/DTLS with no media. |
| 3332 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 3333 | options.offer_to_receive_audio = 1; |
| 3334 | options.offer_to_receive_video = 1; |
| 3335 | caller()->SetOfferAnswerOptions(options); |
| 3336 | caller()->CreateAndSetAndSignalOffer(); |
deadbeef | 7145280 | 2017-05-07 17:21:01 -0700 | [diff] [blame] | 3337 | EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms, |
| 3338 | fake_clock); |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3339 | // Need to free the clients here since they're using things we created on |
| 3340 | // the stack. |
| 3341 | delete SetCallerPcWrapperAndReturnCurrent(nullptr); |
| 3342 | delete SetCalleePcWrapperAndReturnCurrent(nullptr); |
| 3343 | } |
| 3344 | |
Jonas Oreland | bdcee28 | 2017-10-10 14:01:40 +0200 | [diff] [blame] | 3345 | // Verify that a TurnCustomizer passed in through RTCConfiguration |
| 3346 | // is actually used by the underlying TURN candidate pair. |
| 3347 | // Note that turnport_unittest.cc contains more detailed, lower-level tests. |
| 3348 | TEST_F(PeerConnectionIntegrationTest, \ |
| 3349 | TurnCustomizerUsedForTurnConnections) { |
| 3350 | static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0", |
| 3351 | 3478}; |
| 3352 | static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1", |
| 3353 | 0}; |
| 3354 | static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0", |
| 3355 | 3478}; |
| 3356 | static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1", |
| 3357 | 0}; |
| 3358 | cricket::TestTurnServer turn_server_1(network_thread(), |
| 3359 | turn_server_1_internal_address, |
| 3360 | turn_server_1_external_address); |
| 3361 | cricket::TestTurnServer turn_server_2(network_thread(), |
| 3362 | turn_server_2_internal_address, |
| 3363 | turn_server_2_external_address); |
| 3364 | |
| 3365 | PeerConnectionInterface::RTCConfiguration client_1_config; |
| 3366 | webrtc::PeerConnectionInterface::IceServer ice_server_1; |
| 3367 | ice_server_1.urls.push_back("turn:88.88.88.0:3478"); |
| 3368 | ice_server_1.username = "test"; |
| 3369 | ice_server_1.password = "test"; |
| 3370 | client_1_config.servers.push_back(ice_server_1); |
| 3371 | client_1_config.type = webrtc::PeerConnectionInterface::kRelay; |
| 3372 | auto customizer1 = rtc::MakeUnique<cricket::TestTurnCustomizer>(); |
| 3373 | client_1_config.turn_customizer = customizer1.get(); |
| 3374 | |
| 3375 | PeerConnectionInterface::RTCConfiguration client_2_config; |
| 3376 | webrtc::PeerConnectionInterface::IceServer ice_server_2; |
| 3377 | ice_server_2.urls.push_back("turn:99.99.99.0:3478"); |
| 3378 | ice_server_2.username = "test"; |
| 3379 | ice_server_2.password = "test"; |
| 3380 | client_2_config.servers.push_back(ice_server_2); |
| 3381 | client_2_config.type = webrtc::PeerConnectionInterface::kRelay; |
| 3382 | auto customizer2 = rtc::MakeUnique<cricket::TestTurnCustomizer>(); |
| 3383 | client_2_config.turn_customizer = customizer2.get(); |
| 3384 | |
| 3385 | ASSERT_TRUE( |
| 3386 | CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config)); |
| 3387 | ConnectFakeSignaling(); |
| 3388 | |
| 3389 | // Set "offer to receive audio/video" without adding any tracks, so we just |
| 3390 | // set up ICE/DTLS with no media. |
| 3391 | PeerConnectionInterface::RTCOfferAnswerOptions options; |
| 3392 | options.offer_to_receive_audio = 1; |
| 3393 | options.offer_to_receive_video = 1; |
| 3394 | caller()->SetOfferAnswerOptions(options); |
| 3395 | caller()->CreateAndSetAndSignalOffer(); |
| 3396 | ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| 3397 | |
| 3398 | EXPECT_GT(customizer1->allow_channel_data_cnt_, 0u); |
| 3399 | EXPECT_GT(customizer1->modify_cnt_, 0u); |
| 3400 | |
| 3401 | EXPECT_GT(customizer2->allow_channel_data_cnt_, 0u); |
