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| 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 #include <stdio.h> | |
| 12 | |
| 13 #include <algorithm> | |
| 14 #include <list> | |
| 15 #include <map> | |
| 16 #include <memory> | |
| 17 #include <utility> | |
| 18 #include <vector> | |
| 19 | |
| 20 #include "webrtc/api/fakemetricsobserver.h" | |
| 21 #include "webrtc/api/mediastreaminterface.h" | |
| 22 #include "webrtc/api/peerconnectioninterface.h" | |
| 23 #include "webrtc/api/test/fakeconstraints.h" | |
| 24 #include "webrtc/base/fakenetwork.h" | |
| 25 #include "webrtc/base/gunit.h" | |
| 26 #include "webrtc/base/helpers.h" | |
| 27 #include "webrtc/base/physicalsocketserver.h" | |
| 28 #include "webrtc/base/ssladapter.h" | |
| 29 #include "webrtc/base/sslstreamadapter.h" | |
| 30 #include "webrtc/base/thread.h" | |
| 31 #include "webrtc/base/virtualsocketserver.h" | |
| 32 #include "webrtc/media/engine/fakewebrtcvideoengine.h" | |
| 33 #include "webrtc/p2p/base/p2pconstants.h" | |
| 34 #include "webrtc/p2p/base/portinterface.h" | |
| 35 #include "webrtc/p2p/base/sessiondescription.h" | |
| 36 #include "webrtc/p2p/base/testturnserver.h" | |
| 37 #include "webrtc/p2p/client/basicportallocator.h" | |
| 38 #include "webrtc/pc/dtmfsender.h" | |
| 39 #include "webrtc/pc/localaudiosource.h" | |
| 40 #include "webrtc/pc/mediasession.h" | |
| 41 #include "webrtc/pc/peerconnection.h" | |
| 42 #include "webrtc/pc/peerconnectionfactory.h" | |
| 43 #include "webrtc/pc/test/fakeaudiocapturemodule.h" | |
| 44 #include "webrtc/pc/test/fakeperiodicvideocapturer.h" | |
| 45 #include "webrtc/pc/test/fakertccertificategenerator.h" | |
| 46 #include "webrtc/pc/test/fakevideotrackrenderer.h" | |
| 47 #include "webrtc/pc/test/mockpeerconnectionobservers.h" | |
| 48 | |
| 49 using cricket::ContentInfo; | |
| 50 using cricket::FakeWebRtcVideoDecoder; | |
| 51 using cricket::FakeWebRtcVideoDecoderFactory; | |
| 52 using cricket::FakeWebRtcVideoEncoder; | |
| 53 using cricket::FakeWebRtcVideoEncoderFactory; | |
| 54 using cricket::MediaContentDescription; | |
| 55 using webrtc::DataBuffer; | |
| 56 using webrtc::DataChannelInterface; | |
| 57 using webrtc::DtmfSender; | |
| 58 using webrtc::DtmfSenderInterface; | |
| 59 using webrtc::DtmfSenderObserverInterface; | |
| 60 using webrtc::FakeConstraints; | |
| 61 using webrtc::MediaConstraintsInterface; | |
| 62 using webrtc::MediaStreamInterface; | |
| 63 using webrtc::MediaStreamTrackInterface; | |
| 64 using webrtc::MockCreateSessionDescriptionObserver; | |
| 65 using webrtc::MockDataChannelObserver; | |
| 66 using webrtc::MockSetSessionDescriptionObserver; | |
| 67 using webrtc::MockStatsObserver; | |
| 68 using webrtc::ObserverInterface; | |
| 69 using webrtc::PeerConnectionInterface; | |
| 70 using webrtc::PeerConnectionFactory; | |
| 71 using webrtc::SessionDescriptionInterface; | |
| 72 using webrtc::StreamCollectionInterface; | |
| 73 | |
| 74 namespace { | |
| 75 | |
| 76 static const int kMaxWaitMs = 10000; | |
| 77 // Disable for TSan v2, see | |
| 78 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. | |
| 79 // This declaration is also #ifdef'd as it causes uninitialized-variable | |
| 80 // warnings. | |
| 81 #if !defined(THREAD_SANITIZER) | |
| 82 static const int kMaxWaitForStatsMs = 3000; | |
| 83 #endif | |
| 84 static const int kMaxWaitForActivationMs = 5000; | |
| 85 static const int kMaxWaitForFramesMs = 10000; | |
| 86 static const int kEndAudioFrameCount = 3; | |
| 87 static const int kEndVideoFrameCount = 3; | |
| 88 | |
| 89 static const char kStreamLabelBase[] = "stream_label"; | |
| 90 static const char kVideoTrackLabelBase[] = "video_track"; | |
| 91 static const char kAudioTrackLabelBase[] = "audio_track"; | |
| 92 static const char kDataChannelLabel[] = "data_channel"; | |
| 93 | |
| 94 // Disable for TSan v2, see | |
| 95 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. | |
| 96 // This declaration is also #ifdef'd as it causes unused-variable errors. | |
| 97 #if !defined(THREAD_SANITIZER) | |
| 98 // SRTP cipher name negotiated by the tests. This must be updated if the | |
| 99 // default changes. | |
| 100 static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32; | |
| 101 static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM; | |
| 102 #endif | |
| 103 | |
| 104 // Used to simulate signaling ICE/SDP between two PeerConnections. | |
| 105 enum Message { MSG_SDP_MESSAGE, MSG_ICE_MESSAGE }; | |
| 106 | |
| 107 struct SdpMessage { | |
| 108 std::string type; | |
| 109 std::string msg; | |
| 110 }; | |
| 111 | |
| 112 struct IceMessage { | |
| 113 std::string sdp_mid; | |
| 114 int sdp_mline_index; | |
| 115 std::string msg; | |
| 116 }; | |
| 117 | |
| 118 static void RemoveLinesFromSdp(const std::string& line_start, | |
| 119 std::string* sdp) { | |
| 120 const char kSdpLineEnd[] = "\r\n"; | |
| 121 size_t ssrc_pos = 0; | |
| 122 while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) != | |
| 123 std::string::npos) { | |
| 124 size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos); | |
| 125 sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd)); | |
| 126 } | |
| 127 } | |
| 128 | |
| 129 bool StreamsHaveAudioTrack(StreamCollectionInterface* streams) { | |
| 130 for (size_t idx = 0; idx < streams->count(); idx++) { | |
| 131 auto stream = streams->at(idx); | |
| 132 if (stream->GetAudioTracks().size() > 0) { | |
| 133 return true; | |
| 134 } | |
| 135 } | |
| 136 return false; | |
| 137 } | |
| 138 | |
| 139 bool StreamsHaveVideoTrack(StreamCollectionInterface* streams) { | |
| 140 for (size_t idx = 0; idx < streams->count(); idx++) { | |
| 141 auto stream = streams->at(idx); | |
| 142 if (stream->GetVideoTracks().size() > 0) { | |
| 143 return true; | |
| 144 } | |
| 145 } | |
| 146 return false; | |
| 147 } | |
| 148 | |
| 149 class SignalingMessageReceiver { | |
| 150 public: | |
| 151 virtual void ReceiveSdpMessage(const std::string& type, | |
| 152 std::string& msg) = 0; | |
| 153 virtual void ReceiveIceMessage(const std::string& sdp_mid, | |
| 154 int sdp_mline_index, | |
| 155 const std::string& msg) = 0; | |
| 156 | |
| 157 protected: | |
| 158 SignalingMessageReceiver() {} | |
| 159 virtual ~SignalingMessageReceiver() {} | |
| 160 }; | |
| 161 | |
| 162 class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { | |
| 163 public: | |
| 164 MockRtpReceiverObserver(cricket::MediaType media_type) | |
| 165 : expected_media_type_(media_type) {} | |
| 166 | |
| 167 void OnFirstPacketReceived(cricket::MediaType media_type) override { | |
| 168 ASSERT_EQ(expected_media_type_, media_type); | |
| 169 first_packet_received_ = true; | |
| 170 } | |
| 171 | |
| 172 bool first_packet_received() { return first_packet_received_; } | |
| 173 | |
| 174 virtual ~MockRtpReceiverObserver() {} | |
| 175 | |
| 176 private: | |
| 177 bool first_packet_received_ = false; | |
| 178 cricket::MediaType expected_media_type_; | |
| 179 }; | |
| 180 | |
| 181 class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, | |
| 182 public SignalingMessageReceiver, | |
| 183 public ObserverInterface, | |
| 184 public rtc::MessageHandler { | |
| 185 public: | |
| 186 // If |config| is not provided, uses a default constructed RTCConfiguration. | |
| 187 static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore( | |
| 188 const std::string& id, | |
| 189 const MediaConstraintsInterface* constraints, | |
| 190 const PeerConnectionFactory::Options* options, | |
| 191 const PeerConnectionInterface::RTCConfiguration* config, | |
| 192 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | |
| 193 bool prefer_constraint_apis, | |
| 194 rtc::Thread* network_thread, | |
| 195 rtc::Thread* worker_thread) { | |
| 196 PeerConnectionTestClient* client(new PeerConnectionTestClient(id)); | |
| 197 if (!client->Init(constraints, options, config, std::move(cert_generator), | |
| 198 prefer_constraint_apis, network_thread, worker_thread)) { | |
| 199 delete client; | |
| 200 return nullptr; | |
| 201 } | |
| 202 return client; | |
| 203 } | |
| 204 | |
| 205 static PeerConnectionTestClient* CreateClient( | |
| 206 const std::string& id, | |
| 207 const MediaConstraintsInterface* constraints, | |
| 208 const PeerConnectionFactory::Options* options, | |
| 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 | |
| 215 return CreateClientWithDtlsIdentityStore(id, constraints, options, config, | |
| 216 std::move(cert_generator), true, | |
| 217 network_thread, worker_thread); | |
| 218 } | |
| 219 | |
| 220 static PeerConnectionTestClient* CreateClientPreferNoConstraints( | |
| 221 const std::string& id, | |
| 222 const PeerConnectionFactory::Options* options, | |
| 223 rtc::Thread* network_thread, | |
| 224 rtc::Thread* worker_thread) { | |
| 225 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
| 226 new FakeRTCCertificateGenerator()); | |
| 227 | |
| 228 return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr, | |
| 229 std::move(cert_generator), false, | |
| 230 network_thread, worker_thread); | |
| 231 } | |
| 232 | |
| 233 ~PeerConnectionTestClient() { | |
| 234 } | |
| 235 | |
| 236 void Negotiate() { Negotiate(true, true); } | |
| 237 | |
| 238 void Negotiate(bool audio, bool video) { | |
| 239 std::unique_ptr<SessionDescriptionInterface> offer; | |
| 240 ASSERT_TRUE(DoCreateOffer(&offer)); | |
| 241 | |
| 242 if (offer->description()->GetContentByName("audio")) { | |
| 243 offer->description()->GetContentByName("audio")->rejected = !audio; | |
| 244 } | |
| 245 if (offer->description()->GetContentByName("video")) { | |
| 246 offer->description()->GetContentByName("video")->rejected = !video; | |
| 247 } | |
| 248 | |
| 249 std::string sdp; | |
| 250 EXPECT_TRUE(offer->ToString(&sdp)); | |
| 251 EXPECT_TRUE(DoSetLocalDescription(offer.release())); | |
| 252 SendSdpMessage(webrtc::SessionDescriptionInterface::kOffer, sdp); | |
| 253 } | |
| 254 | |
| 255 void SendSdpMessage(const std::string& type, std::string& msg) { | |
| 256 if (signaling_delay_ms_ == 0) { | |
| 257 if (signaling_message_receiver_) { | |
| 258 signaling_message_receiver_->ReceiveSdpMessage(type, msg); | |
| 259 } | |
| 260 } else { | |
| 261 rtc::Thread::Current()->PostDelayed( | |
| 262 RTC_FROM_HERE, signaling_delay_ms_, this, MSG_SDP_MESSAGE, | |
| 263 new rtc::TypedMessageData<SdpMessage>({type, msg})); | |
| 264 } | |
| 265 } | |
| 266 | |
| 267 void SendIceMessage(const std::string& sdp_mid, | |
| 268 int sdp_mline_index, | |
| 269 const std::string& msg) { | |
| 270 if (signaling_delay_ms_ == 0) { | |
| 271 if (signaling_message_receiver_) { | |
| 272 signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index, | |
| 273 msg); | |
| 274 } | |
| 275 } else { | |
| 276 rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, signaling_delay_ms_, | |
| 277 this, MSG_ICE_MESSAGE, | |
| 278 new rtc::TypedMessageData<IceMessage>( | |
| 279 {sdp_mid, sdp_mline_index, msg})); | |
| 280 } | |
| 281 } | |
| 282 | |
| 283 // MessageHandler callback. | |
| 284 void OnMessage(rtc::Message* msg) override { | |
| 285 switch (msg->message_id) { | |
| 286 case MSG_SDP_MESSAGE: { | |
| 287 auto sdp_message = | |
| 288 static_cast<rtc::TypedMessageData<SdpMessage>*>(msg->pdata); | |
| 289 if (signaling_message_receiver_) { | |
| 290 signaling_message_receiver_->ReceiveSdpMessage( | |
| 291 sdp_message->data().type, sdp_message->data().msg); | |
| 292 } | |
| 293 delete sdp_message; | |
| 294 break; | |
| 295 } | |
| 296 case MSG_ICE_MESSAGE: { | |
| 297 auto ice_message = | |
| 298 static_cast<rtc::TypedMessageData<IceMessage>*>(msg->pdata); | |
| 299 if (signaling_message_receiver_) { | |
| 300 signaling_message_receiver_->ReceiveIceMessage( | |
| 301 ice_message->data().sdp_mid, ice_message->data().sdp_mline_index, | |
| 302 ice_message->data().msg); | |
| 303 } | |
| 304 delete ice_message; | |
| 305 break; | |
| 306 } | |
| 307 default: | |
| 308 RTC_CHECK(false); | |
| 309 } | |
| 310 } | |
| 311 | |
| 312 // SignalingMessageReceiver callback. | |
| 313 void ReceiveSdpMessage(const std::string& type, std::string& msg) override { | |
| 314 FilterIncomingSdpMessage(&msg); | |
| 315 if (type == webrtc::SessionDescriptionInterface::kOffer) { | |
| 316 HandleIncomingOffer(msg); | |
| 317 } else { | |
| 318 HandleIncomingAnswer(msg); | |
| 319 } | |
| 320 } | |
| 321 | |
| 322 // SignalingMessageReceiver callback. | |
| 323 void ReceiveIceMessage(const std::string& sdp_mid, | |
| 324 int sdp_mline_index, | |
| 325 const std::string& msg) override { | |
| 326 LOG(INFO) << id_ << "ReceiveIceMessage"; | |
| 327 std::unique_ptr<webrtc::IceCandidateInterface> candidate( | |
| 328 webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr)); | |
| 329 EXPECT_TRUE(pc()->AddIceCandidate(candidate.get())); | |
| 330 } | |
| 331 | |
| 332 // PeerConnectionObserver callbacks. | |
| 333 void OnSignalingChange( | |
| 334 webrtc::PeerConnectionInterface::SignalingState new_state) override { | |
| 335 EXPECT_EQ(pc()->signaling_state(), new_state); | |
| 336 } | |
| 337 void OnAddStream( | |
| 338 rtc::scoped_refptr<MediaStreamInterface> media_stream) override { | |
| 339 media_stream->RegisterObserver(this); | |
| 340 for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) { | |
| 341 const std::string id = media_stream->GetVideoTracks()[i]->id(); | |
| 342 ASSERT_TRUE(fake_video_renderers_.find(id) == | |
| 343 fake_video_renderers_.end()); | |
| 344 fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer( | |
| 345 media_stream->GetVideoTracks()[i])); | |
| 346 } | |
| 347 } | |
| 348 void OnRemoveStream( | |
| 349 rtc::scoped_refptr<MediaStreamInterface> media_stream) override {} | |
| 350 void OnRenegotiationNeeded() override {} | |
| 351 void OnIceConnectionChange( | |
| 352 webrtc::PeerConnectionInterface::IceConnectionState new_state) override { | |
| 353 EXPECT_EQ(pc()->ice_connection_state(), new_state); | |
| 354 } | |
| 355 void OnIceGatheringChange( | |
| 356 webrtc::PeerConnectionInterface::IceGatheringState new_state) override { | |
| 357 EXPECT_EQ(pc()->ice_gathering_state(), new_state); | |
| 358 } | |
| 359 void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { | |
| 360 LOG(INFO) << id_ << "OnIceCandidate"; | |
| 361 | |
| 362 std::string ice_sdp; | |
| 363 EXPECT_TRUE(candidate->ToString(&ice_sdp)); | |
| 364 if (signaling_message_receiver_ == nullptr) { | |
| 365 // Remote party may be deleted. | |
| 366 return; | |
| 367 } | |
| 368 SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp); | |
| 369 } | |
| 370 | |
| 371 // MediaStreamInterface callback | |
| 372 void OnChanged() override { | |
| 373 // Track added or removed from MediaStream, so update our renderers. | |
| 374 rtc::scoped_refptr<StreamCollectionInterface> remote_streams = | |
| 375 pc()->remote_streams(); | |
| 376 // Remove renderers for tracks that were removed. | |
| 377 for (auto it = fake_video_renderers_.begin(); | |
| 378 it != fake_video_renderers_.end();) { | |
| 379 if (remote_streams->FindVideoTrack(it->first) == nullptr) { | |
| 380 auto to_remove = it++; | |
| 381 removed_fake_video_renderers_.push_back(std::move(to_remove->second)); | |
| 382 fake_video_renderers_.erase(to_remove); | |
| 383 } else { | |
| 384 ++it; | |
| 385 } | |
| 386 } | |
| 387 // Create renderers for new video tracks. | |
| 388 for (size_t stream_index = 0; stream_index < remote_streams->count(); | |
| 389 ++stream_index) { | |
| 390 MediaStreamInterface* remote_stream = remote_streams->at(stream_index); | |
| 391 for (size_t track_index = 0; | |
| 392 track_index < remote_stream->GetVideoTracks().size(); | |
| 393 ++track_index) { | |
| 394 const std::string id = | |
| 395 remote_stream->GetVideoTracks()[track_index]->id(); | |
| 396 if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) { | |
| 397 continue; | |
| 398 } | |
| 399 fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer( | |
| 400 remote_stream->GetVideoTracks()[track_index])); | |
| 401 } | |
| 402 } | |
| 403 } | |
| 404 | |
| 405 void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) { | |
| 406 video_constraints_ = video_constraint; | |
| 407 } | |
| 408 | |
| 409 void AddMediaStream(bool audio, bool video) { | |
| 410 std::string stream_label = | |
