OLD | NEW |
---|---|
(Empty) | |
1 /* | |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 // Disable for TSan v2, see | |
12 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. | |
13 #if !defined(THREAD_SANITIZER) | |
14 | |
15 #include <stdio.h> | |
16 | |
17 #include <algorithm> | |
18 #include <functional> | |
19 #include <list> | |
20 #include <map> | |
21 #include <memory> | |
22 #include <utility> | |
23 #include <vector> | |
24 | |
25 #include "webrtc/api/fakemetricsobserver.h" | |
26 #include "webrtc/api/mediastreaminterface.h" | |
27 #include "webrtc/api/peerconnectioninterface.h" | |
28 #include "webrtc/api/test/fakeconstraints.h" | |
29 #include "webrtc/base/asyncinvoker.h" | |
30 #include "webrtc/base/fakenetwork.h" | |
31 #include "webrtc/base/gunit.h" | |
32 #include "webrtc/base/helpers.h" | |
33 #include "webrtc/base/physicalsocketserver.h" | |
34 #include "webrtc/base/ssladapter.h" | |
35 #include "webrtc/base/sslstreamadapter.h" | |
36 #include "webrtc/base/thread.h" | |
37 #include "webrtc/base/virtualsocketserver.h" | |
38 #include "webrtc/media/engine/fakewebrtcvideoengine.h" | |
39 #include "webrtc/p2p/base/p2pconstants.h" | |
40 #include "webrtc/p2p/base/portinterface.h" | |
41 #include "webrtc/p2p/base/sessiondescription.h" | |
42 #include "webrtc/p2p/base/testturnserver.h" | |
43 #include "webrtc/p2p/client/basicportallocator.h" | |
44 #include "webrtc/pc/dtmfsender.h" | |
45 #include "webrtc/pc/localaudiosource.h" | |
46 #include "webrtc/pc/mediasession.h" | |
47 #include "webrtc/pc/peerconnection.h" | |
48 #include "webrtc/pc/peerconnectionfactory.h" | |
49 #include "webrtc/pc/test/fakeaudiocapturemodule.h" | |
50 #include "webrtc/pc/test/fakeperiodicvideocapturer.h" | |
51 #include "webrtc/pc/test/fakertccertificategenerator.h" | |
52 #include "webrtc/pc/test/fakevideotrackrenderer.h" | |
53 #include "webrtc/pc/test/mockpeerconnectionobservers.h" | |
54 | |
55 using cricket::ContentInfo; | |
56 using cricket::FakeWebRtcVideoDecoder; | |
57 using cricket::FakeWebRtcVideoDecoderFactory; | |
58 using cricket::FakeWebRtcVideoEncoder; | |
59 using cricket::FakeWebRtcVideoEncoderFactory; | |
60 using cricket::MediaContentDescription; | |
61 using webrtc::DataBuffer; | |
62 using webrtc::DataChannelInterface; | |
63 using webrtc::DtmfSender; | |
64 using webrtc::DtmfSenderInterface; | |
65 using webrtc::DtmfSenderObserverInterface; | |
66 using webrtc::FakeConstraints; | |
67 using webrtc::MediaConstraintsInterface; | |
68 using webrtc::MediaStreamInterface; | |
69 using webrtc::MediaStreamTrackInterface; | |
70 using webrtc::MockCreateSessionDescriptionObserver; | |
71 using webrtc::MockDataChannelObserver; | |
72 using webrtc::MockSetSessionDescriptionObserver; | |
73 using webrtc::MockStatsObserver; | |
74 using webrtc::ObserverInterface; | |
75 using webrtc::PeerConnectionInterface; | |
76 using webrtc::PeerConnectionFactory; | |
77 using webrtc::SessionDescriptionInterface; | |
78 using webrtc::StreamCollectionInterface; | |
79 | |
80 namespace { | |
81 | |
82 static const int kDefaultTimeout = 10000; | |
83 static const int kMaxWaitForStatsMs = 3000; | |
84 static const int kMaxWaitForActivationMs = 5000; | |
85 static const int kMaxWaitForFramesMs = 10000; | |
86 static const int kEndAudioFrameCount = 3; | |
pthatcher1
2017/03/20 18:23:17
Can you leave a comment explaining what these are?