| 3402 | EXPECT_GT(customizer2->modify_cnt_, 0u); |
| 3403 | |
| 3404 | // Need to free the clients here since they're using things we created on |
| 3405 | // the stack. |
| 3406 | delete SetCallerPcWrapperAndReturnCurrent(nullptr); |
| 3407 | delete SetCalleePcWrapperAndReturnCurrent(nullptr); |
| 3408 | } |
| 3409 | |
deadbeef | c964d0b | 2017-04-03 10:03:35 -0700 | [diff] [blame] | 3410 | // Test that audio and video flow end-to-end when codec names don't use the |
| 3411 | // expected casing, given that they're supposed to be case insensitive. To test |
| 3412 | // this, all but one codec is removed from each media description, and its |
| 3413 | // casing is changed. |
| 3414 | // |
| 3415 | // In the past, this has regressed and caused crashes/black video, due to the |
| 3416 | // fact that code at some layers was doing case-insensitive comparisons and |
| 3417 | // code at other layers was not. |
| 3418 | TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) { |
| 3419 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3420 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3421 | caller()->AddAudioVideoTracks(); |
| 3422 | callee()->AddAudioVideoTracks(); |
deadbeef | c964d0b | 2017-04-03 10:03:35 -0700 | [diff] [blame] | 3423 | |
| 3424 | // Remove all but one audio/video codec (opus and VP8), and change the |
| 3425 | // casing of the caller's generated offer. |
| 3426 | caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) { |
| 3427 | cricket::AudioContentDescription* audio = |
| 3428 | GetFirstAudioContentDescription(description); |
| 3429 | ASSERT_NE(nullptr, audio); |
| 3430 | auto audio_codecs = audio->codecs(); |
| 3431 | audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(), |
| 3432 | [](const cricket::AudioCodec& codec) { |
| 3433 | return codec.name != "opus"; |
| 3434 | }), |
| 3435 | audio_codecs.end()); |
| 3436 | ASSERT_EQ(1u, audio_codecs.size()); |
| 3437 | audio_codecs[0].name = "OpUs"; |
| 3438 | audio->set_codecs(audio_codecs); |
| 3439 | |
| 3440 | cricket::VideoContentDescription* video = |
| 3441 | GetFirstVideoContentDescription(description); |
| 3442 | ASSERT_NE(nullptr, video); |
| 3443 | auto video_codecs = video->codecs(); |
| 3444 | video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(), |
| 3445 | [](const cricket::VideoCodec& codec) { |
| 3446 | return codec.name != "VP8"; |
| 3447 | }), |
| 3448 | video_codecs.end()); |
| 3449 | ASSERT_EQ(1u, video_codecs.size()); |
| 3450 | video_codecs[0].name = "vP8"; |
| 3451 | video->set_codecs(video_codecs); |
| 3452 | }); |
| 3453 | |
| 3454 | caller()->CreateAndSetAndSignalOffer(); |
| 3455 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3456 | |
| 3457 | // Verify frames are still received end-to-end. |
| 3458 | ExpectNewFramesReceivedWithWait( |
| 3459 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3460 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3461 | kMaxWaitForFramesMs); |
| 3462 | } |
| 3463 | |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 3464 | TEST_F(PeerConnectionIntegrationTest, GetSources) { |
| 3465 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3466 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3467 | caller()->AddAudioTrack(); |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 3468 | caller()->CreateAndSetAndSignalOffer(); |
| 3469 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
deadbeef | d8ad788 | 2017-04-18 16:01:17 -0700 | [diff] [blame] | 3470 | // Wait for one audio frame to be received by the callee. |
hbos | 8d609f6 | 2017-04-10 07:39:05 -0700 | [diff] [blame] | 3471 | ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs); |
| 3472 | ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u); |