| 411 kStreamLabelBase + | |
| 412 rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count())); | |
| 413 rtc::scoped_refptr<MediaStreamInterface> stream = | |
| 414 peer_connection_factory_->CreateLocalMediaStream(stream_label); | |
| 415 | |
| 416 if (audio && can_receive_audio()) { | |
| 417 stream->AddTrack(CreateLocalAudioTrack(stream_label)); | |
| 418 } | |
| 419 if (video && can_receive_video()) { | |
| 420 stream->AddTrack(CreateLocalVideoTrack(stream_label)); | |
| 421 } | |
| 422 | |
| 423 EXPECT_TRUE(pc()->AddStream(stream)); | |
| 424 } | |
| 425 | |
| 426 size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); } | |
| 427 | |
| 428 bool SessionActive() { | |
| 429 return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable; | |
| 430 } | |
| 431 | |
| 432 // Automatically add a stream when receiving an offer, if we don't have one. | |
| 433 // Defaults to true. | |
| 434 void set_auto_add_stream(bool auto_add_stream) { | |
| 435 auto_add_stream_ = auto_add_stream; | |
| 436 } | |
| 437 | |
| 438 void set_signaling_message_receiver( | |
| 439 SignalingMessageReceiver* signaling_message_receiver) { | |
| 440 signaling_message_receiver_ = signaling_message_receiver; | |
| 441 } | |
| 442 | |
| 443 void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; } | |
| 444 | |
| 445 void EnableVideoDecoderFactory() { | |
| 446 video_decoder_factory_enabled_ = true; | |
| 447 fake_video_decoder_factory_->AddSupportedVideoCodecType( | |
| 448 webrtc::kVideoCodecVP8); | |
| 449 } | |
| 450 | |
| 451 void IceRestart() { | |
| 452 offer_answer_constraints_.SetMandatoryIceRestart(true); | |
| 453 offer_answer_options_.ice_restart = true; | |
| 454 SetExpectIceRestart(true); | |
| 455 } | |
| 456 | |
| 457 void SetExpectIceRestart(bool expect_restart) { | |
| 458 expect_ice_restart_ = expect_restart; | |
| 459 } | |
| 460 | |
| 461 bool ExpectIceRestart() const { return expect_ice_restart_; } | |
| 462 | |
| 463 void SetExpectIceRenomination(bool expect_renomination) { | |
| 464 expect_ice_renomination_ = expect_renomination; | |
| 465 } | |
| 466 void SetExpectRemoteIceRenomination(bool expect_renomination) { | |
| 467 expect_remote_ice_renomination_ = expect_renomination; | |
| 468 } | |
| 469 bool ExpectIceRenomination() { return expect_ice_renomination_; } | |
| 470 bool ExpectRemoteIceRenomination() { return expect_remote_ice_renomination_; } | |
| 471 | |
| 472 // The below 3 methods assume streams will be offered. | |
| 473 // Thus they'll only set the "offer to receive" flag to true if it's | |
| 474 // currently false, not if it's just unset. | |
| 475 void SetReceiveAudioVideo(bool audio, bool video) { | |
| 476 SetReceiveAudio(audio); | |
| 477 SetReceiveVideo(video); | |
| 478 ASSERT_EQ(audio, can_receive_audio()); | |
| 479 ASSERT_EQ(video, can_receive_video()); | |
| 480 } | |
| 481 | |
| 482 void SetReceiveAudio(bool audio) { | |
| 483 if (audio && can_receive_audio()) { | |
| 484 return; | |
| 485 } | |
| 486 offer_answer_constraints_.SetMandatoryReceiveAudio(audio); | |
| 487 offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0; | |
| 488 } | |
| 489 | |
| 490 void SetReceiveVideo(bool video) { | |
| 491 if (video && can_receive_video()) { | |
| 492 return; | |
| 493 } | |
| 494 offer_answer_constraints_.SetMandatoryReceiveVideo(video); | |
| 495 offer_answer_options_.offer_to_receive_video = video ? 1 : 0; | |
| 496 } | |
| 497 | |
| 498 void SetOfferToReceiveAudioVideo(bool audio, bool video) { | |
| 499 offer_answer_constraints_.SetMandatoryReceiveAudio(audio); | |
| 500 offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0; | |
| 501 offer_answer_constraints_.SetMandatoryReceiveVideo(video); | |
| 502 offer_answer_options_.offer_to_receive_video = video ? 1 : 0; | |
| 503 } | |
| 504 | |
| 505 void RemoveMsidFromReceivedSdp(bool remove) { remove_msid_ = remove; } | |
| 506 | |
| 507 void RemoveSdesCryptoFromReceivedSdp(bool remove) { remove_sdes_ = remove; } | |
| 508 | |
| 509 void RemoveBundleFromReceivedSdp(bool remove) { remove_bundle_ = remove; } | |
| 510 | |
| 511 void RemoveCvoFromReceivedSdp(bool remove) { remove_cvo_ = remove; } | |
| 512 | |
| 513 void MakeSpecCompliantMaxBundleOfferFromReceivedSdp(bool real) { | |
| 514 make_spec_compliant_max_bundle_offer_ = real; | |
| 515 } | |
| 516 | |
| 517 bool can_receive_audio() { | |
| 518 bool value; | |
| 519 if (prefer_constraint_apis_) { | |
| 520 if (webrtc::FindConstraint( | |
| 521 &offer_answer_constraints_, | |
| 522 MediaConstraintsInterface::kOfferToReceiveAudio, &value, | |
| 523 nullptr)) { | |
| 524 return value; | |
| 525 } | |
| 526 return true; | |
| 527 } | |
| 528 return offer_answer_options_.offer_to_receive_audio > 0 || | |
| 529 offer_answer_options_.offer_to_receive_audio == | |
| 530 PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined; | |
| 531 } | |
| 532 | |
| 533 bool can_receive_video() { | |
| 534 bool value; | |
| 535 if (prefer_constraint_apis_) { | |
| 536 if (webrtc::FindConstraint( | |
| 537 &offer_answer_constraints_, | |
| 538 MediaConstraintsInterface::kOfferToReceiveVideo, &value, | |
| 539 nullptr)) { | |
| 540 return value; | |
| 541 } | |
| 542 return true; | |
| 543 } | |
| 544 return offer_answer_options_.offer_to_receive_video > 0 || | |
| 545 offer_answer_options_.offer_to_receive_video == | |
| 546 PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined; | |
| 547 } | |
| 548 | |
| 549 void OnDataChannel( | |
| 550 rtc::scoped_refptr<DataChannelInterface> data_channel) override { | |
| 551 LOG(INFO) << id_ << "OnDataChannel"; | |
| 552 data_channel_ = data_channel; | |
| 553 data_observer_.reset(new MockDataChannelObserver(data_channel)); | |
| 554 } | |
| 555 | |
| 556 void CreateDataChannel() { CreateDataChannel(nullptr); } | |
| 557 | |
| 558 void CreateDataChannel(const webrtc::DataChannelInit* init) { | |
| 559 data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init); | |
| 560 ASSERT_TRUE(data_channel_.get() != nullptr); | |
| 561 data_observer_.reset(new MockDataChannelObserver(data_channel_)); | |
| 562 } | |
| 563 | |
| 564 rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack( | |
| 565 const std::string& stream_label) { | |
| 566 FakeConstraints constraints; | |
| 567 // Disable highpass filter so that we can get all the test audio frames. | |
| 568 constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false); | |
| 569 rtc::scoped_refptr<webrtc::AudioSourceInterface> source = | |
| 570 peer_connection_factory_->CreateAudioSource(&constraints); | |
| 571 // TODO(perkj): Test audio source when it is implemented. Currently audio | |
| 572 // always use the default input. | |
| 573 std::string label = stream_label + kAudioTrackLabelBase; | |
| 574 return peer_connection_factory_->CreateAudioTrack(label, source); | |
| 575 } | |
| 576 | |
| 577 rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack( | |
| 578 const std::string& stream_label) { | |
| 579 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky. | |
| 580 FakeConstraints source_constraints = video_constraints_; | |
| 581 source_constraints.SetMandatoryMaxFrameRate(10); | |
| 582 | |
| 583 cricket::FakeVideoCapturer* fake_capturer = | |
| 584 new webrtc::FakePeriodicVideoCapturer(); | |
| 585 fake_capturer->SetRotation(capture_rotation_); | |
| 586 video_capturers_.push_back(fake_capturer); | |
| 587 rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source = | |
| 588 peer_connection_factory_->CreateVideoSource( | |
| 589 std::unique_ptr<cricket::VideoCapturer>(fake_capturer), | |
| 590 &source_constraints); | |
| 591 std::string label = stream_label + kVideoTrackLabelBase; | |
| 592 | |
| 593 rtc::scoped_refptr<webrtc::VideoTrackInterface> track( | |
| 594 peer_connection_factory_->CreateVideoTrack(label, source)); | |
| 595 if (!local_video_renderer_) { | |
| 596 local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track)); | |
| 597 } | |
| 598 return track; | |
| 599 } | |
| 600 | |
| 601 DataChannelInterface* data_channel() { return data_channel_; } | |
| 602 const MockDataChannelObserver* data_observer() const { | |
| 603 return data_observer_.get(); | |
| 604 } | |
| 605 | |
| 606 webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); } | |
| 607 | |
| 608 void StopVideoCapturers() { | |
| 609 for (auto* capturer : video_capturers_) { | |
| 610 capturer->Stop(); | |
| 611 } | |
| 612 } | |
| 613 | |
| 614 void SetCaptureRotation(webrtc::VideoRotation rotation) { | |
| 615 ASSERT_TRUE(video_capturers_.empty()); | |
| 616 capture_rotation_ = rotation; | |
| 617 } | |
| 618 | |
| 619 bool AudioFramesReceivedCheck(int number_of_frames) const { | |
| 620 return number_of_frames <= fake_audio_capture_module_->frames_received(); | |
| 621 } | |
| 622 | |
| 623 int audio_frames_received() const { | |
| 624 return fake_audio_capture_module_->frames_received(); | |
| 625 } | |
| 626 | |
| 627 bool VideoFramesReceivedCheck(int number_of_frames) { | |
| 628 if (video_decoder_factory_enabled_) { | |
| 629 const std::vector<FakeWebRtcVideoDecoder*>& decoders | |
| 630 = fake_video_decoder_factory_->decoders(); | |
| 631 if (decoders.empty()) { | |
| 632 return number_of_frames <= 0; | |
| 633 } | |
| 634 // Note - this checks that EACH decoder has the requisite number | |
| 635 // of frames. The video_frames_received() function sums them. | |
| 636 for (FakeWebRtcVideoDecoder* decoder : decoders) { | |
| 637 if (number_of_frames > decoder->GetNumFramesReceived()) { | |
| 638 return false; | |
| 639 } | |
| 640 } | |
| 641 return true; | |
| 642 } else { | |
| 643 if (fake_video_renderers_.empty()) { | |
| 644 return number_of_frames <= 0; | |
| 645 } | |
| 646 | |
| 647 for (const auto& pair : fake_video_renderers_) { | |
| 648 if (number_of_frames > pair.second->num_rendered_frames()) { | |
| 649 return false; | |
| 650 } | |
| 651 } | |
| 652 return true; | |
| 653 } | |
| 654 } | |
| 655 | |
| 656 int video_frames_received() const { | |
| 657 int total = 0; | |
| 658 if (video_decoder_factory_enabled_) { | |
| 659 const std::vector<FakeWebRtcVideoDecoder*>& decoders = | |
| 660 fake_video_decoder_factory_->decoders(); | |
| 661 for (const FakeWebRtcVideoDecoder* decoder : decoders) { | |
| 662 total += decoder->GetNumFramesReceived(); | |
| 663 } | |
| 664 } else { | |
| 665 for (const auto& pair : fake_video_renderers_) { | |
| 666 total += pair.second->num_rendered_frames(); | |
| 667 } | |
| 668 for (const auto& renderer : removed_fake_video_renderers_) { | |
| 669 total += renderer->num_rendered_frames(); | |
| 670 } | |
| 671 } | |
| 672 return total; | |
| 673 } | |
| 674 | |
| 675 // Verify the CreateDtmfSender interface | |
| 676 void VerifyDtmf() { | |
| 677 std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver()); | |
| 678 rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender; | |
| 679 | |
| 680 // We can't create a DTMF sender with an invalid audio track or a non local | |
| 681 // track. | |
| 682 EXPECT_TRUE(peer_connection_->CreateDtmfSender(nullptr) == nullptr); | |
| 683 rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack( | |
| 684 peer_connection_factory_->CreateAudioTrack("dummy_track", nullptr)); | |
| 685 EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == nullptr); | |
| 686 | |
| 687 // We should be able to create a DTMF sender from a local track. | |
| 688 webrtc::AudioTrackInterface* localtrack = | |
| 689 peer_connection_->local_streams()->at(0)->GetAudioTracks()[0]; | |
| 690 dtmf_sender = peer_connection_->CreateDtmfSender(localtrack); | |
| 691 EXPECT_TRUE(dtmf_sender.get() != nullptr); | |
| 692 dtmf_sender->RegisterObserver(observer.get()); | |
| 693 | |
| 694 // Test the DtmfSender object just created. | |
| 695 EXPECT_TRUE(dtmf_sender->CanInsertDtmf()); | |
| 696 EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50)); | |
| 697 | |
| 698 // We don't need to verify that the DTMF tones are actually sent out because | |
| 699 // that is already covered by the tests of the lower level components. | |
| 700 | |
| 701 EXPECT_TRUE_WAIT(observer->completed(), kMaxWaitMs); | |
| 702 std::vector<std::string> tones; | |
| 703 tones.push_back("1"); | |
| 704 tones.push_back("a"); | |
| 705 tones.push_back(""); | |
| 706 observer->Verify(tones); | |
| 707 | |
| 708 dtmf_sender->UnregisterObserver(); | |
| 709 } | |
| 710 | |
| 711 // Verifies that the SessionDescription have rejected the appropriate media | |
| 712 // content. | |
| 713 void VerifyRejectedMediaInSessionDescription() { | |
| 714 ASSERT_TRUE(peer_connection_->remote_description() != nullptr); | |
| 715 ASSERT_TRUE(peer_connection_->local_description() != nullptr); | |
| 716 const cricket::SessionDescription* remote_desc = | |
| 717 peer_connection_->remote_description()->description(); | |
| 718 const cricket::SessionDescription* local_desc = | |
| 719 peer_connection_->local_description()->description(); | |
| 720 | |
| 721 const ContentInfo* remote_audio_content = GetFirstAudioContent(remote_desc); | |
| 722 if (remote_audio_content) { | |
| 723 const ContentInfo* audio_content = | |
| 724 GetFirstAudioContent(local_desc); | |
| 725 EXPECT_EQ(can_receive_audio(), !audio_content->rejected); | |
| 726 } | |
| 727 | |
| 728 const ContentInfo* remote_video_content = GetFirstVideoContent(remote_desc); | |
| 729 if (remote_video_content) { | |
| 730 const ContentInfo* video_content = | |
| 731 GetFirstVideoContent(local_desc); | |
| 732 EXPECT_EQ(can_receive_video(), !video_content->rejected); | |
| 733 } | |
| 734 } | |
| 735 | |
| 736 void VerifyLocalIceUfragAndPassword() { | |
| 737 ASSERT_TRUE(peer_connection_->local_description() != nullptr); | |
| 738 const cricket::SessionDescription* desc = | |
| 739 peer_connection_->local_description()->description(); | |
| 740 const cricket::ContentInfos& contents = desc->contents(); | |
| 741 | |
| 742 for (size_t index = 0; index < contents.size(); ++index) { | |
| 743 if (contents[index].rejected) | |
| 744 continue; | |
| 745 const cricket::TransportDescription* transport_desc = | |
| 746 desc->GetTransportDescriptionByName(contents[index].name); | |
| 747 | |
| 748 std::map<int, IceUfragPwdPair>::const_iterator ufragpair_it = | |
| 749 ice_ufrag_pwd_.find(static_cast<int>(index)); | |
| 750 if (ufragpair_it == ice_ufrag_pwd_.end()) { | |
| 751 ASSERT_FALSE(ExpectIceRestart()); | |
| 752 ice_ufrag_pwd_[static_cast<int>(index)] = | |
| 753 IceUfragPwdPair(transport_desc->ice_ufrag, transport_desc->ice_pwd); | |
| 754 } else if (ExpectIceRestart()) { | |
| 755 const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second; | |
| 756 EXPECT_NE(ufrag_pwd.first, transport_desc->ice_ufrag); | |
| 757 EXPECT_NE(ufrag_pwd.second, transport_desc->ice_pwd); | |
| 758 } else { | |
| 759 const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second; | |
| 760 EXPECT_EQ(ufrag_pwd.first, transport_desc->ice_ufrag); | |
| 761 EXPECT_EQ(ufrag_pwd.second, transport_desc->ice_pwd); | |
| 762 } | |
| 763 } | |
| 764 } | |
| 765 | |
| 766 void VerifyLocalIceRenomination() { | |
| 767 ASSERT_TRUE(peer_connection_->local_description() != nullptr); | |
| 768 const cricket::SessionDescription* desc = | |
| 769 peer_connection_->local_description()->description(); | |
| 770 const cricket::ContentInfos& contents = desc->contents(); | |
| 771 | |
| 772 for (auto content : contents) { | |
| 773 if (content.rejected) | |
| 774 continue; | |
| 775 const cricket::TransportDescription* transport_desc = | |
| 776 desc->GetTransportDescriptionByName(content.name); | |
| 777 const auto& options = transport_desc->transport_options; | |
| 778 auto iter = std::find(options.begin(), options.end(), | |
| 779 cricket::ICE_RENOMINATION_STR); | |
| 780 EXPECT_EQ(ExpectIceRenomination(), iter != options.end()); | |
| 781 } | |
| 782 } | |
| 783 | |
| 784 void VerifyRemoteIceRenomination() { | |
| 785 ASSERT_TRUE(peer_connection_->remote_description() != nullptr); | |
| 786 const cricket::SessionDescription* desc = | |
| 787 peer_connection_->remote_description()->description(); | |
| 788 const cricket::ContentInfos& contents = desc->contents(); | |
| 789 | |
| 790 for (auto content : contents) { | |
| 791 if (content.rejected) | |
| 792 continue; | |