Taylor Brandstetter
2017/03/23 04:46:25
Done.
| |
87 static const int kEndVideoFrameCount = 3; | |
88 | |
89 static const char kDefaultStreamLabel[] = "stream_label"; | |
90 static const char kDefaultVideoTrackId[] = "video_track"; | |
91 static const char kDefaultAudioTrackId[] = "audio_track"; | |
92 static const char kDataChannelLabel[] = "data_channel"; | |
93 | |
94 // SRTP cipher name negotiated by the tests. This must be updated if the | |
95 // default changes. | |
96 static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32; | |
97 static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM; | |
98 | |
99 // Helper function for constructing offer/answer options to initiate an ICE | |
100 // restart. | |
101 PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() { | |
102 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
103 options.ice_restart = true; | |
104 return options; | |
105 } | |
106 | |
107 class SignalingMessageReceiver { | |
pthatcher1
2017/03/20 18:23:17
SignalingReceiverInterface might be a better name.
Taylor Brandstetter
2017/03/23 04:46:25
I prefer the current name personally.
| |
108 public: | |
109 virtual void ReceiveSdpMessage(const std::string& type, | |
110 const std::string& msg) = 0; | |
111 virtual void ReceiveIceMessage(const std::string& sdp_mid, | |
112 int sdp_mline_index, | |
113 const std::string& msg) = 0; | |
114 | |
115 protected: | |
116 SignalingMessageReceiver() {} | |
117 virtual ~SignalingMessageReceiver() {} | |
118 }; | |
119 | |
120 class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { | |
121 public: | |
122 explicit MockRtpReceiverObserver(cricket::MediaType media_type) | |
123 : expected_media_type_(media_type) {} | |
124 | |
125 void OnFirstPacketReceived(cricket::MediaType media_type) override { | |
126 ASSERT_EQ(expected_media_type_, media_type); | |
127 first_packet_received_ = true; | |
128 } | |
129 | |
130 bool first_packet_received() const { return first_packet_received_; } | |
131 | |
132 virtual ~MockRtpReceiverObserver() {} | |
133 | |
134 private: | |
135 bool first_packet_received_ = false; | |
136 cricket::MediaType expected_media_type_; | |
137 }; | |
138 | |
139 // Helper class that wraps a peer connection, observes it, and can accept | |
140 // signaling messages from another wrapper. | |
141 // | |
142 // Uses a fake network, fake A/V capture, and optionally fake | |
143 // encoders/decoders, though they aren't used by default since they don't | |
144 // advertise support of any codecs. | |
145 class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, | |
146 public SignalingMessageReceiver, | |
147 public ObserverInterface { | |
148 public: | |
149 // Different factory methods for convenience. | |
150 static PeerConnectionWrapper* CreateWithDtlsIdentityStore( | |
151 const std::string& debug_name, | |
152 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | |
153 rtc::Thread* network_thread, | |
154 rtc::Thread* worker_thread) { | |
155 PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); | |
156 if (!client->Init(nullptr, nullptr, nullptr, std::move(cert_generator), | |
157 network_thread, worker_thread)) { | |
158 delete client; | |
159 return nullptr; | |
160 } | |
161 return client; | |
162 } | |
163 | |
164 static PeerConnectionWrapper* CreateWithConfig( | |
165 const std::string& debug_name, | |
166 const PeerConnectionInterface::RTCConfiguration& config, | |
167 rtc::Thread* network_thread, | |
168 rtc::Thread* worker_thread) { | |
169 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
170 new FakeRTCCertificateGenerator()); | |
171 PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); | |
172 if (!client->Init(nullptr, nullptr, &config, std::move(cert_generator), | |
173 network_thread, worker_thread)) { | |
174 delete client; | |
175 return nullptr; | |
176 } | |
177 return client; | |
178 } | |
179 | |
180 static PeerConnectionWrapper* CreateWithOptions( | |