| 3473 | auto receiver = callee()->pc()->GetReceivers()[0]; |
| 3474 | ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO); |
| 3475 | |
| 3476 | auto contributing_sources = receiver->GetSources(); |
| 3477 | ASSERT_GT(receiver->GetParameters().encodings.size(), 0u); |
| 3478 | EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc, |
| 3479 | contributing_sources[0].source_id()); |
| 3480 | } |
| 3481 | |
deadbeef | 2f425aa | 2017-04-14 10:41:32 -0700 | [diff] [blame] | 3482 | // Test that if a track is removed and added again with a different stream ID, |
| 3483 | // the new stream ID is successfully communicated in SDP and media continues to |
| 3484 | // flow end-to-end. |
| 3485 | TEST_F(PeerConnectionIntegrationTest, RemoveAndAddTrackWithNewStreamId) { |
| 3486 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3487 | ConnectFakeSignaling(); |
| 3488 | |
| 3489 | rtc::scoped_refptr<MediaStreamInterface> stream_1 = |
| 3490 | caller()->pc_factory()->CreateLocalMediaStream("stream_1"); |
| 3491 | rtc::scoped_refptr<MediaStreamInterface> stream_2 = |
| 3492 | caller()->pc_factory()->CreateLocalMediaStream("stream_2"); |
| 3493 | |
| 3494 | // Add track using stream 1, do offer/answer. |
| 3495 | rtc::scoped_refptr<webrtc::AudioTrackInterface> track = |
| 3496 | caller()->CreateLocalAudioTrack(); |
| 3497 | rtc::scoped_refptr<webrtc::RtpSenderInterface> sender = |
| 3498 | caller()->pc()->AddTrack(track, {stream_1.get()}); |
| 3499 | caller()->CreateAndSetAndSignalOffer(); |
| 3500 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3501 | // Wait for one audio frame to be received by the callee. |
| 3502 | ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs); |
| 3503 | |
| 3504 | // Remove the sender, and create a new one with the new stream. |
| 3505 | caller()->pc()->RemoveTrack(sender); |
| 3506 | sender = caller()->pc()->AddTrack(track, {stream_2.get()}); |
| 3507 | caller()->CreateAndSetAndSignalOffer(); |
| 3508 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3509 | // Wait for additional audio frames to be received by the callee. |
| 3510 | ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0, |
| 3511 | kMaxWaitForFramesMs); |
| 3512 | } |
| 3513 | |
Elad Alon | 99c3fe5 | 2017-10-13 16:29:40 +0200 | [diff] [blame] | 3514 | TEST_F(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) { |
| 3515 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3516 | ConnectFakeSignaling(); |
| 3517 | |
| 3518 | auto output = rtc::MakeUnique<testing::NiceMock<MockRtcEventLogOutput>>(); |
| 3519 | ON_CALL(*output, IsActive()).WillByDefault(testing::Return(true)); |
| 3520 | ON_CALL(*output, Write(::testing::_)).WillByDefault(testing::Return(true)); |
| 3521 | EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1)); |
Bjorn Terelius | de93943 | 2017-11-20 17:38:14 +0100 | [diff] [blame] | 3522 | EXPECT_TRUE(caller()->pc()->StartRtcEventLog( |
| 3523 | std::move(output), webrtc::RtcEventLog::kImmediateOutput)); |
Elad Alon | 99c3fe5 | 2017-10-13 16:29:40 +0200 | [diff] [blame] | 3524 | |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3525 | caller()->AddAudioVideoTracks(); |
Elad Alon | 99c3fe5 | 2017-10-13 16:29:40 +0200 | [diff] [blame] | 3526 | caller()->CreateAndSetAndSignalOffer(); |
| 3527 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3528 | } |
| 3529 | |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 3530 | // Test that if candidates are only signaled by applying full session |
| 3531 | // descriptions (instead of using AddIceCandidate), the peers can connect to |
| 3532 | // each other and exchange media. |
| 3533 | TEST_F(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) { |
| 3534 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3535 | // Each side will signal the session descriptions but not candidates. |