| 793 const cricket::TransportDescription* transport_desc = | |
| 794 desc->GetTransportDescriptionByName(content.name); | |
| 795 const auto& options = transport_desc->transport_options; | |
| 796 auto iter = std::find(options.begin(), options.end(), | |
| 797 cricket::ICE_RENOMINATION_STR); | |
| 798 EXPECT_EQ(ExpectRemoteIceRenomination(), iter != options.end()); | |
| 799 } | |
| 800 } | |
| 801 | |
| 802 int GetAudioOutputLevelStats(webrtc::MediaStreamTrackInterface* track) { | |
| 803 rtc::scoped_refptr<MockStatsObserver> | |
| 804 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 805 EXPECT_TRUE(peer_connection_->GetStats( | |
| 806 observer, track, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 807 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 808 EXPECT_NE(0, observer->timestamp()); | |
| 809 return observer->AudioOutputLevel(); | |
| 810 } | |
| 811 | |
| 812 int GetAudioInputLevelStats() { | |
| 813 rtc::scoped_refptr<MockStatsObserver> | |
| 814 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 815 EXPECT_TRUE(peer_connection_->GetStats( | |
| 816 observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 817 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 818 EXPECT_NE(0, observer->timestamp()); | |
| 819 return observer->AudioInputLevel(); | |
| 820 } | |
| 821 | |
| 822 int GetBytesReceivedStats(webrtc::MediaStreamTrackInterface* track) { | |
| 823 rtc::scoped_refptr<MockStatsObserver> | |
| 824 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 825 EXPECT_TRUE(peer_connection_->GetStats( | |
| 826 observer, track, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 827 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 828 EXPECT_NE(0, observer->timestamp()); | |
| 829 return observer->BytesReceived(); | |
| 830 } | |
| 831 | |
| 832 int GetBytesSentStats(webrtc::MediaStreamTrackInterface* track) { | |
| 833 rtc::scoped_refptr<MockStatsObserver> | |
| 834 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 835 EXPECT_TRUE(peer_connection_->GetStats( | |
| 836 observer, track, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 837 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 838 EXPECT_NE(0, observer->timestamp()); | |
| 839 return observer->BytesSent(); | |
| 840 } | |
| 841 | |
| 842 int GetAvailableReceivedBandwidthStats() { | |
| 843 rtc::scoped_refptr<MockStatsObserver> | |
| 844 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 845 EXPECT_TRUE(peer_connection_->GetStats( | |
| 846 observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 847 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 848 EXPECT_NE(0, observer->timestamp()); | |
| 849 int bw = observer->AvailableReceiveBandwidth(); | |
| 850 return bw; | |
| 851 } | |
| 852 | |
| 853 std::string GetDtlsCipherStats() { | |
| 854 rtc::scoped_refptr<MockStatsObserver> | |
| 855 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 856 EXPECT_TRUE(peer_connection_->GetStats( | |
| 857 observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 858 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 859 EXPECT_NE(0, observer->timestamp()); | |
| 860 return observer->DtlsCipher(); | |
| 861 } | |
| 862 | |
| 863 std::string GetSrtpCipherStats() { | |
| 864 rtc::scoped_refptr<MockStatsObserver> | |
| 865 observer(new rtc::RefCountedObject<MockStatsObserver>()); | |
| 866 EXPECT_TRUE(peer_connection_->GetStats( | |
| 867 observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
| 868 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 869 EXPECT_NE(0, observer->timestamp()); | |
| 870 return observer->SrtpCipher(); | |
| 871 } | |
| 872 | |
| 873 int rendered_width() { | |
| 874 EXPECT_FALSE(fake_video_renderers_.empty()); | |
| 875 return fake_video_renderers_.empty() ? 1 : | |
| 876 fake_video_renderers_.begin()->second->width(); | |
| 877 } | |
| 878 | |
| 879 int rendered_height() { | |
| 880 EXPECT_FALSE(fake_video_renderers_.empty()); | |
| 881 return fake_video_renderers_.empty() ? 1 : | |
| 882 fake_video_renderers_.begin()->second->height(); | |
| 883 } | |
| 884 | |
| 885 webrtc::VideoRotation rendered_rotation() { | |
| 886 EXPECT_FALSE(fake_video_renderers_.empty()); | |
| 887 return fake_video_renderers_.empty() | |
| 888 ? webrtc::kVideoRotation_0 | |
| 889 : fake_video_renderers_.begin()->second->rotation(); | |
| 890 } | |
| 891 | |
| 892 int local_rendered_width() { | |
| 893 return local_video_renderer_ ? local_video_renderer_->width() : 1; | |
| 894 } | |
| 895 | |
| 896 int local_rendered_height() { | |
| 897 return local_video_renderer_ ? local_video_renderer_->height() : 1; | |
| 898 } | |
| 899 | |
| 900 size_t number_of_remote_streams() { | |
| 901 if (!pc()) | |
| 902 return 0; | |
| 903 return pc()->remote_streams()->count(); | |
| 904 } | |
| 905 | |
| 906 StreamCollectionInterface* remote_streams() const { | |
| 907 if (!pc()) { | |
| 908 ADD_FAILURE(); | |
| 909 return nullptr; | |
| 910 } | |
| 911 return pc()->remote_streams(); | |
| 912 } | |
| 913 | |
| 914 StreamCollectionInterface* local_streams() { | |
| 915 if (!pc()) { | |
| 916 ADD_FAILURE(); | |
| 917 return nullptr; | |
| 918 } | |
| 919 return pc()->local_streams(); | |
| 920 } | |
| 921 | |
| 922 bool HasLocalAudioTrack() { return StreamsHaveAudioTrack(local_streams()); } | |
| 923 | |
| 924 bool HasLocalVideoTrack() { return StreamsHaveVideoTrack(local_streams()); } | |
| 925 | |
| 926 webrtc::PeerConnectionInterface::SignalingState signaling_state() { | |
| 927 return pc()->signaling_state(); | |
| 928 } | |
| 929 | |
| 930 webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() { | |
| 931 return pc()->ice_connection_state(); | |
| 932 } | |
| 933 | |
| 934 webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { | |
| 935 return pc()->ice_gathering_state(); | |
| 936 } | |
| 937 | |
| 938 std::vector<std::unique_ptr<MockRtpReceiverObserver>> const& | |
| 939 rtp_receiver_observers() { | |
| 940 return rtp_receiver_observers_; | |
| 941 } | |
| 942 | |
| 943 void SetRtpReceiverObservers() { | |
| 944 rtp_receiver_observers_.clear(); | |
| 945 for (auto receiver : pc()->GetReceivers()) { | |
| 946 std::unique_ptr<MockRtpReceiverObserver> observer( | |
| 947 new MockRtpReceiverObserver(receiver->media_type())); | |
| 948 receiver->SetObserver(observer.get()); | |
| 949 rtp_receiver_observers_.push_back(std::move(observer)); | |
| 950 } | |
| 951 } | |
| 952 | |
| 953 private: | |
| 954 class DummyDtmfObserver : public DtmfSenderObserverInterface { | |
| 955 public: | |
| 956 DummyDtmfObserver() : completed_(false) {} | |
| 957 | |
| 958 // Implements DtmfSenderObserverInterface. | |
| 959 void OnToneChange(const std::string& tone) override { | |
| 960 tones_.push_back(tone); | |
| 961 if (tone.empty()) { | |
| 962 completed_ = true; | |
| 963 } | |
| 964 } | |
| 965 | |
| 966 void Verify(const std::vector<std::string>& tones) const { | |
| 967 ASSERT_TRUE(tones_.size() == tones.size()); | |
| 968 EXPECT_TRUE(std::equal(tones.begin(), tones.end(), tones_.begin())); | |
| 969 } | |
| 970 | |
| 971 bool completed() const { return completed_; } | |
| 972 | |
| 973 private: | |
| 974 bool completed_; | |
| 975 std::vector<std::string> tones_; | |
| 976 }; | |
| 977 | |
| 978 explicit PeerConnectionTestClient(const std::string& id) : id_(id) {} | |
| 979 | |
| 980 bool Init( | |
| 981 const MediaConstraintsInterface* constraints, | |
| 982 const PeerConnectionFactory::Options* options, | |
| 983 const PeerConnectionInterface::RTCConfiguration* config, | |
| 984 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | |
| 985 bool prefer_constraint_apis, | |
| 986 rtc::Thread* network_thread, | |
| 987 rtc::Thread* worker_thread) { | |
| 988 EXPECT_TRUE(!peer_connection_); | |
| 989 EXPECT_TRUE(!peer_connection_factory_); | |
| 990 if (!prefer_constraint_apis) { | |
| 991 EXPECT_TRUE(!constraints); | |
| 992 } | |
| 993 prefer_constraint_apis_ = prefer_constraint_apis; | |
| 994 | |
| 995 fake_network_manager_.reset(new rtc::FakeNetworkManager()); | |
| 996 fake_network_manager_->AddInterface(rtc::SocketAddress("192.168.1.1", 0)); | |
| 997 | |
| 998 std::unique_ptr<cricket::PortAllocator> port_allocator( | |
| 999 new cricket::BasicPortAllocator(fake_network_manager_.get())); | |
| 1000 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); | |
| 1001 | |
| 1002 if (fake_audio_capture_module_ == nullptr) { | |
| 1003 return false; | |
| 1004 } | |
| 1005 fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory(); | |
| 1006 fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory(); | |
| 1007 rtc::Thread* const signaling_thread = rtc::Thread::Current(); | |
| 1008 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( | |
| 1009 network_thread, worker_thread, signaling_thread, | |
| 1010 fake_audio_capture_module_, fake_video_encoder_factory_, | |
| 1011 fake_video_decoder_factory_); | |
| 1012 if (!peer_connection_factory_) { | |
| 1013 return false; | |
| 1014 } | |
| 1015 if (options) { | |
| 1016 peer_connection_factory_->SetOptions(*options); | |
| 1017 } | |
| 1018 peer_connection_ = | |
| 1019 CreatePeerConnection(std::move(port_allocator), constraints, config, | |
| 1020 std::move(cert_generator)); | |
| 1021 return peer_connection_.get() != nullptr; | |
| 1022 } | |
| 1023 | |
| 1024 rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( | |
| 1025 std::unique_ptr<cricket::PortAllocator> port_allocator, | |
| 1026 const MediaConstraintsInterface* constraints, | |
| 1027 const PeerConnectionInterface::RTCConfiguration* config, | |
| 1028 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) { | |
| 1029 // CreatePeerConnection with RTCConfiguration. | |
| 1030 PeerConnectionInterface::RTCConfiguration default_config; | |
| 1031 | |
| 1032 if (!config) { | |
| 1033 config = &default_config; | |
| 1034 } | |
| 1035 | |
| 1036 return peer_connection_factory_->CreatePeerConnection( | |
| 1037 *config, constraints, std::move(port_allocator), | |
| 1038 std::move(cert_generator), this); | |
| 1039 } | |
| 1040 | |
| 1041 void HandleIncomingOffer(const std::string& msg) { | |
| 1042 LOG(INFO) << id_ << "HandleIncomingOffer "; | |
| 1043 if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) { | |
| 1044 // If we are not sending any streams ourselves it is time to add some. | |
| 1045 AddMediaStream(true, true); | |
| 1046 } | |
| 1047 std::unique_ptr<SessionDescriptionInterface> desc( | |
| 1048 webrtc::CreateSessionDescription("offer", msg, nullptr)); | |
| 1049 | |
| 1050 // Do the equivalent of setting the port to 0, adding a=bundle-only, and | |
| 1051 // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all but | |
| 1052 // the first m= section. | |
| 1053 if (make_spec_compliant_max_bundle_offer_) { | |
| 1054 bool first = true; | |
| 1055 for (cricket::ContentInfo& content : desc->description()->contents()) { | |
| 1056 if (first) { | |
| 1057 first = false; | |
| 1058 continue; | |
| 1059 } | |
| 1060 content.bundle_only = true; | |
| 1061 } | |
| 1062 first = true; | |
| 1063 for (cricket::TransportInfo& transport : | |
| 1064 desc->description()->transport_infos()) { | |
| 1065 if (first) { | |
| 1066 first = false; | |
| 1067 continue; | |
| 1068 } | |
| 1069 transport.description.ice_ufrag.clear(); | |
| 1070 transport.description.ice_pwd.clear(); | |
| 1071 transport.description.connection_role = cricket::CONNECTIONROLE_NONE; | |
| 1072 transport.description.identity_fingerprint.reset(nullptr); | |
| 1073 } | |
| 1074 } | |
| 1075 | |
| 1076 EXPECT_TRUE(DoSetRemoteDescription(desc.release())); | |
| 1077 // Set the RtpReceiverObserver after receivers are created. | |
| 1078 SetRtpReceiverObservers(); | |
| 1079 std::unique_ptr<SessionDescriptionInterface> answer; | |
| 1080 EXPECT_TRUE(DoCreateAnswer(&answer)); | |
| 1081 std::string sdp; | |
| 1082 EXPECT_TRUE(answer->ToString(&sdp)); | |
| 1083 EXPECT_TRUE(DoSetLocalDescription(answer.release())); | |
| 1084 SendSdpMessage(webrtc::SessionDescriptionInterface::kAnswer, sdp); | |
| 1085 } | |
| 1086 | |
| 1087 void HandleIncomingAnswer(const std::string& msg) { | |
| 1088 LOG(INFO) << id_ << "HandleIncomingAnswer"; | |
| 1089 std::unique_ptr<SessionDescriptionInterface> desc( | |
| 1090 webrtc::CreateSessionDescription("answer", msg, nullptr)); | |
| 1091 EXPECT_TRUE(DoSetRemoteDescription(desc.release())); | |
| 1092 // Set the RtpReceiverObserver after receivers are created. | |
| 1093 SetRtpReceiverObservers(); | |
| 1094 } | |
| 1095 | |
| 1096 bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc, | |
| 1097 bool offer) { | |
| 1098 rtc::scoped_refptr<MockCreateSessionDescriptionObserver> | |
| 1099 observer(new rtc::RefCountedObject< | |
| 1100 MockCreateSessionDescriptionObserver>()); | |
| 1101 if (prefer_constraint_apis_) { | |
| 1102 if (offer) { | |
| 1103 pc()->CreateOffer(observer, &offer_answer_constraints_); | |
| 1104 } else { | |
| 1105 pc()->CreateAnswer(observer, &offer_answer_constraints_); | |
| 1106 } | |
| 1107 } else { | |
| 1108 if (offer) { | |
| 1109 pc()->CreateOffer(observer, offer_answer_options_); | |
| 1110 } else { | |
| 1111 pc()->CreateAnswer(observer, offer_answer_options_); | |
| 1112 } | |
| 1113 } | |
| 1114 EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs); | |
| 1115 desc->reset(observer->release_desc()); | |
| 1116 if (observer->result() && ExpectIceRestart()) { | |
| 1117 EXPECT_EQ(0u, (*desc)->candidates(0)->count()); | |
| 1118 } | |
| 1119 return observer->result(); | |
| 1120 } | |
| 1121 | |
| 1122 bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) { | |
| 1123 return DoCreateOfferAnswer(desc, true); | |
| 1124 } | |
| 1125 | |
| 1126 bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) { | |
| 1127 return DoCreateOfferAnswer(desc, false); | |
| 1128 } | |
| 1129 | |
| 1130 bool DoSetLocalDescription(SessionDescriptionInterface* desc) { | |
| 1131 rtc::scoped_refptr<MockSetSessionDescriptionObserver> | |
| 1132 observer(new rtc::RefCountedObject< | |
| 1133 MockSetSessionDescriptionObserver>()); | |
| 1134 LOG(INFO) << id_ << "SetLocalDescription "; | |
| 1135 pc()->SetLocalDescription(observer, desc); | |
| 1136 // Ignore the observer result. If we wait for the result with | |
| 1137 // EXPECT_TRUE_WAIT, local ice candidates might be sent to the remote peer | |
| 1138 // before the offer which is an error. | |
| 1139 // The reason is that EXPECT_TRUE_WAIT uses | |
| 1140 // rtc::Thread::Current()->ProcessMessages(1); | |
| 1141 // ProcessMessages waits at least 1ms but processes all messages before | |
| 1142 // returning. Since this test is synchronous and send messages to the remote | |
| 1143 // peer whenever a callback is invoked, this can lead to messages being | |
| 1144 // sent to the remote peer in the wrong order. | |
| 1145 // TODO(perkj): Find a way to check the result without risking that the | |
| 1146 // order of sent messages are changed. Ex- by posting all messages that are | |
| 1147 // sent to the remote peer. | |
| 1148 return true; | |
| 1149 } | |
| 1150 | |
| 1151 bool DoSetRemoteDescription(SessionDescriptionInterface* desc) { | |
| 1152 rtc::scoped_refptr<MockSetSessionDescriptionObserver> | |
| 1153 observer(new rtc::RefCountedObject< | |
| 1154 MockSetSessionDescriptionObserver>()); | |
| 1155 LOG(INFO) << id_ << "SetRemoteDescription "; | |
| 1156 pc()->SetRemoteDescription(observer, desc); | |
| 1157 EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs); | |
| 1158 return observer->result(); | |
| 1159 } | |
| 1160 | |
| 1161 // This modifies all received SDP messages before they are processed. | |
| 1162 void FilterIncomingSdpMessage(std::string* sdp) { | |
| 1163 if (remove_msid_) { | |
| 1164 const char kSdpSsrcAttribute[] = "a=ssrc:"; | |
| 1165 RemoveLinesFromSdp(kSdpSsrcAttribute, sdp); | |
| 1166 const char kSdpMsidSupportedAttribute[] = "a=msid-semantic:"; | |
| 1167 RemoveLinesFromSdp(kSdpMsidSupportedAttribute, sdp); | |
| 1168 } | |
| 1169 if (remove_bundle_) { | |
| 1170 const char kSdpBundleAttribute[] = "a=group:BUNDLE"; | |
| 1171 RemoveLinesFromSdp(kSdpBundleAttribute, sdp); | |
| 1172 } | |
| 1173 if (remove_sdes_) { | |
| 1174 const char kSdpSdesCryptoAttribute[] = "a=crypto"; | |