181 const std::string& debug_name, | |
182 const PeerConnectionFactory::Options& options, | |
183 rtc::Thread* network_thread, | |
184 rtc::Thread* worker_thread) { | |
185 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
186 new FakeRTCCertificateGenerator()); | |
187 PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); | |
188 if (!client->Init(nullptr, &options, nullptr, std::move(cert_generator), | |
189 network_thread, worker_thread)) { | |
190 delete client; | |
191 return nullptr; | |
192 } | |
193 return client; | |
194 } | |
195 | |
196 static PeerConnectionWrapper* CreateWithConstraints( | |
197 const std::string& debug_name, | |
198 const MediaConstraintsInterface* constraints, | |
199 rtc::Thread* network_thread, | |
200 rtc::Thread* worker_thread) { | |
201 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
202 new FakeRTCCertificateGenerator()); | |
203 PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); | |
204 if (!client->Init(constraints, nullptr, nullptr, std::move(cert_generator), | |
205 network_thread, worker_thread)) { | |
206 delete client; | |
207 return nullptr; | |
208 } | |
209 return client; | |
pthatcher1
2017/03/20 18:23:17
This is repeated 4 times. A simple Create method
Taylor Brandstetter
2017/03/23 04:46:25
That's what we had before which I intentionally ch
pthatcher1
2017/03/25 03:38:34
What I mean was to have the 7-parameter version an
Taylor Brandstetter
2017/03/27 17:12:16
Oh; that makes more sense. Sorry I misunderstood.
| |
210 } | |
211 | |
212 webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); } | |
213 | |
214 // If a signaling message receiver is set (via ConnectFakeSignaling), this | |
215 // will set the whole offer/answer exchange in motion. Just need to wait for | |
216 // the signaling state to reach "stable". | |
217 void CreateSetAndSignalOffer() { | |
pthatcher1
2017/03/20 18:23:17
I read that as "(Create set) and (Signal offer)".
Taylor Brandstetter
2017/03/23 04:46:25
Went with first suggestion.
| |
218 auto offer = CreateOffer(); | |
219 ASSERT_NE(nullptr, offer); | |
220 EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer))); | |
221 } | |
222 | |
223 // Sets the options to be used when CreateSetAndSignalOffer is called, or | |
224 // when a remote offer is received (via fake signaling) and an answer is | |
225 // generated. By default, uses default options. | |
226 void SetOfferAnswerOptions( | |
227 const PeerConnectionInterface::RTCOfferAnswerOptions& options) { | |
228 offer_answer_options_ = options; | |
229 } | |
230 | |
231 // Set a callback to be invoked when SDP is received via the fake signaling | |
232 // channel, which provides an opportunity to munge (modify) the SDP. This is | |
233 // used to test SDP being applied that a PeerConnection would normally not | |
234 // generate, but a non-JSEP endpoint might. | |
235 void SetReceivedSdpMunger( | |
236 std::function<void(cricket::SessionDescription*)> munger) { | |
237 received_sdp_munger_ = munger; | |
238 } | |
239 | |
240 // Siimlar to the above, but this is run on SDP immediately after it's | |
241 // generated. | |
242 void SetGeneratedSdpMunger( | |
243 std::function<void(cricket::SessionDescription*)> munger) { | |
244 generated_sdp_munger_ = munger; | |
245 } | |
246 | |
247 // Number of times the gathering state has transitioned to "gathering". | |
248 // Useful for telling if an ICE restart occurred as expected. | |
249 int transitions_to_gathering_state() const { | |
250 return transitions_to_gathering_state_; | |
251 } | |
252 | |
253 // TODO(deadbeef): Switch the majority of these tests to use AddTrack instead | |
254 // of AddStream since AddStream is deprecated. | |
255 void AddAudioVideoMediaStream() { | |
256 AddMediaStreamFromTracks(CreateLocalAudioTrack(), CreateLocalVideoTrack()); | |
257 } | |
258 | |
259 void AddAudioOnlyMediaStream() { | |
260 AddMediaStreamFromTracks(CreateLocalAudioTrack(), nullptr); | |