| 3536 | ConnectFakeSignalingForSdpOnly(); |
| 3537 | |
| 3538 | // Add audio video track and exchange the initial offer/answer with media |
| 3539 | // information only. This will start ICE gathering on each side. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3540 | caller()->AddAudioVideoTracks(); |
| 3541 | callee()->AddAudioVideoTracks(); |
Steve Anton | ede9ca5 | 2017-10-16 13:04:27 -0700 | [diff] [blame] | 3542 | caller()->CreateAndSetAndSignalOffer(); |
| 3543 | |
| 3544 | // Wait for all candidates to be gathered on both the caller and callee. |
| 3545 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete, |
| 3546 | caller()->ice_gathering_state(), kDefaultTimeout); |
| 3547 | ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete, |
| 3548 | callee()->ice_gathering_state(), kDefaultTimeout); |
| 3549 | |
| 3550 | // The candidates will now be included in the session description, so |
| 3551 | // signaling them will start the ICE connection. |
| 3552 | caller()->CreateAndSetAndSignalOffer(); |
| 3553 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3554 | |
| 3555 | // Ensure that media flows in both directions. |
| 3556 | ExpectNewFramesReceivedWithWait( |
| 3557 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3558 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3559 | kMaxWaitForFramesMs); |
| 3560 | } |
| 3561 | |
henrika | 5f6bf24 | 2017-11-01 11:06:56 +0100 | [diff] [blame] | 3562 | // Test that SetAudioPlayout can be used to disable audio playout from the |
| 3563 | // start, then later enable it. This may be useful, for example, if the caller |
| 3564 | // needs to play a local ringtone until some event occurs, after which it |
| 3565 | // switches to playing the received audio. |
| 3566 | TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) { |
| 3567 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3568 | ConnectFakeSignaling(); |
| 3569 | |
| 3570 | // Set up audio-only call where audio playout is disabled on caller's side. |
| 3571 | caller()->pc()->SetAudioPlayout(false); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3572 | caller()->AddAudioTrack(); |
| 3573 | callee()->AddAudioTrack(); |
henrika | 5f6bf24 | 2017-11-01 11:06:56 +0100 | [diff] [blame] | 3574 | caller()->CreateAndSetAndSignalOffer(); |
| 3575 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3576 | |
| 3577 | // Pump messages for a second. |
| 3578 | WAIT(false, 1000); |
| 3579 | // Since audio playout is disabled, the caller shouldn't have received |
| 3580 | // anything (at the playout level, at least). |
| 3581 | EXPECT_EQ(0, caller()->audio_frames_received()); |
| 3582 | // As a sanity check, make sure the callee (for which playout isn't disabled) |
| 3583 | // did still see frames on its audio level. |
| 3584 | ASSERT_GT(callee()->audio_frames_received(), 0); |
| 3585 | |
| 3586 | // Enable playout again, and ensure audio starts flowing. |
| 3587 | caller()->pc()->SetAudioPlayout(true); |
| 3588 | ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0, |
| 3589 | kDefaultExpectedAudioFrameCount, 0, |
| 3590 | kMaxWaitForFramesMs); |
| 3591 | } |
| 3592 | |
| 3593 | double GetAudioEnergyStat(PeerConnectionWrapper* pc) { |
| 3594 | auto report = pc->NewGetStats(); |
| 3595 | auto track_stats_list = |
| 3596 | report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| 3597 | const webrtc::RTCMediaStreamTrackStats* remote_track_stats = nullptr; |
| 3598 | for (const auto* track_stats : track_stats_list) { |
| 3599 | if (track_stats->remote_source.is_defined() && |
| 3600 | *track_stats->remote_source) { |
| 3601 | remote_track_stats = track_stats; |
| 3602 | break; |
| 3603 | } |
| 3604 | } |
| 3605 | |