| 1175 RemoveLinesFromSdp(kSdpSdesCryptoAttribute, sdp); | |
| 1176 } | |
| 1177 if (remove_cvo_) { | |
| 1178 const char kSdpCvoExtenstion[] = "urn:3gpp:video-orientation"; | |
| 1179 RemoveLinesFromSdp(kSdpCvoExtenstion, sdp); | |
| 1180 } | |
| 1181 } | |
| 1182 | |
| 1183 std::string id_; | |
| 1184 | |
| 1185 std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_; | |
| 1186 | |
| 1187 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | |
| 1188 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | |
| 1189 peer_connection_factory_; | |
| 1190 | |
| 1191 bool prefer_constraint_apis_ = true; | |
| 1192 bool auto_add_stream_ = true; | |
| 1193 | |
| 1194 typedef std::pair<std::string, std::string> IceUfragPwdPair; | |
| 1195 std::map<int, IceUfragPwdPair> ice_ufrag_pwd_; | |
| 1196 bool expect_ice_restart_ = false; | |
| 1197 bool expect_ice_renomination_ = false; | |
| 1198 bool expect_remote_ice_renomination_ = false; | |
| 1199 | |
| 1200 // Needed to keep track of number of frames sent. | |
| 1201 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | |
| 1202 // Needed to keep track of number of frames received. | |
| 1203 std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>> | |
| 1204 fake_video_renderers_; | |
| 1205 // Needed to ensure frames aren't received for removed tracks. | |
| 1206 std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>> | |
| 1207 removed_fake_video_renderers_; | |
| 1208 // Needed to keep track of number of frames received when external decoder | |
| 1209 // used. | |
| 1210 FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr; | |
| 1211 FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr; | |
| 1212 bool video_decoder_factory_enabled_ = false; | |
| 1213 webrtc::FakeConstraints video_constraints_; | |
| 1214 | |
| 1215 // For remote peer communication. | |
| 1216 SignalingMessageReceiver* signaling_message_receiver_ = nullptr; | |
| 1217 int signaling_delay_ms_ = 0; | |
| 1218 | |
| 1219 // Store references to the video capturers we've created, so that we can stop | |
| 1220 // them, if required. | |
| 1221 std::vector<cricket::FakeVideoCapturer*> video_capturers_; | |
| 1222 webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0; | |
| 1223 // |local_video_renderer_| attached to the first created local video track. | |
| 1224 std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_; | |
| 1225 | |
| 1226 webrtc::FakeConstraints offer_answer_constraints_; | |
| 1227 PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_; | |
| 1228 bool remove_msid_ = false; // True if MSID should be removed in received SDP. | |
| 1229 bool remove_bundle_ = | |
| 1230 false; // True if bundle should be removed in received SDP. | |
| 1231 bool remove_sdes_ = | |
| 1232 false; // True if a=crypto should be removed in received SDP. | |
| 1233 // |remove_cvo_| is true if extension urn:3gpp:video-orientation should be | |
| 1234 // removed in the received SDP. | |
| 1235 bool remove_cvo_ = false; | |
| 1236 // See LocalP2PTestWithSpecCompliantMaxBundleOffer. | |
| 1237 bool make_spec_compliant_max_bundle_offer_ = false; | |
| 1238 | |
| 1239 rtc::scoped_refptr<DataChannelInterface> data_channel_; | |
| 1240 std::unique_ptr<MockDataChannelObserver> data_observer_; | |
| 1241 | |
| 1242 std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_; | |
| 1243 }; | |
| 1244 | |
| 1245 class P2PTestConductor : public testing::Test { | |
| 1246 public: | |
| 1247 P2PTestConductor() | |
| 1248 : pss_(new rtc::PhysicalSocketServer), | |
| 1249 ss_(new rtc::VirtualSocketServer(pss_.get())), | |
| 1250 network_thread_(new rtc::Thread(ss_.get())), | |
| 1251 worker_thread_(rtc::Thread::Create()) { | |
| 1252 RTC_CHECK(network_thread_->Start()); | |
| 1253 RTC_CHECK(worker_thread_->Start()); | |
| 1254 } | |
| 1255 | |
| 1256 bool SessionActive() { | |
| 1257 return initiating_client_->SessionActive() && | |
| 1258 receiving_client_->SessionActive(); | |
| 1259 } | |
| 1260 | |
| 1261 // Return true if the number of frames provided have been received | |
| 1262 // on the video and audio tracks provided. | |
| 1263 bool FramesHaveArrived(int audio_frames_to_receive, | |
| 1264 int video_frames_to_receive) { | |
| 1265 bool all_good = true; | |
| 1266 if (initiating_client_->HasLocalAudioTrack() && | |
| 1267 receiving_client_->can_receive_audio()) { | |
| 1268 all_good &= | |
| 1269 receiving_client_->AudioFramesReceivedCheck(audio_frames_to_receive); | |
| 1270 } | |
| 1271 if (initiating_client_->HasLocalVideoTrack() && | |
| 1272 receiving_client_->can_receive_video()) { | |
| 1273 all_good &= | |
| 1274 receiving_client_->VideoFramesReceivedCheck(video_frames_to_receive); | |
| 1275 } | |
| 1276 if (receiving_client_->HasLocalAudioTrack() && | |
| 1277 initiating_client_->can_receive_audio()) { | |
| 1278 all_good &= | |
| 1279 initiating_client_->AudioFramesReceivedCheck(audio_frames_to_receive); | |
| 1280 } | |
| 1281 if (receiving_client_->HasLocalVideoTrack() && | |
| 1282 initiating_client_->can_receive_video()) { | |
| 1283 all_good &= | |
| 1284 initiating_client_->VideoFramesReceivedCheck(video_frames_to_receive); | |
| 1285 } | |
| 1286 return all_good; | |
| 1287 } | |
| 1288 | |
| 1289 void VerifyDtmf() { | |
| 1290 initiating_client_->VerifyDtmf(); | |
| 1291 receiving_client_->VerifyDtmf(); | |
| 1292 } | |
| 1293 | |
| 1294 void TestUpdateOfferWithRejectedContent() { | |
| 1295 // Renegotiate, rejecting the video m-line. | |
| 1296 initiating_client_->Negotiate(true, false); | |
| 1297 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | |
| 1298 | |
| 1299 int pc1_audio_received = initiating_client_->audio_frames_received(); | |
| 1300 int pc1_video_received = initiating_client_->video_frames_received(); | |
| 1301 int pc2_audio_received = receiving_client_->audio_frames_received(); | |
| 1302 int pc2_video_received = receiving_client_->video_frames_received(); | |
| 1303 | |
| 1304 // Wait for some additional audio frames to be received. | |
| 1305 EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck( | |
| 1306 pc1_audio_received + kEndAudioFrameCount) && | |
| 1307 receiving_client_->AudioFramesReceivedCheck( | |
| 1308 pc2_audio_received + kEndAudioFrameCount), | |
| 1309 kMaxWaitForFramesMs); | |
| 1310 | |
| 1311 // During this time, we shouldn't have received any additional video frames | |
| 1312 // for the rejected video tracks. | |
| 1313 EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received()); | |
| 1314 EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received()); | |
| 1315 } | |
| 1316 | |
| 1317 void VerifyRenderedAspectRatio(int width, int height) { | |
| 1318 VerifyRenderedAspectRatio(width, height, webrtc::kVideoRotation_0); | |
| 1319 } | |
| 1320 | |
| 1321 void VerifyRenderedAspectRatio(int width, | |
| 1322 int height, | |
| 1323 webrtc::VideoRotation rotation) { | |
| 1324 double expected_aspect_ratio = static_cast<double>(width) / height; | |
| 1325 double receiving_client_rendered_aspect_ratio = | |
| 1326 static_cast<double>(receiving_client()->rendered_width()) / | |
| 1327 receiving_client()->rendered_height(); | |
| 1328 double initializing_client_rendered_aspect_ratio = | |
| 1329 static_cast<double>(initializing_client()->rendered_width()) / | |
| 1330 initializing_client()->rendered_height(); | |
| 1331 double initializing_client_local_rendered_aspect_ratio = | |
| 1332 static_cast<double>(initializing_client()->local_rendered_width()) / | |
| 1333 initializing_client()->local_rendered_height(); | |
| 1334 // Verify end-to-end rendered aspect ratio. | |
| 1335 EXPECT_EQ(expected_aspect_ratio, receiving_client_rendered_aspect_ratio); | |
| 1336 EXPECT_EQ(expected_aspect_ratio, initializing_client_rendered_aspect_ratio); | |
| 1337 // Verify aspect ratio of the local preview. | |
| 1338 EXPECT_EQ(expected_aspect_ratio, | |
| 1339 initializing_client_local_rendered_aspect_ratio); | |
| 1340 | |
| 1341 // Verify rotation. | |
| 1342 EXPECT_EQ(rotation, receiving_client()->rendered_rotation()); | |
| 1343 EXPECT_EQ(rotation, initializing_client()->rendered_rotation()); | |
| 1344 } | |
| 1345 | |
| 1346 void VerifySessionDescriptions() { | |
| 1347 initiating_client_->VerifyRejectedMediaInSessionDescription(); | |
| 1348 receiving_client_->VerifyRejectedMediaInSessionDescription(); | |
| 1349 initiating_client_->VerifyLocalIceUfragAndPassword(); | |
| 1350 receiving_client_->VerifyLocalIceUfragAndPassword(); | |
| 1351 } | |
| 1352 | |
| 1353 ~P2PTestConductor() { | |
| 1354 if (initiating_client_) { | |
| 1355 initiating_client_->set_signaling_message_receiver(nullptr); | |
| 1356 } | |
| 1357 if (receiving_client_) { | |
| 1358 receiving_client_->set_signaling_message_receiver(nullptr); | |
| 1359 } | |
| 1360 } | |
| 1361 | |
| 1362 bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); } | |
| 1363 | |
| 1364 bool CreateTestClients(MediaConstraintsInterface* init_constraints, | |
| 1365 MediaConstraintsInterface* recv_constraints) { | |
| 1366 return CreateTestClients(init_constraints, nullptr, nullptr, | |
| 1367 recv_constraints, nullptr, nullptr); | |
| 1368 } | |
| 1369 | |
| 1370 bool CreateTestClients( | |
| 1371 const PeerConnectionInterface::RTCConfiguration& init_config, | |
| 1372 const PeerConnectionInterface::RTCConfiguration& recv_config) { | |
| 1373 return CreateTestClients(nullptr, nullptr, &init_config, nullptr, nullptr, | |
| 1374 &recv_config); | |
| 1375 } | |
| 1376 | |
| 1377 bool CreateTestClientsThatPreferNoConstraints() { | |
| 1378 initiating_client_.reset( | |
| 1379 PeerConnectionTestClient::CreateClientPreferNoConstraints( | |
| 1380 "Caller: ", nullptr, network_thread_.get(), worker_thread_.get())); | |
| 1381 receiving_client_.reset( | |
| 1382 PeerConnectionTestClient::CreateClientPreferNoConstraints( | |
| 1383 "Callee: ", nullptr, network_thread_.get(), worker_thread_.get())); | |
| 1384 if (!initiating_client_ || !receiving_client_) { | |
| 1385 return false; | |
| 1386 } | |
| 1387 // Remember the choice for possible later resets of the clients. | |
| 1388 prefer_constraint_apis_ = false; | |
| 1389 SetSignalingReceivers(); | |
| 1390 return true; | |
| 1391 } | |
| 1392 | |
| 1393 bool CreateTestClients( | |
| 1394 MediaConstraintsInterface* init_constraints, | |
| 1395 PeerConnectionFactory::Options* init_options, | |
| 1396 const PeerConnectionInterface::RTCConfiguration* init_config, | |
| 1397 MediaConstraintsInterface* recv_constraints, | |
| 1398 PeerConnectionFactory::Options* recv_options, | |
| 1399 const PeerConnectionInterface::RTCConfiguration* recv_config) { | |
| 1400 initiating_client_.reset(PeerConnectionTestClient::CreateClient( | |
| 1401 "Caller: ", init_constraints, init_options, init_config, | |
| 1402 network_thread_.get(), worker_thread_.get())); | |
| 1403 receiving_client_.reset(PeerConnectionTestClient::CreateClient( | |
| 1404 "Callee: ", recv_constraints, recv_options, recv_config, | |
| 1405 network_thread_.get(), worker_thread_.get())); | |
| 1406 if (!initiating_client_ || !receiving_client_) { | |
| 1407 return false; | |
| 1408 } | |
| 1409 SetSignalingReceivers(); | |
| 1410 return true; | |
| 1411 } | |
| 1412 | |
| 1413 void SetSignalingReceivers() { | |
| 1414 initiating_client_->set_signaling_message_receiver(receiving_client_.get()); | |
| 1415 receiving_client_->set_signaling_message_receiver(initiating_client_.get()); | |
| 1416 } | |
| 1417 | |
| 1418 void SetSignalingDelayMs(int delay_ms) { | |
| 1419 initiating_client_->set_signaling_delay_ms(delay_ms); | |
| 1420 receiving_client_->set_signaling_delay_ms(delay_ms); | |
| 1421 } | |
| 1422 | |
| 1423 void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints, | |
| 1424 const webrtc::FakeConstraints& recv_constraints) { | |
| 1425 initiating_client_->SetVideoConstraints(init_constraints); | |
| 1426 receiving_client_->SetVideoConstraints(recv_constraints); | |
| 1427 } | |
| 1428 | |
| 1429 void SetCaptureRotation(webrtc::VideoRotation rotation) { | |
| 1430 initiating_client_->SetCaptureRotation(rotation); | |
| 1431 receiving_client_->SetCaptureRotation(rotation); | |
| 1432 } | |
| 1433 | |
| 1434 void EnableVideoDecoderFactory() { | |
| 1435 initiating_client_->EnableVideoDecoderFactory(); | |
| 1436 receiving_client_->EnableVideoDecoderFactory(); | |
| 1437 } | |
| 1438 | |
| 1439 // This test sets up a call between two parties. Both parties send static | |
| 1440 // frames to each other. Once the test is finished the number of sent frames | |
| 1441 // is compared to the number of received frames. | |
| 1442 void LocalP2PTest() { | |
| 1443 if (initiating_client_->NumberOfLocalMediaStreams() == 0) { | |
| 1444 initiating_client_->AddMediaStream(true, true); | |
| 1445 } | |
| 1446 initiating_client_->Negotiate(); | |
| 1447 // Assert true is used here since next tests are guaranteed to fail and | |
| 1448 // would eat up 5 seconds. | |
| 1449 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | |
| 1450 VerifySessionDescriptions(); | |
| 1451 | |
| 1452 int audio_frame_count = kEndAudioFrameCount; | |
| 1453 int video_frame_count = kEndVideoFrameCount; | |
| 1454 // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly. | |
| 1455 | |
| 1456 if ((!initiating_client_->can_receive_audio() && | |
| 1457 !initiating_client_->can_receive_video()) || | |
| 1458 (!receiving_client_->can_receive_audio() && | |
| 1459 !receiving_client_->can_receive_video())) { | |
| 1460 // Neither audio nor video will flow, so connections won't be | |
| 1461 // established. There's nothing more to check. | |
| 1462 // TODO(hta): Check connection if there's a data channel. | |
| 1463 return; | |
| 1464 } | |
| 1465 | |
| 1466 // Audio or video is expected to flow, so both clients should reach the | |
| 1467 // Connected state, and the offerer (ICE controller) should proceed to | |
| 1468 // Completed. | |
| 1469 // Note: These tests have been observed to fail under heavy load at | |
| 1470 // shorter timeouts, so they may be flaky. | |
| 1471 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
| 1472 initiating_client_->ice_connection_state(), | |
| 1473 kMaxWaitForFramesMs); | |
| 1474 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
| 1475 receiving_client_->ice_connection_state(), | |
| 1476 kMaxWaitForFramesMs); | |
| 1477 | |
| 1478 // The ICE gathering state should end up in kIceGatheringComplete, | |
| 1479 // but there's a bug that prevents this at the moment, and the state | |
| 1480 // machine is being updated by the WEBRTC WG. | |
| 1481 // TODO(hta): Update this check when spec revisions finish. | |
| 1482 EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew, | |
| 1483 initiating_client_->ice_gathering_state()); | |
| 1484 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, | |
| 1485 receiving_client_->ice_gathering_state(), | |
| 1486 kMaxWaitForFramesMs); | |
| 1487 | |
| 1488 // Check that the expected number of frames have arrived. | |
| 1489 EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count), | |
| 1490 kMaxWaitForFramesMs); | |
| 1491 } | |
| 1492 | |
| 1493 void SetupAndVerifyDtlsCall() { | |
| 1494 FakeConstraints setup_constraints; | |
| 1495 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
| 1496 true); | |
| 1497 // Disable resolution adaptation, we don't want it interfering with the | |
| 1498 // test results. | |
| 1499 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; | |
| 1500 rtc_config.set_cpu_adaptation(false); | |
| 1501 | |
| 1502 ASSERT_TRUE(CreateTestClients(&setup_constraints, nullptr, &rtc_config, | |
| 1503 &setup_constraints, nullptr, &rtc_config)); | |
| 1504 LocalP2PTest(); | |
| 1505 VerifyRenderedAspectRatio(640, 480); | |
| 1506 } | |
| 1507 | |
| 1508 PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() { | |
| 1509 FakeConstraints setup_constraints; | |
| 1510 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
| 1511 true); | |
| 1512 // Disable resolution adaptation, we don't want it interfering with the | |
| 1513 // test results. | |
| 1514 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; | |
| 1515 rtc_config.set_cpu_adaptation(false); | |
| 1516 | |
| 1517 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
| 1518 new FakeRTCCertificateGenerator()); | |