261 } | |
262 | |
263 void AddVideoOnlyMediaStream() { | |
264 AddMediaStreamFromTracks(nullptr, CreateLocalVideoTrack()); | |
265 } | |
266 | |
267 rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() { | |
268 FakeConstraints constraints; | |
269 // Disable highpass filter so that we can get all the test audio frames. | |
270 constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false); | |
271 rtc::scoped_refptr<webrtc::AudioSourceInterface> source = | |
272 peer_connection_factory_->CreateAudioSource(&constraints); | |
273 // TODO(perkj): Test audio source when it is implemented. Currently audio | |
274 // always use the default input. | |
275 return peer_connection_factory_->CreateAudioTrack(kDefaultAudioTrackId, | |
276 source); | |
277 } | |
278 | |
279 rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() { | |
280 return CreateLocalVideoTrackInternal( | |
281 kDefaultVideoTrackId, FakeConstraints(), webrtc::kVideoRotation_0); | |
282 } | |
283 | |
284 rtc::scoped_refptr<webrtc::VideoTrackInterface> | |
285 CreateLocalVideoTrackWithConstraints(const FakeConstraints& constraints) { | |
286 return CreateLocalVideoTrackInternal(kDefaultVideoTrackId, constraints, | |
287 webrtc::kVideoRotation_0); | |
288 } | |
289 | |
290 rtc::scoped_refptr<webrtc::VideoTrackInterface> | |
291 CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) { | |
292 return CreateLocalVideoTrackInternal(kDefaultVideoTrackId, | |
293 FakeConstraints(), rotation); | |
294 } | |
295 | |
296 rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackWithId( | |
297 const std::string& id) { | |
298 return CreateLocalVideoTrackInternal(id, FakeConstraints(), | |
299 webrtc::kVideoRotation_0); | |
300 } | |
301 | |
302 void AddMediaStreamFromTracks( | |
303 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio, | |
304 rtc::scoped_refptr<webrtc::VideoTrackInterface> video) { | |
305 AddMediaStreamFromTracksWithLabel(audio, video, kDefaultStreamLabel); | |
306 } | |
307 | |
308 void AddMediaStreamFromTracksWithLabel( | |
309 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio, | |
310 rtc::scoped_refptr<webrtc::VideoTrackInterface> video, | |
311 const std::string& stream_label) { | |
312 rtc::scoped_refptr<MediaStreamInterface> stream = | |
313 peer_connection_factory_->CreateLocalMediaStream(stream_label); | |
314 if (audio) { | |
315 stream->AddTrack(audio); | |
316 } | |
317 if (video) { | |
318 stream->AddTrack(video); | |
319 } | |
320 EXPECT_TRUE(pc()->AddStream(stream)); | |
321 } | |
322 | |
323 bool SignalingStateStable() { | |
324 return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable; | |
325 } | |
326 | |
327 void CreateDataChannel() { CreateDataChannel(nullptr); } | |
328 | |
329 void CreateDataChannel(const webrtc::DataChannelInit* init) { | |
330 data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init); | |
331 ASSERT_TRUE(data_channel_.get() != nullptr); | |
332 data_observer_.reset(new MockDataChannelObserver(data_channel_)); | |
333 } | |
334 | |
335 DataChannelInterface* data_channel() { return data_channel_; } | |
336 const MockDataChannelObserver* data_observer() const { | |
337 return data_observer_.get(); | |
338 } | |
339 | |
340 bool ReceivedAudioFrames(int number_of_frames) const { | |
341 return number_of_frames <= fake_audio_capture_module_->frames_received(); | |
342 } | |
343 | |
344 int audio_frames_received() const { | |
345 return fake_audio_capture_module_->frames_received(); | |
346 } | |
347 | |
348 bool ReceivedVideoFramesForEachTrack(int number_of_frames) { | |
pthatcher1
2017/03/20 18:23:17
Would it give more information to failing tests to
Taylor Brandstetter
2017/03/23 04:46:25
I'll make a helper function that handles this. It'
| |
349 if (video_decoder_factory_enabled_) { | |
350 const std::vector<FakeWebRtcVideoDecoder*>& decoders = | |
351 fake_video_decoder_factory_->decoders(); | |
352 if (decoders.empty()) { | |