| 3606 | if (!remote_track_stats->total_audio_energy.is_defined()) { |
| 3607 | return 0.0; |
| 3608 | } |
| 3609 | return *remote_track_stats->total_audio_energy; |
| 3610 | } |
| 3611 | |
| 3612 | // Test that if audio playout is disabled via the SetAudioPlayout() method, then |
| 3613 | // incoming audio is still processed and statistics are generated. |
| 3614 | TEST_F(PeerConnectionIntegrationTest, |
| 3615 | DisableAudioPlayoutStillGeneratesAudioStats) { |
| 3616 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3617 | ConnectFakeSignaling(); |
| 3618 | |
| 3619 | // Set up audio-only call where playout is disabled but audio-processing is |
| 3620 | // still active. |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3621 | caller()->AddAudioTrack(); |
| 3622 | callee()->AddAudioTrack(); |
henrika | 5f6bf24 | 2017-11-01 11:06:56 +0100 | [diff] [blame] | 3623 | caller()->pc()->SetAudioPlayout(false); |
| 3624 | |
| 3625 | caller()->CreateAndSetAndSignalOffer(); |
| 3626 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3627 | |
| 3628 | // Wait for the callee to receive audio stats. |
| 3629 | EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs); |
| 3630 | } |
| 3631 | |
henrika | 4f167df | 2017-11-01 14:45:55 +0100 | [diff] [blame] | 3632 | // Test that SetAudioRecording can be used to disable audio recording from the |
| 3633 | // start, then later enable it. This may be useful, for example, if the caller |
| 3634 | // wants to ensure that no audio resources are active before a certain state |
| 3635 | // is reached. |
| 3636 | TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) { |
| 3637 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3638 | ConnectFakeSignaling(); |
| 3639 | |
| 3640 | // Set up audio-only call where audio recording is disabled on caller's side. |
| 3641 | caller()->pc()->SetAudioRecording(false); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3642 | caller()->AddAudioTrack(); |
| 3643 | callee()->AddAudioTrack(); |
henrika | 4f167df | 2017-11-01 14:45:55 +0100 | [diff] [blame] | 3644 | caller()->CreateAndSetAndSignalOffer(); |
| 3645 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3646 | |
| 3647 | // Pump messages for a second. |
| 3648 | WAIT(false, 1000); |
| 3649 | // Since caller has disabled audio recording, the callee shouldn't have |
| 3650 | // received anything. |
| 3651 | EXPECT_EQ(0, callee()->audio_frames_received()); |
| 3652 | // As a sanity check, make sure the caller did still see frames on its |
| 3653 | // audio level since audio recording is enabled on the calle side. |
| 3654 | ASSERT_GT(caller()->audio_frames_received(), 0); |
| 3655 | |
| 3656 | // Enable audio recording again, and ensure audio starts flowing. |
| 3657 | caller()->pc()->SetAudioRecording(true); |
| 3658 | ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0, |
| 3659 | kDefaultExpectedAudioFrameCount, 0, |
| 3660 | kMaxWaitForFramesMs); |
| 3661 | } |
| 3662 | |
Taylor Brandstetter | 389a97c | 2018-01-03 16:26:06 -0800 | [diff] [blame] | 3663 | // Test that after closing PeerConnections, they stop sending any packets (ICE, |
| 3664 | // DTLS, RTP...). |
| 3665 | TEST_F(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) { |
| 3666 | // Set up audio/video/data, wait for some frames to be received. |
| 3667 | ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| 3668 | ConnectFakeSignaling(); |
Steve Anton | 1532477 | 2018-01-16 10:26:49 -0800 | [diff] [blame] | 3669 | caller()->AddAudioVideoTracks(); |
Taylor Brandstetter | 389a97c | 2018-01-03 16:26:06 -0800 | [diff] [blame] | 3670 | #ifdef HAVE_SCTP |
| 3671 | caller()->CreateDataChannel(); |
| 3672 | #endif |
| 3673 | caller()->CreateAndSetAndSignalOffer(); |