| 1519 cert_generator->use_alternate_key(); | |
| 1520 | |
| 1521 // Make sure the new client is using a different certificate. | |
| 1522 return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore( | |
| 1523 "New Peer: ", &setup_constraints, nullptr, &rtc_config, | |
| 1524 std::move(cert_generator), prefer_constraint_apis_, | |
| 1525 network_thread_.get(), worker_thread_.get()); | |
| 1526 } | |
| 1527 | |
| 1528 void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) { | |
| 1529 // Messages may get lost on the unreliable DataChannel, so we send multiple | |
| 1530 // times to avoid test flakiness. | |
| 1531 static const size_t kSendAttempts = 5; | |
| 1532 | |
| 1533 for (size_t i = 0; i < kSendAttempts; ++i) { | |
| 1534 dc->Send(DataBuffer(data)); | |
| 1535 } | |
| 1536 } | |
| 1537 | |
| 1538 rtc::Thread* network_thread() { return network_thread_.get(); } | |
| 1539 | |
| 1540 rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); } | |
| 1541 | |
| 1542 PeerConnectionTestClient* initializing_client() { | |
| 1543 return initiating_client_.get(); | |
| 1544 } | |
| 1545 | |
| 1546 // Set the |initiating_client_| to the |client| passed in and return the | |
| 1547 // original |initiating_client_|. | |
| 1548 PeerConnectionTestClient* set_initializing_client( | |
| 1549 PeerConnectionTestClient* client) { | |
| 1550 PeerConnectionTestClient* old = initiating_client_.release(); | |
| 1551 initiating_client_.reset(client); | |
| 1552 return old; | |
| 1553 } | |
| 1554 | |
| 1555 PeerConnectionTestClient* receiving_client() { | |
| 1556 return receiving_client_.get(); | |
| 1557 } | |
| 1558 | |
| 1559 // Set the |receiving_client_| to the |client| passed in and return the | |
| 1560 // original |receiving_client_|. | |
| 1561 PeerConnectionTestClient* set_receiving_client( | |
| 1562 PeerConnectionTestClient* client) { | |
| 1563 PeerConnectionTestClient* old = receiving_client_.release(); | |
| 1564 receiving_client_.reset(client); | |
| 1565 return old; | |
| 1566 } | |
| 1567 | |
| 1568 bool AllObserversReceived( | |
| 1569 const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& observers) { | |
| 1570 for (auto& observer : observers) { | |
| 1571 if (!observer->first_packet_received()) { | |
| 1572 return false; | |
| 1573 } | |
| 1574 } | |
| 1575 return true; | |
| 1576 } | |
| 1577 | |
| 1578 void TestGcmNegotiation(bool local_gcm_enabled, bool remote_gcm_enabled, | |
| 1579 int expected_cipher_suite) { | |
| 1580 PeerConnectionFactory::Options init_options; | |
| 1581 init_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled; | |
| 1582 PeerConnectionFactory::Options recv_options; | |
| 1583 recv_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled; | |
| 1584 ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr, | |
| 1585 &recv_options, nullptr)); | |
| 1586 rtc::scoped_refptr<webrtc::FakeMetricsObserver> | |
| 1587 init_observer = | |
| 1588 new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
| 1589 initializing_client()->pc()->RegisterUMAObserver(init_observer); | |
| 1590 LocalP2PTest(); | |
| 1591 | |
| 1592 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite), | |
| 1593 initializing_client()->GetSrtpCipherStats(), | |
| 1594 kMaxWaitMs); | |
| 1595 EXPECT_EQ(1, | |
| 1596 init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
| 1597 expected_cipher_suite)); | |
| 1598 } | |
| 1599 | |
| 1600 private: | |
| 1601 // |ss_| is used by |network_thread_| so it must be destroyed later. | |
| 1602 std::unique_ptr<rtc::PhysicalSocketServer> pss_; | |
| 1603 std::unique_ptr<rtc::VirtualSocketServer> ss_; | |
| 1604 // |network_thread_| and |worker_thread_| are used by both | |
| 1605 // |initiating_client_| and |receiving_client_| so they must be destroyed | |
| 1606 // later. | |
| 1607 std::unique_ptr<rtc::Thread> network_thread_; | |
| 1608 std::unique_ptr<rtc::Thread> worker_thread_; | |
| 1609 std::unique_ptr<PeerConnectionTestClient> initiating_client_; | |
| 1610 std::unique_ptr<PeerConnectionTestClient> receiving_client_; | |
| 1611 bool prefer_constraint_apis_ = true; | |
| 1612 }; | |
| 1613 | |
| 1614 // Disable for TSan v2, see | |
| 1615 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. | |
| 1616 #if !defined(THREAD_SANITIZER) | |
| 1617 | |
| 1618 TEST_F(P2PTestConductor, TestRtpReceiverObserverCallbackFunction) { | |
| 1619 ASSERT_TRUE(CreateTestClients()); | |
| 1620 LocalP2PTest(); | |
| 1621 EXPECT_TRUE_WAIT( | |
| 1622 AllObserversReceived(initializing_client()->rtp_receiver_observers()), | |
| 1623 kMaxWaitForFramesMs); | |
| 1624 EXPECT_TRUE_WAIT( | |
| 1625 AllObserversReceived(receiving_client()->rtp_receiver_observers()), | |
| 1626 kMaxWaitForFramesMs); | |
| 1627 } | |
| 1628 | |
| 1629 // The observers are expected to fire the signal even if they are set after the | |
| 1630 // first packet is received. | |
| 1631 TEST_F(P2PTestConductor, TestSetRtpReceiverObserverAfterFirstPacketIsReceived) { | |
| 1632 ASSERT_TRUE(CreateTestClients()); | |
| 1633 LocalP2PTest(); | |
| 1634 // Reset the RtpReceiverObservers. | |
| 1635 initializing_client()->SetRtpReceiverObservers(); | |
| 1636 receiving_client()->SetRtpReceiverObservers(); | |
| 1637 EXPECT_TRUE_WAIT( | |
| 1638 AllObserversReceived(initializing_client()->rtp_receiver_observers()), | |
| 1639 kMaxWaitForFramesMs); | |
| 1640 EXPECT_TRUE_WAIT( | |
| 1641 AllObserversReceived(receiving_client()->rtp_receiver_observers()), | |
| 1642 kMaxWaitForFramesMs); | |
| 1643 } | |
| 1644 | |
| 1645 // This test sets up a Jsep call between two parties and test Dtmf. | |
| 1646 // TODO(holmer): Disabled due to sometimes crashing on buildbots. | |
| 1647 // See issue webrtc/2378. | |
| 1648 TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) { | |
| 1649 ASSERT_TRUE(CreateTestClients()); | |
| 1650 LocalP2PTest(); | |
| 1651 VerifyDtmf(); | |
| 1652 } | |
| 1653 | |
| 1654 // This test sets up a Jsep call between two parties and test that we can get a | |
| 1655 // video aspect ratio of 16:9. | |
| 1656 TEST_F(P2PTestConductor, LocalP2PTest16To9) { | |
| 1657 ASSERT_TRUE(CreateTestClients()); | |
| 1658 FakeConstraints constraint; | |
| 1659 double requested_ratio = 640.0/360; | |
| 1660 constraint.SetMandatoryMinAspectRatio(requested_ratio); | |
| 1661 SetVideoConstraints(constraint, constraint); | |
| 1662 LocalP2PTest(); | |
| 1663 | |
| 1664 ASSERT_LE(0, initializing_client()->rendered_height()); | |
| 1665 double initiating_video_ratio = | |
| 1666 static_cast<double>(initializing_client()->rendered_width()) / | |
| 1667 initializing_client()->rendered_height(); | |
| 1668 EXPECT_LE(requested_ratio, initiating_video_ratio); | |
| 1669 | |
| 1670 ASSERT_LE(0, receiving_client()->rendered_height()); | |
| 1671 double receiving_video_ratio = | |
| 1672 static_cast<double>(receiving_client()->rendered_width()) / | |
| 1673 receiving_client()->rendered_height(); | |
| 1674 EXPECT_LE(requested_ratio, receiving_video_ratio); | |
| 1675 } | |
| 1676 | |
| 1677 // This test sets up a Jsep call between two parties and test that the | |
| 1678 // received video has a resolution of 1280*720. | |
| 1679 // TODO(mallinath): Enable when | |
| 1680 // http://code.google.com/p/webrtc/issues/detail?id=981 is fixed. | |
| 1681 TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) { | |
| 1682 ASSERT_TRUE(CreateTestClients()); | |
| 1683 FakeConstraints constraint; | |
| 1684 constraint.SetMandatoryMinWidth(1280); | |
| 1685 constraint.SetMandatoryMinHeight(720); | |
| 1686 SetVideoConstraints(constraint, constraint); | |
| 1687 LocalP2PTest(); | |
| 1688 VerifyRenderedAspectRatio(1280, 720); | |
| 1689 } | |
| 1690 | |
| 1691 // This test sets up a call between two endpoints that are configured to use | |
| 1692 // DTLS key agreement. As a result, DTLS is negotiated and used for transport. | |
| 1693 TEST_F(P2PTestConductor, LocalP2PTestDtls) { | |
| 1694 SetupAndVerifyDtlsCall(); | |
| 1695 } | |
| 1696 | |
| 1697 // This test sets up an one-way call, with media only from initiator to | |
| 1698 // responder. | |
| 1699 TEST_F(P2PTestConductor, OneWayMediaCall) { | |
| 1700 ASSERT_TRUE(CreateTestClients()); | |
| 1701 receiving_client()->set_auto_add_stream(false); | |
| 1702 LocalP2PTest(); | |
| 1703 } | |
| 1704 | |
| 1705 TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) { | |
| 1706 ASSERT_TRUE(CreateTestClientsThatPreferNoConstraints()); | |
| 1707 receiving_client()->set_auto_add_stream(false); | |
| 1708 LocalP2PTest(); | |
| 1709 } | |
| 1710 | |
| 1711 // This test sets up a audio call initially and then upgrades to audio/video, | |
| 1712 // using DTLS. | |
| 1713 TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { | |
| 1714 FakeConstraints setup_constraints; | |
| 1715 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
| 1716 true); | |
| 1717 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 1718 receiving_client()->SetReceiveAudioVideo(true, false); | |
| 1719 LocalP2PTest(); | |
| 1720 receiving_client()->SetReceiveAudioVideo(true, true); | |
| 1721 receiving_client()->Negotiate(); | |
| 1722 } | |
| 1723 | |
| 1724 // This test sets up a call transfer to a new caller with a different DTLS | |
| 1725 // fingerprint. | |
| 1726 TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) { | |
| 1727 SetupAndVerifyDtlsCall(); | |
| 1728 | |
| 1729 // Keeping the original peer around which will still send packets to the | |
| 1730 // receiving client. These SRTP packets will be dropped. | |
| 1731 std::unique_ptr<PeerConnectionTestClient> original_peer( | |
| 1732 set_initializing_client(CreateDtlsClientWithAlternateKey())); | |
| 1733 original_peer->pc()->Close(); | |
| 1734 | |
| 1735 SetSignalingReceivers(); | |
| 1736 receiving_client()->SetExpectIceRestart(true); | |
| 1737 LocalP2PTest(); | |
| 1738 VerifyRenderedAspectRatio(640, 480); | |
| 1739 } | |
| 1740 | |
| 1741 // This test sets up a non-bundle call and apply bundle during ICE restart. When | |
| 1742 // bundle is in effect in the restart, the channel can successfully reset its | |
| 1743 // DTLS-SRTP context. | |
| 1744 TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) { | |
| 1745 FakeConstraints setup_constraints; | |
| 1746 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
| 1747 true); | |
| 1748 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 1749 receiving_client()->RemoveBundleFromReceivedSdp(true); | |
| 1750 LocalP2PTest(); | |
| 1751 VerifyRenderedAspectRatio(640, 480); | |
| 1752 | |
| 1753 initializing_client()->IceRestart(); | |
| 1754 receiving_client()->SetExpectIceRestart(true); | |
| 1755 receiving_client()->RemoveBundleFromReceivedSdp(false); | |
| 1756 LocalP2PTest(); | |
| 1757 VerifyRenderedAspectRatio(640, 480); | |
| 1758 } | |
| 1759 | |
| 1760 // This test sets up a call transfer to a new callee with a different DTLS | |
| 1761 // fingerprint. | |
| 1762 TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) { | |
| 1763 SetupAndVerifyDtlsCall(); | |
| 1764 | |
| 1765 // Keeping the original peer around which will still send packets to the | |
| 1766 // receiving client. These SRTP packets will be dropped. | |
| 1767 std::unique_ptr<PeerConnectionTestClient> original_peer( | |
| 1768 set_receiving_client(CreateDtlsClientWithAlternateKey())); | |
| 1769 original_peer->pc()->Close(); | |
| 1770 | |
| 1771 SetSignalingReceivers(); | |
| 1772 initializing_client()->IceRestart(); | |
| 1773 LocalP2PTest(); | |
| 1774 VerifyRenderedAspectRatio(640, 480); | |
| 1775 } | |
| 1776 | |
| 1777 TEST_F(P2PTestConductor, LocalP2PTestCVO) { | |
| 1778 ASSERT_TRUE(CreateTestClients()); | |
| 1779 SetCaptureRotation(webrtc::kVideoRotation_90); | |
| 1780 LocalP2PTest(); | |
| 1781 VerifyRenderedAspectRatio(640, 480, webrtc::kVideoRotation_90); | |
| 1782 } | |
| 1783 | |
| 1784 TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) { | |
| 1785 ASSERT_TRUE(CreateTestClients()); | |
| 1786 SetCaptureRotation(webrtc::kVideoRotation_90); | |
| 1787 receiving_client()->RemoveCvoFromReceivedSdp(true); | |
| 1788 LocalP2PTest(); | |
| 1789 VerifyRenderedAspectRatio(480, 640, webrtc::kVideoRotation_0); | |
| 1790 } | |
| 1791 | |
| 1792 // This test sets up a call between two endpoints that are configured to use | |
| 1793 // DTLS key agreement. The offerer don't support SDES. As a result, DTLS is | |
| 1794 // negotiated and used for transport. | |
| 1795 TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) { | |
| 1796 FakeConstraints setup_constraints; | |
| 1797 setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
| 1798 true); | |
| 1799 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 1800 receiving_client()->RemoveSdesCryptoFromReceivedSdp(true); | |
| 1801 LocalP2PTest(); | |
| 1802 VerifyRenderedAspectRatio(640, 480); | |
| 1803 } | |
| 1804 | |
| 1805 #ifdef HAVE_SCTP | |
| 1806 // This test verifies that the negotiation will succeed with data channel only | |
| 1807 // in max-bundle mode. | |
| 1808 TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) { | |
| 1809 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; | |
| 1810 rtc_config.bundle_policy = | |
| 1811 webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle; | |
| 1812 ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config)); | |
| 1813 initializing_client()->CreateDataChannel(); | |
| 1814 initializing_client()->Negotiate(); | |
| 1815 } | |
| 1816 #endif | |
| 1817 | |
| 1818 // This test sets up a Jsep call between two parties, and the callee only | |
| 1819 // accept to receive video. | |
| 1820 TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) { | |
| 1821 ASSERT_TRUE(CreateTestClients()); | |
| 1822 receiving_client()->SetReceiveAudioVideo(false, true); | |
| 1823 LocalP2PTest(); | |
| 1824 } | |
| 1825 | |
| 1826 // This test sets up a Jsep call between two parties, and the callee only | |
| 1827 // accept to receive audio. | |
| 1828 TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) { | |
| 1829 ASSERT_TRUE(CreateTestClients()); | |
| 1830 receiving_client()->SetReceiveAudioVideo(true, false); | |
| 1831 LocalP2PTest(); | |
| 1832 } | |
| 1833 | |
| 1834 // This test sets up a Jsep call between two parties, and the callee reject both | |
| 1835 // audio and video. | |
| 1836 TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) { | |
| 1837 ASSERT_TRUE(CreateTestClients()); | |
| 1838 receiving_client()->SetReceiveAudioVideo(false, false); | |
| 1839 LocalP2PTest(); | |
| 1840 } | |
| 1841 | |
| 1842 // This test sets up an audio and video call between two parties. After the call | |
| 1843 // runs for a while (10 frames), the caller sends an update offer with video | |
| 1844 // being rejected. Once the re-negotiation is done, the video flow should stop | |
| 1845 // and the audio flow should continue. | |
| 1846 TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) { | |
| 1847 ASSERT_TRUE(CreateTestClients()); | |
| 1848 LocalP2PTest(); | |
| 1849 TestUpdateOfferWithRejectedContent(); | |
| 1850 } | |
| 1851 | |
| 1852 // This test sets up a Jsep call between two parties. The MSID is removed from | |
| 1853 // the SDP strings from the caller. | |
| 1854 TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) { | |
| 1855 ASSERT_TRUE(CreateTestClients()); | |
| 1856 receiving_client()->RemoveMsidFromReceivedSdp(true); | |
| 1857 // TODO(perkj): Currently there is a bug that cause audio to stop playing if | |
| 1858 // audio and video is muxed when MSID is disabled. Remove | |
| 1859 // SetRemoveBundleFromSdp once | |
| 1860 // https://code.google.com/p/webrtc/issues/detail?id=1193 is fixed. | |
| 1861 receiving_client()->RemoveBundleFromReceivedSdp(true); | |
| 1862 LocalP2PTest(); | |
| 1863 } | |
| 1864 | |
| 1865 TEST_F(P2PTestConductor, LocalP2PTestTwoStreams) { | |
| 1866 ASSERT_TRUE(CreateTestClients()); | |
| 1867 // Set optional video constraint to max 320pixels to decrease CPU usage. | |
| 1868 FakeConstraints constraint; | |
| 1869 constraint.SetOptionalMaxWidth(320); | |
| 1870 SetVideoConstraints(constraint, constraint); | |
| 1871 initializing_client()->AddMediaStream(true, true); | |
| 1872 initializing_client()->AddMediaStream(false, true); | |