353 return number_of_frames <= 0; | |
354 } | |
355 // Note - this checks that EACH decoder has the requisite number | |
356 // of frames. The video_frames_received() function sums them. | |
357 for (FakeWebRtcVideoDecoder* decoder : decoders) { | |
358 if (number_of_frames > decoder->GetNumFramesReceived()) { | |
359 return false; | |
360 } | |
361 } | |
362 return true; | |
363 } else { | |
364 if (fake_video_renderers_.empty()) { | |
365 return number_of_frames <= 0; | |
366 } | |
367 | |
368 for (const auto& pair : fake_video_renderers_) { | |
369 if (number_of_frames > pair.second->num_rendered_frames()) { | |
370 return false; | |
371 } | |
372 } | |
373 return true; | |
374 } | |
375 } | |
376 | |
377 int video_frames_received() const { | |
378 int total = 0; | |
379 if (video_decoder_factory_enabled_) { | |
380 const std::vector<FakeWebRtcVideoDecoder*>& decoders = | |
381 fake_video_decoder_factory_->decoders(); | |
382 for (const FakeWebRtcVideoDecoder* decoder : decoders) { | |
383 total += decoder->GetNumFramesReceived(); | |
384 } | |
385 } else { | |
386 for (const auto& pair : fake_video_renderers_) { | |
387 total += pair.second->num_rendered_frames(); | |
388 } | |
389 for (const auto& renderer : removed_fake_video_renderers_) { | |
390 total += renderer->num_rendered_frames(); | |
391 } | |
392 } | |
393 return total; | |
394 } | |
395 | |
396 // Returns a MockStatsObserver in a state after stats gathering finished, | |
397 // which can be used to access the gathered stats. | |
398 rtc::scoped_refptr<MockStatsObserver> GetStatsForTrack( | |
399 webrtc::MediaStreamTrackInterface* track) { | |
400 rtc::scoped_refptr<MockStatsObserver> observer( | |
401 new rtc::RefCountedObject<MockStatsObserver>()); | |
402 EXPECT_TRUE(peer_connection_->GetStats( | |
403 observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); | |
404 EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); | |
405 return observer; | |
406 } | |
407 | |
408 // Version that doesn't take a track "filter", and gathers all stats. | |
409 rtc::scoped_refptr<MockStatsObserver> GetStats() { | |
410 return GetStatsForTrack(nullptr); | |
411 } | |
412 | |
413 int rendered_width() { | |
414 EXPECT_FALSE(fake_video_renderers_.empty()); | |
415 return fake_video_renderers_.empty() | |
416 ? 0 | |
417 : fake_video_renderers_.begin()->second->width(); | |
418 } | |
419 | |
420 int rendered_height() { | |
421 EXPECT_FALSE(fake_video_renderers_.empty()); | |
422 return fake_video_renderers_.empty() | |
423 ? 0 | |
424 : fake_video_renderers_.begin()->second->height(); | |
425 } | |
426 | |
427 double rendered_aspect_ratio() { | |
428 if (rendered_height() == 0) { | |
429 return 0.0; | |
430 } | |
431 return static_cast<double>(rendered_width()) / rendered_height(); | |
432 } | |
433 | |
434 webrtc::VideoRotation rendered_rotation() { | |
435 EXPECT_FALSE(fake_video_renderers_.empty()); | |
436 return fake_video_renderers_.empty() | |
437 ? webrtc::kVideoRotation_0 | |
438 : fake_video_renderers_.begin()->second->rotation(); | |
439 } | |
440 | |
441 int local_rendered_width() { | |
442 return local_video_renderer_ ? local_video_renderer_->width() : 0; | |
443 } | |
444 | |
445 int local_rendered_height() { | |
446 return local_video_renderer_ ? local_video_renderer_->height() : 0; | |
447 } | |
448 | |
449 double local_rendered_aspect_ratio() { | |
450 if (local_rendered_height() == 0) { | |
451 return 0.0; | |
452 } | |
453 return static_cast<double>(local_rendered_width()) / | |
454 local_rendered_height(); | |
455 } | |
456 | |
457 size_t number_of_remote_streams() { | |
458 if (!pc()) { | |
459 return 0; | |
460 } | |
461 return pc()->remote_streams()->count(); | |
462 } | |
463 | |
464 StreamCollectionInterface* remote_streams() const { | |
465 if (!pc()) { | |
466 ADD_FAILURE(); | |
467 return nullptr; | |
468 } | |
469 return pc()->remote_streams(); | |
470 } | |
471 | |
472 StreamCollectionInterface* local_streams() { | |