| 3674 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3675 | ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, |
| 3676 | kDefaultExpectedAudioFrameCount, |
| 3677 | kMaxWaitForFramesMs); |
| 3678 | // Close PeerConnections. |
| 3679 | caller()->pc()->Close(); |
| 3680 | callee()->pc()->Close(); |
| 3681 | // Pump messages for a second, and ensure no new packets end up sent. |
| 3682 | uint32_t sent_packets_a = virtual_socket_server()->sent_packets(); |
| 3683 | WAIT(false, 1000); |
| 3684 | uint32_t sent_packets_b = virtual_socket_server()->sent_packets(); |
| 3685 | EXPECT_EQ(sent_packets_a, sent_packets_b); |
| 3686 | } |
| 3687 | |
Steve Anton | d367921 | 2018-01-17 17:41:02 -0800 | [diff] [blame] | 3688 | // Test that a basic 1 audio and 1 video track call works when Unified Plan |
| 3689 | // semantics configured for both sides. |
| 3690 | TEST_F(PeerConnectionIntegrationTest, UnifiedPlanMediaFlows) { |
| 3691 | PeerConnectionInterface::RTCConfiguration config; |
| 3692 | config.sdp_semantics = SdpSemantics::kUnifiedPlan; |
| 3693 | ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); |
| 3694 | ConnectFakeSignaling(); |
| 3695 | caller()->AddAudioVideoTracks(); |
| 3696 | callee()->AddAudioVideoTracks(); |
| 3697 | caller()->CreateAndSetAndSignalOffer(); |
| 3698 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3699 | ExpectNewFramesReceivedWithWait( |
| 3700 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3701 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3702 | kMaxWaitForFramesMs); |
| 3703 | } |
| 3704 | |
Steve Anton | 74255ff | 2018-01-24 18:32:57 -0800 | [diff] [blame] | 3705 | // Tests that verify interoperability between Plan B and Unified Plan |
| 3706 | // PeerConnections. |
| 3707 | class PeerConnectionIntegrationInteropTest |
| 3708 | : public PeerConnectionIntegrationTest, |
| 3709 | public ::testing::WithParamInterface< |
| 3710 | std::tuple<SdpSemantics, SdpSemantics>> { |
| 3711 | protected: |
| 3712 | PeerConnectionIntegrationInteropTest() |
| 3713 | : caller_semantics_(std::get<0>(GetParam())), |
| 3714 | callee_semantics_(std::get<1>(GetParam())) {} |
| 3715 | |
| 3716 | bool CreatePeerConnectionWrappersWithSemantics() { |
| 3717 | RTCConfiguration caller_config; |
| 3718 | caller_config.sdp_semantics = caller_semantics_; |
| 3719 | RTCConfiguration callee_config; |
| 3720 | callee_config.sdp_semantics = callee_semantics_; |
| 3721 | return CreatePeerConnectionWrappersWithConfig(caller_config, callee_config); |
| 3722 | } |
| 3723 | |
| 3724 | const SdpSemantics caller_semantics_; |
| 3725 | const SdpSemantics callee_semantics_; |
| 3726 | }; |
| 3727 | |
| 3728 | TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) { |
| 3729 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3730 | ConnectFakeSignaling(); |
| 3731 | |
| 3732 | caller()->CreateAndSetAndSignalOffer(); |
| 3733 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3734 | } |
| 3735 | |
| 3736 | TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) { |
| 3737 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3738 | ConnectFakeSignaling(); |
| 3739 | auto audio_sender = caller()->AddAudioTrack(); |
| 3740 | |
| 3741 | caller()->CreateAndSetAndSignalOffer(); |
| 3742 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3743 | |
| 3744 | // Verify that one audio receiver has been created on the remote and that it |
| 3745 | // has the same track ID as the sending track. |
| 3746 | auto receivers = callee()->pc()->GetReceivers(); |
| 3747 | ASSERT_EQ(1u, receivers.size()); |
| 3748 | EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type()); |
| 3749 | EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id()); |
| 3750 | |
| 3751 | // Expect to receive only audio frames on the callee. |