| 1873 ASSERT_EQ(2u, initializing_client()->NumberOfLocalMediaStreams()); | |
| 1874 LocalP2PTest(); | |
| 1875 EXPECT_EQ(2u, receiving_client()->number_of_remote_streams()); | |
| 1876 } | |
| 1877 | |
| 1878 // Test that if applying a true "max bundle" offer, which uses ports of 0, | |
| 1879 // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and | |
| 1880 // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes | |
| 1881 // successfully and media flows. | |
| 1882 // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works. | |
| 1883 // TODO(deadbeef): Won't need this test once we start generating actual | |
| 1884 // standards-compliant SDP. | |
| 1885 TEST_F(P2PTestConductor, LocalP2PTestWithSpecCompliantMaxBundleOffer) { | |
| 1886 ASSERT_TRUE(CreateTestClients()); | |
| 1887 receiving_client()->MakeSpecCompliantMaxBundleOfferFromReceivedSdp(true); | |
| 1888 LocalP2PTest(); | |
| 1889 } | |
| 1890 | |
| 1891 // Test that we can receive the audio output level from a remote audio track. | |
| 1892 TEST_F(P2PTestConductor, GetAudioOutputLevelStats) { | |
| 1893 ASSERT_TRUE(CreateTestClients()); | |
| 1894 LocalP2PTest(); | |
| 1895 | |
| 1896 StreamCollectionInterface* remote_streams = | |
| 1897 initializing_client()->remote_streams(); | |
| 1898 ASSERT_GT(remote_streams->count(), 0u); | |
| 1899 ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u); | |
| 1900 MediaStreamTrackInterface* remote_audio_track = | |
| 1901 remote_streams->at(0)->GetAudioTracks()[0]; | |
| 1902 | |
| 1903 // Get the audio output level stats. Note that the level is not available | |
| 1904 // until a RTCP packet has been received. | |
| 1905 EXPECT_TRUE_WAIT( | |
| 1906 initializing_client()->GetAudioOutputLevelStats(remote_audio_track) > 0, | |
| 1907 kMaxWaitForStatsMs); | |
| 1908 } | |
| 1909 | |
| 1910 // Test that an audio input level is reported. | |
| 1911 TEST_F(P2PTestConductor, GetAudioInputLevelStats) { | |
| 1912 ASSERT_TRUE(CreateTestClients()); | |
| 1913 LocalP2PTest(); | |
| 1914 | |
| 1915 // Get the audio input level stats. The level should be available very | |
| 1916 // soon after the test starts. | |
| 1917 EXPECT_TRUE_WAIT(initializing_client()->GetAudioInputLevelStats() > 0, | |
| 1918 kMaxWaitForStatsMs); | |
| 1919 } | |
| 1920 | |
| 1921 // Test that we can get incoming byte counts from both audio and video tracks. | |
| 1922 TEST_F(P2PTestConductor, GetBytesReceivedStats) { | |
| 1923 ASSERT_TRUE(CreateTestClients()); | |
| 1924 LocalP2PTest(); | |
| 1925 | |
| 1926 StreamCollectionInterface* remote_streams = | |
| 1927 initializing_client()->remote_streams(); | |
| 1928 ASSERT_GT(remote_streams->count(), 0u); | |
| 1929 ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u); | |
| 1930 MediaStreamTrackInterface* remote_audio_track = | |
| 1931 remote_streams->at(0)->GetAudioTracks()[0]; | |
| 1932 EXPECT_TRUE_WAIT( | |
| 1933 initializing_client()->GetBytesReceivedStats(remote_audio_track) > 0, | |
| 1934 kMaxWaitForStatsMs); | |
| 1935 | |
| 1936 MediaStreamTrackInterface* remote_video_track = | |
| 1937 remote_streams->at(0)->GetVideoTracks()[0]; | |
| 1938 EXPECT_TRUE_WAIT( | |
| 1939 initializing_client()->GetBytesReceivedStats(remote_video_track) > 0, | |
| 1940 kMaxWaitForStatsMs); | |
| 1941 } | |
| 1942 | |
| 1943 // Test that we can get outgoing byte counts from both audio and video tracks. | |
| 1944 TEST_F(P2PTestConductor, GetBytesSentStats) { | |
| 1945 ASSERT_TRUE(CreateTestClients()); | |
| 1946 LocalP2PTest(); | |
| 1947 | |
| 1948 StreamCollectionInterface* local_streams = | |
| 1949 initializing_client()->local_streams(); | |
| 1950 ASSERT_GT(local_streams->count(), 0u); | |
| 1951 ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u); | |
| 1952 MediaStreamTrackInterface* local_audio_track = | |
| 1953 local_streams->at(0)->GetAudioTracks()[0]; | |
| 1954 EXPECT_TRUE_WAIT( | |
| 1955 initializing_client()->GetBytesSentStats(local_audio_track) > 0, | |
| 1956 kMaxWaitForStatsMs); | |
| 1957 | |
| 1958 MediaStreamTrackInterface* local_video_track = | |
| 1959 local_streams->at(0)->GetVideoTracks()[0]; | |
| 1960 EXPECT_TRUE_WAIT( | |
| 1961 initializing_client()->GetBytesSentStats(local_video_track) > 0, | |
| 1962 kMaxWaitForStatsMs); | |
| 1963 } | |
| 1964 | |
| 1965 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. | |
| 1966 TEST_F(P2PTestConductor, GetDtls12None) { | |
| 1967 PeerConnectionFactory::Options init_options; | |
| 1968 init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
| 1969 PeerConnectionFactory::Options recv_options; | |
| 1970 recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
| 1971 ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr, | |
| 1972 &recv_options, nullptr)); | |
| 1973 rtc::scoped_refptr<webrtc::FakeMetricsObserver> | |
| 1974 init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
| 1975 initializing_client()->pc()->RegisterUMAObserver(init_observer); | |
| 1976 LocalP2PTest(); | |
| 1977 | |
| 1978 EXPECT_TRUE_WAIT( | |
| 1979 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
| 1980 initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT), | |
| 1981 kMaxWaitForStatsMs); | |
| 1982 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
| 1983 initializing_client()->GetSrtpCipherStats(), | |
| 1984 kMaxWaitForStatsMs); | |
| 1985 EXPECT_EQ(1, | |
| 1986 init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
| 1987 kDefaultSrtpCryptoSuite)); | |
| 1988 } | |
| 1989 | |
| 1990 // Test that DTLS 1.2 is used if both ends support it. | |
| 1991 TEST_F(P2PTestConductor, GetDtls12Both) { | |
| 1992 PeerConnectionFactory::Options init_options; | |
| 1993 init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
| 1994 PeerConnectionFactory::Options recv_options; | |
| 1995 recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
| 1996 ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr, | |
| 1997 &recv_options, nullptr)); | |
| 1998 rtc::scoped_refptr<webrtc::FakeMetricsObserver> | |
| 1999 init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
| 2000 initializing_client()->pc()->RegisterUMAObserver(init_observer); | |
| 2001 LocalP2PTest(); | |
| 2002 | |
| 2003 EXPECT_TRUE_WAIT( | |
| 2004 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
| 2005 initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT), | |
| 2006 kMaxWaitForStatsMs); | |
| 2007 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
| 2008 initializing_client()->GetSrtpCipherStats(), | |
| 2009 kMaxWaitForStatsMs); | |
| 2010 EXPECT_EQ(1, | |
| 2011 init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
| 2012 kDefaultSrtpCryptoSuite)); | |
| 2013 } | |
| 2014 | |
| 2015 // Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the | |
| 2016 // received supports 1.0. | |
| 2017 TEST_F(P2PTestConductor, GetDtls12Init) { | |
| 2018 PeerConnectionFactory::Options init_options; | |
| 2019 init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
| 2020 PeerConnectionFactory::Options recv_options; | |
| 2021 recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
| 2022 ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr, | |
| 2023 &recv_options, nullptr)); | |
| 2024 rtc::scoped_refptr<webrtc::FakeMetricsObserver> | |
| 2025 init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
| 2026 initializing_client()->pc()->RegisterUMAObserver(init_observer); | |
| 2027 LocalP2PTest(); | |
| 2028 | |
| 2029 EXPECT_TRUE_WAIT( | |
| 2030 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
| 2031 initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT), | |
| 2032 kMaxWaitForStatsMs); | |
| 2033 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
| 2034 initializing_client()->GetSrtpCipherStats(), | |
| 2035 kMaxWaitForStatsMs); | |
| 2036 EXPECT_EQ(1, | |
| 2037 init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
| 2038 kDefaultSrtpCryptoSuite)); | |
| 2039 } | |
| 2040 | |
| 2041 // Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the | |
| 2042 // received supports 1.2. | |
| 2043 TEST_F(P2PTestConductor, GetDtls12Recv) { | |
| 2044 PeerConnectionFactory::Options init_options; | |
| 2045 init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
| 2046 PeerConnectionFactory::Options recv_options; | |
| 2047 recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
| 2048 ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr, | |
| 2049 &recv_options, nullptr)); | |
| 2050 rtc::scoped_refptr<webrtc::FakeMetricsObserver> | |
| 2051 init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
| 2052 initializing_client()->pc()->RegisterUMAObserver(init_observer); | |
| 2053 LocalP2PTest(); | |
| 2054 | |
| 2055 EXPECT_TRUE_WAIT( | |
| 2056 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
| 2057 initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT), | |
| 2058 kMaxWaitForStatsMs); | |
| 2059 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
| 2060 initializing_client()->GetSrtpCipherStats(), | |
| 2061 kMaxWaitForStatsMs); | |
| 2062 EXPECT_EQ(1, | |
| 2063 init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
| 2064 kDefaultSrtpCryptoSuite)); | |
| 2065 } | |
| 2066 | |
| 2067 // Test that a non-GCM cipher is used if both sides only support non-GCM. | |
| 2068 TEST_F(P2PTestConductor, GetGcmNone) { | |
| 2069 TestGcmNegotiation(false, false, kDefaultSrtpCryptoSuite); | |
| 2070 } | |
| 2071 | |
| 2072 // Test that a GCM cipher is used if both ends support it. | |
| 2073 TEST_F(P2PTestConductor, GetGcmBoth) { | |
| 2074 TestGcmNegotiation(true, true, kDefaultSrtpCryptoSuiteGcm); | |
| 2075 } | |
| 2076 | |
| 2077 // Test that GCM isn't used if only the initiator supports it. | |
| 2078 TEST_F(P2PTestConductor, GetGcmInit) { | |
| 2079 TestGcmNegotiation(true, false, kDefaultSrtpCryptoSuite); | |
| 2080 } | |
| 2081 | |
| 2082 // Test that GCM isn't used if only the receiver supports it. | |
| 2083 TEST_F(P2PTestConductor, GetGcmRecv) { | |
| 2084 TestGcmNegotiation(false, true, kDefaultSrtpCryptoSuite); | |
| 2085 } | |
| 2086 | |
| 2087 // This test sets up a call between two parties with audio, video and an RTP | |
| 2088 // data channel. | |
| 2089 TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) { | |
| 2090 FakeConstraints setup_constraints; | |
| 2091 setup_constraints.SetAllowRtpDataChannels(); | |
| 2092 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 2093 initializing_client()->CreateDataChannel(); | |
| 2094 LocalP2PTest(); | |
| 2095 ASSERT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2096 ASSERT_TRUE(receiving_client()->data_channel() != nullptr); | |
| 2097 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2098 kMaxWaitMs); | |
| 2099 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), | |
| 2100 kMaxWaitMs); | |
| 2101 | |
| 2102 std::string data = "hello world"; | |
| 2103 | |
| 2104 SendRtpData(initializing_client()->data_channel(), data); | |
| 2105 EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(), | |
| 2106 kMaxWaitMs); | |
| 2107 | |
| 2108 SendRtpData(receiving_client()->data_channel(), data); | |
| 2109 EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(), | |
| 2110 kMaxWaitMs); | |
| 2111 | |
| 2112 receiving_client()->data_channel()->Close(); | |
| 2113 // Send new offer and answer. | |
| 2114 receiving_client()->Negotiate(); | |
| 2115 EXPECT_FALSE(initializing_client()->data_observer()->IsOpen()); | |
| 2116 EXPECT_FALSE(receiving_client()->data_observer()->IsOpen()); | |
| 2117 } | |
| 2118 | |
| 2119 #ifdef HAVE_SCTP | |
| 2120 // This test sets up a call between two parties with audio, video and an SCTP | |
| 2121 // data channel. | |
| 2122 TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) { | |
| 2123 ASSERT_TRUE(CreateTestClients()); | |
| 2124 initializing_client()->CreateDataChannel(); | |
| 2125 LocalP2PTest(); | |
| 2126 ASSERT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2127 EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs); | |
| 2128 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2129 kMaxWaitMs); | |
| 2130 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs); | |
| 2131 | |
| 2132 std::string data = "hello world"; | |
| 2133 | |
| 2134 initializing_client()->data_channel()->Send(DataBuffer(data)); | |
| 2135 EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(), | |
| 2136 kMaxWaitMs); | |
| 2137 | |
| 2138 receiving_client()->data_channel()->Send(DataBuffer(data)); | |
| 2139 EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(), | |
| 2140 kMaxWaitMs); | |
| 2141 | |
| 2142 receiving_client()->data_channel()->Close(); | |
| 2143 EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(), | |
| 2144 kMaxWaitMs); | |
| 2145 EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs); | |
| 2146 } | |
| 2147 | |
| 2148 TEST_F(P2PTestConductor, UnorderedSctpDataChannel) { | |
| 2149 ASSERT_TRUE(CreateTestClients()); | |
| 2150 webrtc::DataChannelInit init; | |
| 2151 init.ordered = false; | |
| 2152 initializing_client()->CreateDataChannel(&init); | |
| 2153 | |
| 2154 // Introduce random network delays. | |
| 2155 // Otherwise it's not a true "unordered" test. | |
| 2156 virtual_socket_server()->set_delay_mean(20); | |
| 2157 virtual_socket_server()->set_delay_stddev(5); | |
| 2158 virtual_socket_server()->UpdateDelayDistribution(); | |
| 2159 | |
| 2160 initializing_client()->Negotiate(); | |
| 2161 ASSERT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2162 EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs); | |
| 2163 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2164 kMaxWaitMs); | |
| 2165 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs); | |
| 2166 | |
| 2167 static constexpr int kNumMessages = 100; | |
| 2168 // Deliberately chosen to be larger than the MTU so messages get fragmented. | |
| 2169 static constexpr size_t kMaxMessageSize = 4096; | |
| 2170 // Create and send random messages. | |
| 2171 std::vector<std::string> sent_messages; | |
| 2172 for (int i = 0; i < kNumMessages; ++i) { | |
| 2173 size_t length = (rand() % kMaxMessageSize) + 1; | |
| 2174 std::string message; | |
| 2175 ASSERT_TRUE(rtc::CreateRandomString(length, &message)); | |
| 2176 initializing_client()->data_channel()->Send(DataBuffer(message)); | |
| 2177 receiving_client()->data_channel()->Send(DataBuffer(message)); | |
| 2178 sent_messages.push_back(message); | |
| 2179 } | |
| 2180 | |
| 2181 EXPECT_EQ_WAIT( | |
| 2182 kNumMessages, | |
| 2183 initializing_client()->data_observer()->received_message_count(), | |
| 2184 kMaxWaitMs); | |
| 2185 EXPECT_EQ_WAIT(kNumMessages, | |
| 2186 receiving_client()->data_observer()->received_message_count(), | |
| 2187 kMaxWaitMs); | |
| 2188 | |
| 2189 // Sort and compare to make sure none of the messages were corrupted. | |
| 2190 std::vector<std::string> initializing_client_received_messages = | |
| 2191 initializing_client()->data_observer()->messages(); | |
| 2192 std::vector<std::string> receiving_client_received_messages = | |
| 2193 receiving_client()->data_observer()->messages(); | |
| 2194 std::sort(sent_messages.begin(), sent_messages.end()); | |
| 2195 std::sort(initializing_client_received_messages.begin(), | |
| 2196 initializing_client_received_messages.end()); | |
| 2197 std::sort(receiving_client_received_messages.begin(), | |
| 2198 receiving_client_received_messages.end()); | |
| 2199 EXPECT_EQ(sent_messages, initializing_client_received_messages); | |
| 2200 EXPECT_EQ(sent_messages, receiving_client_received_messages); | |
| 2201 | |
| 2202 receiving_client()->data_channel()->Close(); | |
| 2203 EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(), | |
| 2204 kMaxWaitMs); | |
| 2205 EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs); | |
| 2206 } | |
| 2207 #endif // HAVE_SCTP | |
| 2208 | |
| 2209 // This test sets up a call between two parties and creates a data channel. | |
| 2210 // The test tests that received data is buffered unless an observer has been | |
| 2211 // registered. | |
| 2212 // Rtp data channels can receive data before the underlying | |
| 2213 // transport has detected that a channel is writable and thus data can be | |
| 2214 // received before the data channel state changes to open. That is hard to test | |