473 if (!pc()) { | |
474 ADD_FAILURE(); | |
475 return nullptr; | |
476 } | |
477 return pc()->local_streams(); | |
478 } | |
479 | |
480 webrtc::PeerConnectionInterface::SignalingState signaling_state() { | |
481 return pc()->signaling_state(); | |
482 } | |
483 | |
484 webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() { | |
485 return pc()->ice_connection_state(); | |
486 } | |
487 | |
488 webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { | |
489 return pc()->ice_gathering_state(); | |
490 } | |
491 | |
492 // Returns a MockRtpReceiverObserver for each RtpReceiver returned by | |
493 // GetReceivers. They're updated automatically when a remote offer/answer | |
494 // from the fake signaling channel is applied, or when | |
495 // ResetRtpReceiverObservers below is called. | |
496 const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& | |
497 rtp_receiver_observers() { | |
498 return rtp_receiver_observers_; | |
499 } | |
500 | |
501 void ResetRtpReceiverObservers() { | |
502 rtp_receiver_observers_.clear(); | |
503 for (auto receiver : pc()->GetReceivers()) { | |
504 std::unique_ptr<MockRtpReceiverObserver> observer( | |
505 new MockRtpReceiverObserver(receiver->media_type())); | |
506 receiver->SetObserver(observer.get()); | |
507 rtp_receiver_observers_.push_back(std::move(observer)); | |
508 } | |
509 } | |
510 | |
511 private: | |
512 explicit PeerConnectionWrapper(const std::string& debug_name) | |
513 : debug_name_(debug_name) {} | |
514 | |
515 bool Init( | |
516 const MediaConstraintsInterface* constraints, | |
517 const PeerConnectionFactory::Options* options, | |
518 const PeerConnectionInterface::RTCConfiguration* config, | |
519 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | |
520 rtc::Thread* network_thread, | |
521 rtc::Thread* worker_thread) { | |
522 // There's an error in this test code if Init ends up being called twice. | |
523 RTC_DCHECK(!peer_connection_); | |
524 RTC_DCHECK(!peer_connection_factory_); | |
525 | |
526 fake_network_manager_.reset(new rtc::FakeNetworkManager()); | |
527 fake_network_manager_->AddInterface(rtc::SocketAddress("192.168.1.1", 0)); | |
528 | |
529 std::unique_ptr<cricket::PortAllocator> port_allocator( | |
530 new cricket::BasicPortAllocator(fake_network_manager_.get())); | |
531 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); | |
532 if (!fake_audio_capture_module_) { | |
533 return false; | |
534 } | |
535 // Note that these factories don't end up getting used unless supported | |
536 // codecs are added to them. | |
537 fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory(); | |
538 fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory(); | |
539 rtc::Thread* const signaling_thread = rtc::Thread::Current(); | |
540 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( | |
541 network_thread, worker_thread, signaling_thread, | |
542 fake_audio_capture_module_, fake_video_encoder_factory_, | |
543 fake_video_decoder_factory_); | |
544 if (!peer_connection_factory_) { | |
545 return false; | |
546 } | |
547 if (options) { | |
548 peer_connection_factory_->SetOptions(*options); | |
549 } | |
550 peer_connection_ = | |
551 CreatePeerConnection(std::move(port_allocator), constraints, config, | |
552 std::move(cert_generator)); | |
553 return peer_connection_.get() != nullptr; | |
554 } | |
555 | |
556 rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( | |
557 std::unique_ptr<cricket::PortAllocator> port_allocator, | |
558 const MediaConstraintsInterface* constraints, | |
559 const PeerConnectionInterface::RTCConfiguration* config, | |
560 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) { | |
561 PeerConnectionInterface::RTCConfiguration modified_config; | |
562 // If |config| is null, this will result in a default configuration being | |
563 // used. | |
564 if (config) { | |
565 modified_config = *config; | |
566 } | |