| 3752 | ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0, |
| 3753 | kMaxWaitForFramesMs); |
| 3754 | } |
| 3755 | |
| 3756 | TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) { |
| 3757 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3758 | ConnectFakeSignaling(); |
| 3759 | auto video_sender = caller()->AddVideoTrack(); |
| 3760 | auto audio_sender = caller()->AddAudioTrack(); |
| 3761 | |
| 3762 | caller()->CreateAndSetAndSignalOffer(); |
| 3763 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3764 | |
| 3765 | // Verify that one audio and one video receiver have been created on the |
| 3766 | // remote and that they have the same track IDs as the sending tracks. |
| 3767 | auto audio_receivers = |
| 3768 | callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO); |
| 3769 | ASSERT_EQ(1u, audio_receivers.size()); |
| 3770 | EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id()); |
| 3771 | auto video_receivers = |
| 3772 | callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO); |
| 3773 | ASSERT_EQ(1u, video_receivers.size()); |
| 3774 | EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id()); |
| 3775 | |
| 3776 | // Expect to receive audio and video frames only on the callee. |
| 3777 | ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, |
| 3778 | kDefaultExpectedVideoFrameCount, |
| 3779 | kMaxWaitForFramesMs); |
| 3780 | } |
| 3781 | |
| 3782 | TEST_P(PeerConnectionIntegrationInteropTest, |
| 3783 | OneAudioOneVideoLocalToOneAudioOneVideoRemote) { |
| 3784 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3785 | ConnectFakeSignaling(); |
| 3786 | caller()->AddAudioVideoTracks(); |
| 3787 | callee()->AddAudioVideoTracks(); |
| 3788 | |
| 3789 | caller()->CreateAndSetAndSignalOffer(); |
| 3790 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3791 | |
| 3792 | ExpectNewFramesReceivedWithWait( |
| 3793 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3794 | kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount, |
| 3795 | kMaxWaitForFramesMs); |
| 3796 | } |
| 3797 | |
| 3798 | TEST_P(PeerConnectionIntegrationInteropTest, |
| 3799 | ReverseRolesOneAudioLocalToOneVideoRemote) { |
| 3800 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3801 | ConnectFakeSignaling(); |
| 3802 | caller()->AddAudioTrack(); |
| 3803 | callee()->AddVideoTrack(); |
| 3804 | |
| 3805 | caller()->CreateAndSetAndSignalOffer(); |
| 3806 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3807 | |
| 3808 | // Verify that only the audio track has been negotiated. |
| 3809 | EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size()); |
| 3810 | // Might also check that the callee's NegotiationNeeded flag is set. |
| 3811 | |
| 3812 | // Reverse roles. |
| 3813 | callee()->CreateAndSetAndSignalOffer(); |
| 3814 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3815 | |
| 3816 | // Expect to receive audio frames on the callee and video frames on the |
| 3817 | // caller. |
| 3818 | ExpectNewFramesReceivedWithWait(0, kDefaultExpectedVideoFrameCount, |
| 3819 | kDefaultExpectedAudioFrameCount, 0, |
| 3820 | kMaxWaitForFramesMs); |
| 3821 | } |
| 3822 | |
| 3823 | // Test that if one side offers two video tracks then the other side will only |
| 3824 | // see the first one and ignore the second. |
| 3825 | TEST_P(PeerConnectionIntegrationInteropTest, TwoVideoLocalToNoMediaRemote) { |
| 3826 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3827 | ConnectFakeSignaling(); |
| 3828 | auto first_sender = caller()->AddVideoTrack(); |
| 3829 | caller()->AddVideoTrack(); |
| 3830 | |
| 3831 | caller()->CreateAndSetAndSignalOffer(); |
| 3832 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3833 | |
| 3834 | // Verify that there is only one receiver and it corresponds to the first |