| 2215 // but the same buffering is used in that case. | |
| 2216 TEST_F(P2PTestConductor, RegisterDataChannelObserver) { | |
| 2217 FakeConstraints setup_constraints; | |
| 2218 setup_constraints.SetAllowRtpDataChannels(); | |
| 2219 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 2220 initializing_client()->CreateDataChannel(); | |
| 2221 initializing_client()->Negotiate(); | |
| 2222 | |
| 2223 ASSERT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2224 ASSERT_TRUE(receiving_client()->data_channel() != nullptr); | |
| 2225 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2226 kMaxWaitMs); | |
| 2227 EXPECT_EQ_WAIT(DataChannelInterface::kOpen, | |
| 2228 receiving_client()->data_channel()->state(), kMaxWaitMs); | |
| 2229 | |
| 2230 // Unregister the existing observer. | |
| 2231 receiving_client()->data_channel()->UnregisterObserver(); | |
| 2232 | |
| 2233 std::string data = "hello world"; | |
| 2234 SendRtpData(initializing_client()->data_channel(), data); | |
| 2235 | |
| 2236 // Wait a while to allow the sent data to arrive before an observer is | |
| 2237 // registered.. | |
| 2238 rtc::Thread::Current()->ProcessMessages(100); | |
| 2239 | |
| 2240 MockDataChannelObserver new_observer(receiving_client()->data_channel()); | |
| 2241 EXPECT_EQ_WAIT(data, new_observer.last_message(), kMaxWaitMs); | |
| 2242 } | |
| 2243 | |
| 2244 // This test sets up a call between two parties with audio, video and but only | |
| 2245 // the initiating client support data. | |
| 2246 TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) { | |
| 2247 FakeConstraints setup_constraints_1; | |
| 2248 setup_constraints_1.SetAllowRtpDataChannels(); | |
| 2249 // Must disable DTLS to make negotiation succeed. | |
| 2250 setup_constraints_1.SetMandatory( | |
| 2251 MediaConstraintsInterface::kEnableDtlsSrtp, false); | |
| 2252 FakeConstraints setup_constraints_2; | |
| 2253 setup_constraints_2.SetMandatory( | |
| 2254 MediaConstraintsInterface::kEnableDtlsSrtp, false); | |
| 2255 ASSERT_TRUE(CreateTestClients(&setup_constraints_1, &setup_constraints_2)); | |
| 2256 initializing_client()->CreateDataChannel(); | |
| 2257 LocalP2PTest(); | |
| 2258 EXPECT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2259 EXPECT_FALSE(receiving_client()->data_channel()); | |
| 2260 EXPECT_FALSE(initializing_client()->data_observer()->IsOpen()); | |
| 2261 } | |
| 2262 | |
| 2263 // This test sets up a call between two parties with audio, video. When audio | |
| 2264 // and video is setup and flowing and data channel is negotiated. | |
| 2265 TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) { | |
| 2266 FakeConstraints setup_constraints; | |
| 2267 setup_constraints.SetAllowRtpDataChannels(); | |
| 2268 ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); | |
| 2269 LocalP2PTest(); | |
| 2270 initializing_client()->CreateDataChannel(); | |
| 2271 // Send new offer and answer. | |
| 2272 initializing_client()->Negotiate(); | |
| 2273 ASSERT_TRUE(initializing_client()->data_channel() != nullptr); | |
| 2274 ASSERT_TRUE(receiving_client()->data_channel() != nullptr); | |
| 2275 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2276 kMaxWaitMs); | |
| 2277 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), | |
| 2278 kMaxWaitMs); | |
| 2279 } | |
| 2280 | |
| 2281 // This test sets up a Jsep call with SCTP DataChannel and verifies the | |
| 2282 // negotiation is completed without error. | |
| 2283 #ifdef HAVE_SCTP | |
| 2284 TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) { | |
| 2285 FakeConstraints constraints; | |
| 2286 constraints.SetMandatory( | |
| 2287 MediaConstraintsInterface::kEnableDtlsSrtp, true); | |
| 2288 ASSERT_TRUE(CreateTestClients(&constraints, &constraints)); | |
| 2289 initializing_client()->CreateDataChannel(); | |
| 2290 initializing_client()->Negotiate(false, false); | |
| 2291 } | |
| 2292 #endif | |
| 2293 | |
| 2294 // This test sets up a call between two parties with audio, and video. | |
| 2295 // During the call, the initializing side restart ice and the test verifies that | |
| 2296 // new ice candidates are generated and audio and video still can flow. | |
| 2297 TEST_F(P2PTestConductor, IceRestart) { | |
| 2298 ASSERT_TRUE(CreateTestClients()); | |
| 2299 | |
| 2300 // Negotiate and wait for ice completion and make sure audio and video plays. | |
| 2301 LocalP2PTest(); | |
| 2302 | |
| 2303 // Create a SDP string of the first audio candidate for both clients. | |
| 2304 const webrtc::IceCandidateCollection* audio_candidates_initiator = | |
| 2305 initializing_client()->pc()->local_description()->candidates(0); | |
| 2306 const webrtc::IceCandidateCollection* audio_candidates_receiver = | |
| 2307 receiving_client()->pc()->local_description()->candidates(0); | |
| 2308 ASSERT_GT(audio_candidates_initiator->count(), 0u); | |
| 2309 ASSERT_GT(audio_candidates_receiver->count(), 0u); | |
| 2310 std::string initiator_candidate; | |
| 2311 EXPECT_TRUE( | |
| 2312 audio_candidates_initiator->at(0)->ToString(&initiator_candidate)); | |
| 2313 std::string receiver_candidate; | |
| 2314 EXPECT_TRUE(audio_candidates_receiver->at(0)->ToString(&receiver_candidate)); | |
| 2315 | |
| 2316 // Restart ice on the initializing client. | |
| 2317 receiving_client()->SetExpectIceRestart(true); | |
| 2318 initializing_client()->IceRestart(); | |
| 2319 | |
| 2320 // Negotiate and wait for ice completion again and make sure audio and video | |
| 2321 // plays. | |
| 2322 LocalP2PTest(); | |
| 2323 | |
| 2324 // Create a SDP string of the first audio candidate for both clients again. | |
| 2325 const webrtc::IceCandidateCollection* audio_candidates_initiator_restart = | |
| 2326 initializing_client()->pc()->local_description()->candidates(0); | |
| 2327 const webrtc::IceCandidateCollection* audio_candidates_reciever_restart = | |
| 2328 receiving_client()->pc()->local_description()->candidates(0); | |
| 2329 ASSERT_GT(audio_candidates_initiator_restart->count(), 0u); | |
| 2330 ASSERT_GT(audio_candidates_reciever_restart->count(), 0u); | |
| 2331 std::string initiator_candidate_restart; | |
| 2332 EXPECT_TRUE(audio_candidates_initiator_restart->at(0)->ToString( | |
| 2333 &initiator_candidate_restart)); | |
| 2334 std::string receiver_candidate_restart; | |
| 2335 EXPECT_TRUE(audio_candidates_reciever_restart->at(0)->ToString( | |
| 2336 &receiver_candidate_restart)); | |
| 2337 | |
| 2338 // Verify that the first candidates in the local session descriptions has | |
| 2339 // changed. | |
| 2340 EXPECT_NE(initiator_candidate, initiator_candidate_restart); | |
| 2341 EXPECT_NE(receiver_candidate, receiver_candidate_restart); | |
| 2342 } | |
| 2343 | |
| 2344 TEST_F(P2PTestConductor, IceRenominationDisabled) { | |
| 2345 PeerConnectionInterface::RTCConfiguration config; | |
| 2346 config.enable_ice_renomination = false; | |
| 2347 ASSERT_TRUE(CreateTestClients(config, config)); | |
| 2348 LocalP2PTest(); | |
| 2349 | |
| 2350 initializing_client()->VerifyLocalIceRenomination(); | |
| 2351 receiving_client()->VerifyLocalIceRenomination(); | |
| 2352 initializing_client()->VerifyRemoteIceRenomination(); | |
| 2353 receiving_client()->VerifyRemoteIceRenomination(); | |
| 2354 } | |
| 2355 | |
| 2356 TEST_F(P2PTestConductor, IceRenominationEnabled) { | |
| 2357 PeerConnectionInterface::RTCConfiguration config; | |
| 2358 config.enable_ice_renomination = true; | |
| 2359 ASSERT_TRUE(CreateTestClients(config, config)); | |
| 2360 initializing_client()->SetExpectIceRenomination(true); | |
| 2361 initializing_client()->SetExpectRemoteIceRenomination(true); | |
| 2362 receiving_client()->SetExpectIceRenomination(true); | |
| 2363 receiving_client()->SetExpectRemoteIceRenomination(true); | |
| 2364 LocalP2PTest(); | |
| 2365 | |
| 2366 initializing_client()->VerifyLocalIceRenomination(); | |
| 2367 receiving_client()->VerifyLocalIceRenomination(); | |
| 2368 initializing_client()->VerifyRemoteIceRenomination(); | |
| 2369 receiving_client()->VerifyRemoteIceRenomination(); | |
| 2370 } | |
| 2371 | |
| 2372 // This test sets up a call between two parties with audio, and video. | |
| 2373 // It then renegotiates setting the video m-line to "port 0", then later | |
| 2374 // renegotiates again, enabling video. | |
| 2375 TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) { | |
| 2376 ASSERT_TRUE(CreateTestClients()); | |
| 2377 | |
| 2378 // Do initial negotiation. Will result in video and audio sendonly m-lines. | |
| 2379 receiving_client()->set_auto_add_stream(false); | |
| 2380 initializing_client()->AddMediaStream(true, true); | |
| 2381 initializing_client()->Negotiate(); | |
| 2382 | |
| 2383 // Negotiate again, disabling the video m-line (receiving client will | |
| 2384 // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint). | |
| 2385 receiving_client()->SetReceiveVideo(false); | |
| 2386 initializing_client()->Negotiate(); | |
| 2387 | |
| 2388 // Enable video and do negotiation again, making sure video is received | |
| 2389 // end-to-end. | |
| 2390 receiving_client()->SetReceiveVideo(true); | |
| 2391 receiving_client()->AddMediaStream(true, true); | |
| 2392 LocalP2PTest(); | |
| 2393 } | |
| 2394 | |
| 2395 // This test sets up a Jsep call between two parties with external | |
| 2396 // VideoDecoderFactory. | |
| 2397 // TODO(holmer): Disabled due to sometimes crashing on buildbots. | |
| 2398 // See issue webrtc/2378. | |
| 2399 TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) { | |
| 2400 ASSERT_TRUE(CreateTestClients()); | |
| 2401 EnableVideoDecoderFactory(); | |
| 2402 LocalP2PTest(); | |
| 2403 } | |
| 2404 | |
| 2405 // This tests that if we negotiate after calling CreateSender but before we | |
| 2406 // have a track, then set a track later, frames from the newly-set track are | |
| 2407 // received end-to-end. | |
| 2408 TEST_F(P2PTestConductor, EarlyWarmupTest) { | |
| 2409 ASSERT_TRUE(CreateTestClients()); | |
| 2410 auto audio_sender = | |
| 2411 initializing_client()->pc()->CreateSender("audio", "stream_id"); | |
| 2412 auto video_sender = | |
| 2413 initializing_client()->pc()->CreateSender("video", "stream_id"); | |
| 2414 initializing_client()->Negotiate(); | |
| 2415 // Wait for ICE connection to complete, without any tracks. | |
| 2416 // Note that the receiving client WILL (in HandleIncomingOffer) create | |
| 2417 // tracks, so it's only the initiator here that's doing early warmup. | |
| 2418 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | |
| 2419 VerifySessionDescriptions(); | |
| 2420 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
| 2421 initializing_client()->ice_connection_state(), | |
| 2422 kMaxWaitForFramesMs); | |
| 2423 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
| 2424 receiving_client()->ice_connection_state(), | |
| 2425 kMaxWaitForFramesMs); | |
| 2426 // Now set the tracks, and expect frames to immediately start flowing. | |
| 2427 EXPECT_TRUE( | |
| 2428 audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack(""))); | |
| 2429 EXPECT_TRUE( | |
| 2430 video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack(""))); | |
| 2431 EXPECT_TRUE_WAIT(FramesHaveArrived(kEndAudioFrameCount, kEndVideoFrameCount), | |
| 2432 kMaxWaitForFramesMs); | |
| 2433 } | |
| 2434 | |
| 2435 #ifdef HAVE_QUIC | |
| 2436 // This test sets up a call between two parties using QUIC instead of DTLS for | |
| 2437 // audio and video, and a QUIC data channel. | |
| 2438 TEST_F(P2PTestConductor, LocalP2PTestQuicDataChannel) { | |
| 2439 PeerConnectionInterface::RTCConfiguration quic_config; | |
| 2440 quic_config.enable_quic = true; | |
| 2441 ASSERT_TRUE(CreateTestClients(quic_config, quic_config)); | |
| 2442 webrtc::DataChannelInit init; | |
| 2443 init.ordered = false; | |
| 2444 init.reliable = true; | |
| 2445 init.id = 1; | |
| 2446 initializing_client()->CreateDataChannel(&init); | |
| 2447 receiving_client()->CreateDataChannel(&init); | |
| 2448 LocalP2PTest(); | |
| 2449 ASSERT_NE(nullptr, initializing_client()->data_channel()); | |
| 2450 ASSERT_NE(nullptr, receiving_client()->data_channel()); | |
| 2451 EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(), | |
| 2452 kMaxWaitMs); | |
| 2453 EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs); | |
| 2454 | |
| 2455 std::string data = "hello world"; | |
| 2456 | |
| 2457 initializing_client()->data_channel()->Send(DataBuffer(data)); | |
| 2458 EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(), | |
| 2459 kMaxWaitMs); | |
| 2460 | |
| 2461 receiving_client()->data_channel()->Send(DataBuffer(data)); | |
| 2462 EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(), | |
| 2463 kMaxWaitMs); | |
| 2464 } | |
| 2465 | |
| 2466 // Tests that negotiation of QUIC data channels is completed without error. | |
| 2467 TEST_F(P2PTestConductor, NegotiateQuicDataChannel) { | |
| 2468 PeerConnectionInterface::RTCConfiguration quic_config; | |
| 2469 quic_config.enable_quic = true; | |
| 2470 ASSERT_TRUE(CreateTestClients(quic_config, quic_config)); | |
| 2471 FakeConstraints constraints; | |
| 2472 constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); | |
| 2473 ASSERT_TRUE(CreateTestClients(&constraints, &constraints)); | |
| 2474 webrtc::DataChannelInit init; | |
| 2475 init.ordered = false; | |
| 2476 init.reliable = true; | |
| 2477 init.id = 1; | |
| 2478 initializing_client()->CreateDataChannel(&init); | |
| 2479 initializing_client()->Negotiate(false, false); | |
| 2480 } | |
| 2481 | |
| 2482 // This test sets up a JSEP call using QUIC. The callee only receives video. | |
| 2483 TEST_F(P2PTestConductor, LocalP2PTestVideoOnlyWithQuic) { | |
| 2484 PeerConnectionInterface::RTCConfiguration quic_config; | |
| 2485 quic_config.enable_quic = true; | |
| 2486 ASSERT_TRUE(CreateTestClients(quic_config, quic_config)); | |
| 2487 receiving_client()->SetReceiveAudioVideo(false, true); | |
| 2488 LocalP2PTest(); | |
| 2489 } | |
| 2490 | |
| 2491 // This test sets up a JSEP call using QUIC. The callee only receives audio. | |
| 2492 TEST_F(P2PTestConductor, LocalP2PTestAudioOnlyWithQuic) { | |
| 2493 PeerConnectionInterface::RTCConfiguration quic_config; | |
| 2494 quic_config.enable_quic = true; | |
| 2495 ASSERT_TRUE(CreateTestClients(quic_config, quic_config)); | |
| 2496 receiving_client()->SetReceiveAudioVideo(true, false); | |
| 2497 LocalP2PTest(); | |
| 2498 } | |
| 2499 | |
| 2500 // This test sets up a JSEP call using QUIC. The callee rejects both audio and | |
| 2501 // video. | |
| 2502 TEST_F(P2PTestConductor, LocalP2PTestNoVideoAudioWithQuic) { | |
| 2503 PeerConnectionInterface::RTCConfiguration quic_config; | |
| 2504 quic_config.enable_quic = true; | |
| 2505 ASSERT_TRUE(CreateTestClients(quic_config, quic_config)); | |
| 2506 receiving_client()->SetReceiveAudioVideo(false, false); | |
| 2507 LocalP2PTest(); | |
| 2508 } | |
| 2509 | |
| 2510 #endif // HAVE_QUIC | |
| 2511 | |
| 2512 TEST_F(P2PTestConductor, ForwardVideoOnlyStream) { | |
| 2513 ASSERT_TRUE(CreateTestClients()); | |
| 2514 // One-way stream | |
| 2515 receiving_client()->set_auto_add_stream(false); | |
| 2516 // Video only, audio forwarding not expected to work. | |
| 2517 initializing_client()->AddMediaStream(false, true); | |
| 2518 initializing_client()->Negotiate(); | |
| 2519 | |
| 2520 ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs); | |
| 2521 VerifySessionDescriptions(); | |
| 2522 | |
| 2523 ASSERT_TRUE(initializing_client()->can_receive_video()); | |
| 2524 ASSERT_TRUE(receiving_client()->can_receive_video()); | |
| 2525 | |
| 2526 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
| 2527 initializing_client()->ice_connection_state(), | |
| 2528 kMaxWaitForFramesMs); | |
| 2529 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
| 2530 receiving_client()->ice_connection_state(), | |
| 2531 kMaxWaitForFramesMs); | |
| 2532 | |
| 2533 ASSERT_TRUE(receiving_client()->remote_streams()->count() == 1); | |
| 2534 | |
| 2535 // Echo the stream back. | |
| 2536 receiving_client()->pc()->AddStream( | |
| 2537 receiving_client()->remote_streams()->at(0)); | |
| 2538 receiving_client()->Negotiate(); | |
| 2539 | |
| 2540 EXPECT_TRUE_WAIT( | |
| 2541 initializing_client()->VideoFramesReceivedCheck(kEndVideoFrameCount), | |
| 2542 kMaxWaitForFramesMs); | |
| 2543 } | |
| 2544 | |
| 2545 // Test that we achieve the expected end-to-end connection time, using a | |
| 2546 // fake clock and simulated latency on the media and signaling paths. | |
| 2547 // We use a TURN<->TURN connection because this is usually the quickest to | |
| 2548 // set up initially, especially when we're confident the connection will work | |
| 2549 // and can start sending media before we get a STUN response. | |
| 2550 // | |
| 2551 // With various optimizations enabled, here are the network delays we expect to | |
| 2552 // be on the critical path: | |
| 2553 // 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then | |
| 2554 // signaling answer (with DTLS fingerprint). | |
| 2555 // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when | |
| 2556 // using TURN<->TURN pair, and DTLS exchange is 4 packets, | |
| 2557 // the first of which should have arrived before the answer. | |
| 2558 TEST_F(P2PTestConductor, EndToEndConnectionTimeWithTurnTurnPair) { | |
| 2559 rtc::ScopedFakeClock fake_clock; | |
| 2560 // Some things use a time of "0" as a special value, so we need to start out | |
| 2561 // the fake clock at a nonzero time. | |
| 2562 // TODO(deadbeef): Fix this. | |
| 2563 fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1)); | |
| 2564 | |
| 2565 static constexpr int media_hop_delay_ms = 50; | |
| 2566 static constexpr int signaling_trip_delay_ms = 500; | |
| 2567 // For explanation of these values, see comment above. | |
| 2568 static constexpr int required_media_hops = 9; | |
| 2569 static constexpr int required_signaling_trips = 2; | |
| 2570 // For internal delays (such as posting an event asychronously). | |
| 2571 static constexpr int allowed_internal_delay_ms = 20; | |
| 2572 static constexpr int total_connection_time_ms = | |
| 2573 media_hop_delay_ms * required_media_hops + | |
| 2574 signaling_trip_delay_ms * required_signaling_trips + | |
| 2575 allowed_internal_delay_ms; | |
| 2576 | |
| 2577 static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0", | |
| 2578 3478}; | |
| 2579 static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1", | |
| 2580 0}; | |
| 2581 static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0", | |
| 2582 3478}; | |
| 2583 static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1", | |
| 2584 0}; | |
| 2585 cricket::TestTurnServer turn_server_1(network_thread(), | |
| 2586 turn_server_1_internal_address, | |
| 2587 turn_server_1_external_address); | |
| 2588 cricket::TestTurnServer turn_server_2(network_thread(), | |
| 2589 turn_server_2_internal_address, | |
| 2590 turn_server_2_external_address); | |
| 2591 // Bypass permission check on received packets so media can be sent before | |
| 2592 // the candidate is signaled. | |
| 2593 turn_server_1.set_enable_permission_checks(false); | |
| 2594 turn_server_2.set_enable_permission_checks(false); | |
| 2595 | |
| 2596 PeerConnectionInterface::RTCConfiguration client_1_config; | |
| 2597 webrtc::PeerConnectionInterface::IceServer ice_server_1; | |
| 2598 ice_server_1.urls.push_back("turn:88.88.88.0:3478"); | |
| 2599 ice_server_1.username = "test"; | |
| 2600 ice_server_1.password = "test"; | |
| 2601 client_1_config.servers.push_back(ice_server_1); | |
| 2602 client_1_config.type = webrtc::PeerConnectionInterface::kRelay; | |
| 2603 client_1_config.presume_writable_when_fully_relayed = true; | |
| 2604 | |
| 2605 PeerConnectionInterface::RTCConfiguration client_2_config; | |
| 2606 webrtc::PeerConnectionInterface::IceServer ice_server_2; | |
| 2607 ice_server_2.urls.push_back("turn:99.99.99.0:3478"); | |
| 2608 ice_server_2.username = "test"; | |
| 2609 ice_server_2.password = "test"; | |
| 2610 client_2_config.servers.push_back(ice_server_2); | |
| 2611 client_2_config.type = webrtc::PeerConnectionInterface::kRelay; | |
| 2612 client_2_config.presume_writable_when_fully_relayed = true; | |
| 2613 | |
| 2614 ASSERT_TRUE(CreateTestClients(client_1_config, client_2_config)); | |
| 2615 // Set up the simulated delays. | |
| 2616 SetSignalingDelayMs(signaling_trip_delay_ms); | |
| 2617 virtual_socket_server()->set_delay_mean(media_hop_delay_ms); | |
| 2618 virtual_socket_server()->UpdateDelayDistribution(); | |
| 2619 | |
| 2620 initializing_client()->SetOfferToReceiveAudioVideo(true, true); | |
| 2621 initializing_client()->Negotiate(); | |
| 2622 // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS | |
| 2623 // are connected. This is an important distinction. Once we have separate ICE | |
| 2624 // and DTLS state, this check needs to use the DTLS state. | |
| 2625 EXPECT_TRUE_SIMULATED_WAIT( | |
| 2626 (receiving_client()->ice_connection_state() == | |
| 2627 webrtc::PeerConnectionInterface::kIceConnectionConnected || | |
| 2628 receiving_client()->ice_connection_state() == | |
| 2629 webrtc::PeerConnectionInterface::kIceConnectionCompleted) && | |
| 2630 (initializing_client()->ice_connection_state() == | |
| 2631 webrtc::PeerConnectionInterface::kIceConnectionConnected || | |
| 2632 initializing_client()->ice_connection_state() == | |
| 2633 webrtc::PeerConnectionInterface::kIceConnectionCompleted), | |
| 2634 total_connection_time_ms, fake_clock); | |
| 2635 // Need to free the clients here since they're using things we created on | |
| 2636 // the stack. | |
| 2637 delete set_initializing_client(nullptr); | |
| 2638 delete set_receiving_client(nullptr); | |
| 2639 } | |
| 2640 | |
| 2641 class IceServerParsingTest : public testing::Test { | |
| 2642 public: | |
| 2643 // Convenience for parsing a single URL. | |
| 2644 bool ParseUrl(const std::string& url) { | |
| 2645 return ParseUrl(url, std::string(), std::string()); | |
| 2646 } | |
| 2647 | |
| 2648 bool ParseTurnUrl(const std::string& url) { | |
| 2649 return ParseUrl(url, "username", "password"); | |
| 2650 } | |
| 2651 | |
| 2652 bool ParseUrl(const std::string& url, | |
| 2653 const std::string& username, | |
| 2654 const std::string& password) { | |
| 2655 return ParseUrl( | |
| 2656 url, username, password, | |
| 2657 PeerConnectionInterface::TlsCertPolicy::kTlsCertPolicySecure); | |
| 2658 } | |
| 2659 | |
| 2660 bool ParseUrl(const std::string& url, | |
| 2661 const std::string& username, | |
| 2662 const std::string& password, | |
| 2663 PeerConnectionInterface::TlsCertPolicy tls_certificate_policy) { | |
| 2664 PeerConnectionInterface::IceServers servers; | |
| 2665 PeerConnectionInterface::IceServer server; | |
| 2666 server.urls.push_back(url); | |
| 2667 server.username = username; | |
| 2668 server.password = password; | |
| 2669 server.tls_cert_policy = tls_certificate_policy; | |
| 2670 servers.push_back(server); | |
| 2671 return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_) == | |
| 2672 webrtc::RTCErrorType::NONE; | |
| 2673 } | |
| 2674 | |
| 2675 protected: | |
| 2676 cricket::ServerAddresses stun_servers_; | |
| 2677 std::vector<cricket::RelayServerConfig> turn_servers_; | |
| 2678 }; | |
| 2679 | |
| 2680 // Make sure all STUN/TURN prefixes are parsed correctly. | |
| 2681 TEST_F(IceServerParsingTest, ParseStunPrefixes) { | |
| 2682 EXPECT_TRUE(ParseUrl("stun:hostname")); | |
| 2683 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2684 EXPECT_EQ(0U, turn_servers_.size()); | |
| 2685 stun_servers_.clear(); | |
| 2686 | |
| 2687 EXPECT_TRUE(ParseUrl("stuns:hostname")); | |
| 2688 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2689 EXPECT_EQ(0U, turn_servers_.size()); | |
| 2690 stun_servers_.clear(); | |
| 2691 | |
| 2692 EXPECT_TRUE(ParseTurnUrl("turn:hostname")); | |
| 2693 EXPECT_EQ(0U, stun_servers_.size()); | |
| 2694 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2695 EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto); | |
| 2696 turn_servers_.clear(); | |
| 2697 | |
| 2698 EXPECT_TRUE(ParseTurnUrl("turns:hostname")); | |
| 2699 EXPECT_EQ(0U, stun_servers_.size()); | |
| 2700 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2701 EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto); | |
| 2702 EXPECT_TRUE(turn_servers_[0].tls_cert_policy == | |
| 2703 cricket::TlsCertPolicy::TLS_CERT_POLICY_SECURE); | |
| 2704 turn_servers_.clear(); | |
| 2705 | |
| 2706 EXPECT_TRUE(ParseUrl( | |
| 2707 "turns:hostname", "username", "password", | |
| 2708 PeerConnectionInterface::TlsCertPolicy::kTlsCertPolicyInsecureNoCheck)); | |
| 2709 EXPECT_EQ(0U, stun_servers_.size()); | |
| 2710 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2711 EXPECT_TRUE(turn_servers_[0].tls_cert_policy == | |
| 2712 cricket::TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK); | |
| 2713 EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto); | |
| 2714 turn_servers_.clear(); | |
| 2715 | |
| 2716 // invalid prefixes | |
| 2717 EXPECT_FALSE(ParseUrl("stunn:hostname")); | |
| 2718 EXPECT_FALSE(ParseUrl(":hostname")); | |
| 2719 EXPECT_FALSE(ParseUrl(":")); | |
| 2720 EXPECT_FALSE(ParseUrl("")); | |
| 2721 } | |
| 2722 | |
| 2723 TEST_F(IceServerParsingTest, VerifyDefaults) { | |
| 2724 // TURNS defaults | |
| 2725 EXPECT_TRUE(ParseTurnUrl("turns:hostname")); | |
| 2726 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2727 EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port()); | |
| 2728 EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto); | |
| 2729 turn_servers_.clear(); | |
| 2730 | |
| 2731 // TURN defaults | |
| 2732 EXPECT_TRUE(ParseTurnUrl("turn:hostname")); | |
| 2733 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2734 EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port()); | |
| 2735 EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto); | |
| 2736 turn_servers_.clear(); | |
| 2737 | |
| 2738 // STUN defaults | |
| 2739 EXPECT_TRUE(ParseUrl("stun:hostname")); | |
| 2740 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2741 EXPECT_EQ(3478, stun_servers_.begin()->port()); | |
| 2742 stun_servers_.clear(); | |
| 2743 } | |
| 2744 | |
| 2745 // Check that the 6 combinations of IPv4/IPv6/hostname and with/without port | |
| 2746 // can be parsed correctly. | |
| 2747 TEST_F(IceServerParsingTest, ParseHostnameAndPort) { | |
| 2748 EXPECT_TRUE(ParseUrl("stun:1.2.3.4:1234")); | |
| 2749 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2750 EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname()); | |
| 2751 EXPECT_EQ(1234, stun_servers_.begin()->port()); | |
| 2752 stun_servers_.clear(); | |
| 2753 | |
| 2754 EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]:4321")); | |
| 2755 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2756 EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname()); | |
| 2757 EXPECT_EQ(4321, stun_servers_.begin()->port()); | |
| 2758 stun_servers_.clear(); | |
| 2759 | |
| 2760 EXPECT_TRUE(ParseUrl("stun:hostname:9999")); | |
| 2761 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2762 EXPECT_EQ("hostname", stun_servers_.begin()->hostname()); | |
| 2763 EXPECT_EQ(9999, stun_servers_.begin()->port()); | |
| 2764 stun_servers_.clear(); | |
| 2765 | |
| 2766 EXPECT_TRUE(ParseUrl("stun:1.2.3.4")); | |
| 2767 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2768 EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname()); | |
| 2769 EXPECT_EQ(3478, stun_servers_.begin()->port()); | |
| 2770 stun_servers_.clear(); | |
| 2771 | |
| 2772 EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]")); | |
| 2773 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2774 EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname()); | |
| 2775 EXPECT_EQ(3478, stun_servers_.begin()->port()); | |
| 2776 stun_servers_.clear(); | |
| 2777 | |
| 2778 EXPECT_TRUE(ParseUrl("stun:hostname")); | |
| 2779 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2780 EXPECT_EQ("hostname", stun_servers_.begin()->hostname()); | |
| 2781 EXPECT_EQ(3478, stun_servers_.begin()->port()); | |
| 2782 stun_servers_.clear(); | |
| 2783 | |
| 2784 // Try some invalid hostname:port strings. | |
| 2785 EXPECT_FALSE(ParseUrl("stun:hostname:99a99")); | |
| 2786 EXPECT_FALSE(ParseUrl("stun:hostname:-1")); | |
| 2787 EXPECT_FALSE(ParseUrl("stun:hostname:port:more")); | |
| 2788 EXPECT_FALSE(ParseUrl("stun:hostname:port more")); | |
| 2789 EXPECT_FALSE(ParseUrl("stun:hostname:")); | |
| 2790 EXPECT_FALSE(ParseUrl("stun:[1:2:3:4:5:6:7:8]junk:1000")); | |
| 2791 EXPECT_FALSE(ParseUrl("stun::5555")); | |
| 2792 EXPECT_FALSE(ParseUrl("stun:")); | |
| 2793 } | |
| 2794 | |
| 2795 // Test parsing the "?transport=xxx" part of the URL. | |
| 2796 TEST_F(IceServerParsingTest, ParseTransport) { | |
| 2797 EXPECT_TRUE(ParseTurnUrl("turn:hostname:1234?transport=tcp")); | |
| 2798 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2799 EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto); | |
| 2800 turn_servers_.clear(); | |
| 2801 | |
| 2802 EXPECT_TRUE(ParseTurnUrl("turn:hostname?transport=udp")); | |
| 2803 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2804 EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto); | |
| 2805 turn_servers_.clear(); | |
| 2806 | |
| 2807 EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport=invalid")); | |
| 2808 EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport=")); | |
| 2809 EXPECT_FALSE(ParseTurnUrl("turn:hostname?=")); | |
| 2810 EXPECT_FALSE(ParseTurnUrl("turn:hostname?")); | |
| 2811 EXPECT_FALSE(ParseTurnUrl("?")); | |
| 2812 } | |
| 2813 | |
| 2814 // Test parsing ICE username contained in URL. | |
| 2815 TEST_F(IceServerParsingTest, ParseUsername) { | |
| 2816 EXPECT_TRUE(ParseTurnUrl("turn:user@hostname")); | |
| 2817 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2818 EXPECT_EQ("user", turn_servers_[0].credentials.username); | |
| 2819 turn_servers_.clear(); | |
| 2820 | |
| 2821 EXPECT_FALSE(ParseTurnUrl("turn:@hostname")); | |
| 2822 EXPECT_FALSE(ParseTurnUrl("turn:username@")); | |
| 2823 EXPECT_FALSE(ParseTurnUrl("turn:@")); | |
| 2824 EXPECT_FALSE(ParseTurnUrl("turn:user@name@hostname")); | |
| 2825 } | |
| 2826 | |
| 2827 // Test that username and password from IceServer is copied into the resulting | |
| 2828 // RelayServerConfig. | |
| 2829 TEST_F(IceServerParsingTest, CopyUsernameAndPasswordFromIceServer) { | |
| 2830 EXPECT_TRUE(ParseUrl("turn:hostname", "username", "password")); | |
| 2831 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2832 EXPECT_EQ("username", turn_servers_[0].credentials.username); | |
| 2833 EXPECT_EQ("password", turn_servers_[0].credentials.password); | |
| 2834 } | |
| 2835 | |
| 2836 // Ensure that if a server has multiple URLs, each one is parsed. | |
| 2837 TEST_F(IceServerParsingTest, ParseMultipleUrls) { | |
| 2838 PeerConnectionInterface::IceServers servers; | |
| 2839 PeerConnectionInterface::IceServer server; | |
| 2840 server.urls.push_back("stun:hostname"); | |
| 2841 server.urls.push_back("turn:hostname"); | |
| 2842 server.username = "foo"; | |
| 2843 server.password = "bar"; | |
| 2844 servers.push_back(server); | |
| 2845 EXPECT_EQ(webrtc::RTCErrorType::NONE, | |
| 2846 webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); | |
| 2847 EXPECT_EQ(1U, stun_servers_.size()); | |
| 2848 EXPECT_EQ(1U, turn_servers_.size()); | |
| 2849 } | |
| 2850 | |
| 2851 // Ensure that TURN servers are given unique priorities, | |
| 2852 // so that their resulting candidates have unique priorities. | |
| 2853 TEST_F(IceServerParsingTest, TurnServerPrioritiesUnique) { | |
| 2854 PeerConnectionInterface::IceServers servers; | |
| 2855 PeerConnectionInterface::IceServer server; | |
| 2856 server.urls.push_back("turn:hostname"); | |
| 2857 server.urls.push_back("turn:hostname2"); | |
| 2858 server.username = "foo"; | |
| 2859 server.password = "bar"; | |
| 2860 servers.push_back(server); | |
| 2861 EXPECT_EQ(webrtc::RTCErrorType::NONE, | |
| 2862 webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_)); | |
| 2863 EXPECT_EQ(2U, turn_servers_.size()); | |
| 2864 EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority); | |
| 2865 } | |
| 2866 | |
| 2867 #endif // if !defined(THREAD_SANITIZER) | |
| 2868 | |
| 2869 } // namespace | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2738353003:
Rewrite PeerConnection integration tests using better testing practices. (Closed)