| 3835 | // added track. |
| 3836 | auto receivers = callee()->pc()->GetReceivers(); |
| 3837 | ASSERT_EQ(1u, receivers.size()); |
| 3838 | EXPECT_TRUE(receivers[0]->track()->enabled()); |
| 3839 | EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id()); |
| 3840 | |
| 3841 | // Expect to receive video frames from the one track. |
| 3842 | ExpectNewFramesReceivedWithWait(0, 0, |
| 3843 | 0, kDefaultExpectedVideoFrameCount, |
| 3844 | kMaxWaitForFramesMs); |
| 3845 | } |
| 3846 | |
| 3847 | // Test that in the multi-track case each endpoint only looks at the first track |
| 3848 | // and ignores the second one. |
| 3849 | TEST_P(PeerConnectionIntegrationInteropTest, TwoVideoLocalToTwoVideoRemote) { |
| 3850 | ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics()); |
| 3851 | ConnectFakeSignaling(); |
| 3852 | caller()->AddVideoTrack(); |
| 3853 | caller()->AddVideoTrack(); |
| 3854 | callee()->AddVideoTrack(); |
| 3855 | callee()->AddVideoTrack(); |
| 3856 | |
| 3857 | caller()->CreateAndSetAndSignalOffer(); |
| 3858 | ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| 3859 | |
| 3860 | PeerConnectionWrapper* plan_b = |
| 3861 | (caller_semantics_ == SdpSemantics::kPlanB ? caller() : callee()); |
| 3862 | PeerConnectionWrapper* unified_plan = |
| 3863 | (caller_semantics_ == SdpSemantics::kUnifiedPlan ? caller() : callee()); |
| 3864 | |
| 3865 | // Should have two senders each, one for each track. |
| 3866 | EXPECT_EQ(2u, plan_b->pc()->GetSenders().size()); |
| 3867 | EXPECT_EQ(2u, unified_plan->pc()->GetSenders().size()); |
| 3868 | |
| 3869 | // Plan B will have one receiver since it only looks at the first video |
| 3870 | // section. The receiver should have the same track ID as the sender's first |
| 3871 | // track. |
| 3872 | ASSERT_EQ(1u, plan_b->pc()->GetReceivers().size()); |
| 3873 | EXPECT_EQ(unified_plan->pc()->GetSenders()[0]->track()->id(), |
| 3874 | plan_b->pc()->GetReceivers()[0]->track()->id()); |
| 3875 | |
| 3876 | // Unified Plan will have two receivers since they were created with the |
| 3877 | // transceivers when the tracks were added. |
| 3878 | ASSERT_EQ(2u, unified_plan->pc()->GetReceivers().size()); |
| 3879 | |
| 3880 | if (unified_plan == caller()) { |
| 3881 | // If the Unified Plan endpoint was the caller, then the Plan B endpoint |
| 3882 | // would have rejected the second video media section so we would expect the |
| 3883 | // transceiver to be stopped. |
| 3884 | EXPECT_FALSE(unified_plan->pc()->GetTransceivers()[0]->stopped()); |
| 3885 | EXPECT_TRUE(unified_plan->pc()->GetTransceivers()[1]->stopped()); |
| 3886 | } else { |
| 3887 | // If the Unified Plan endpoint was the callee, then the Plan B endpoint |
| 3888 | // would have offered only one video section so we would expect the first |
| 3889 | // transceiver to map to the first track and the second transceiver to be |
| 3890 | // missing a mid. |
| 3891 | EXPECT_TRUE(unified_plan->pc()->GetTransceivers()[0]->mid()); |
| 3892 | EXPECT_FALSE(unified_plan->pc()->GetTransceivers()[1]->mid()); |
| 3893 | } |
| 3894 | |
| 3895 | // Should be exchanging video frames for the first tracks on each endpoint. |
| 3896 | ExpectNewFramesReceivedWithWait(0, kDefaultExpectedVideoFrameCount, 0, |
| 3897 | kDefaultExpectedVideoFrameCount, |
| 3898 | kMaxWaitForFramesMs); |
| 3899 | } |
| 3900 | |
| 3901 | INSTANTIATE_TEST_CASE_P( |
| 3902 | PeerConnectionIntegrationTest, |
| 3903 | PeerConnectionIntegrationInteropTest, |
| 3904 | Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan), |
| 3905 | std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB))); |
| 3906 | |
deadbeef | 1dcb164 | 2017-03-29 21:08:16 -0700 | [diff] [blame] | 3907 | } // namespace |
| 3908 | |
| 3909 | #endif // if !defined(THREAD_SANITIZER) |