567 // Disable resolution adaptation; we don't want it interfering with the | |
568 // test results. | |
569 // TODO(deadbeef): Do something more robust. Since we're testing for aspect | |
570 // ratios and not specific resolutions, is this even necessary? | |
571 modified_config.set_cpu_adaptation(false); | |
572 | |
573 return peer_connection_factory_->CreatePeerConnection( | |
574 modified_config, constraints, std::move(port_allocator), | |
575 std::move(cert_generator), this); | |
576 } | |
577 | |
578 void set_signaling_message_receiver( | |
579 SignalingMessageReceiver* signaling_message_receiver) { | |
580 signaling_message_receiver_ = signaling_message_receiver; | |
581 } | |
582 | |
583 void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; } | |
584 | |
585 void EnableVideoDecoderFactory() { | |
586 video_decoder_factory_enabled_ = true; | |
587 fake_video_decoder_factory_->AddSupportedVideoCodecType( | |
588 webrtc::kVideoCodecVP8); | |
589 } | |
590 | |
591 rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal( | |
592 const std::string& track_id, | |
593 const FakeConstraints& constraints, | |
594 webrtc::VideoRotation rotation) { | |
595 // Set max frame rate to 10fps to reduce the risk of test flakiness. | |
596 // TODO(deadbeef): Do something more robust. | |
597 FakeConstraints source_constraints = constraints; | |
598 source_constraints.SetMandatoryMaxFrameRate(10); | |
599 | |
600 cricket::FakeVideoCapturer* fake_capturer = | |
601 new webrtc::FakePeriodicVideoCapturer(); | |
602 fake_capturer->SetRotation(rotation); | |
603 video_capturers_.push_back(fake_capturer); | |
604 rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source = | |
605 peer_connection_factory_->CreateVideoSource(fake_capturer, | |
606 &source_constraints); | |
607 rtc::scoped_refptr<webrtc::VideoTrackInterface> track( | |
608 peer_connection_factory_->CreateVideoTrack(track_id, source)); | |
609 if (!local_video_renderer_) { | |
610 local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track)); | |
611 } | |
612 return track; | |
613 } | |
614 | |
615 void HandleIncomingOffer(const std::string& msg) { | |
616 LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer"; | |
617 std::unique_ptr<SessionDescriptionInterface> desc( | |
618 webrtc::CreateSessionDescription("offer", msg, nullptr)); | |
619 if (received_sdp_munger_) { | |
620 received_sdp_munger_(desc->description()); | |
621 } | |
622 | |
623 EXPECT_TRUE(SetRemoteDescription(std::move(desc))); | |
624 // Setting a remote description may have changed the number of receivers, | |
625 // so reset the receiver observers. | |
626 ResetRtpReceiverObservers(); | |
627 auto answer = CreateAnswer(); | |
628 ASSERT_NE(nullptr, answer); | |
629 EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer))); | |
630 } | |
631 | |
632 void HandleIncomingAnswer(const std::string& msg) { | |
633 LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer"; | |
634 std::unique_ptr<SessionDescriptionInterface> desc( | |
635 webrtc::CreateSessionDescription("answer", msg, nullptr)); | |
636 if (received_sdp_munger_) { | |
637 received_sdp_munger_(desc->description()); | |
638 } | |
639 | |
640 EXPECT_TRUE(SetRemoteDescription(std::move(desc))); | |
641 // Set the RtpReceiverObserver after receivers are created. | |
642 ResetRtpReceiverObservers(); | |
643 } | |
644 | |
645 // Returns null on failure. | |
646 std::unique_ptr<SessionDescriptionInterface> CreateOfferOrAnswer(bool offer) { | |
647 rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( | |
648 new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); | |
649 if (offer) { | |
650 pc()->CreateOffer(observer, offer_answer_options_); | |
651 } else { | |
652 pc()->CreateAnswer(observer, offer_answer_options_); | |
653 } | |
654 | |
655 EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout); | |
656 if (!observer->result()) { | |
657 return nullptr; | |
658 } | |
659 auto description = observer->MoveDescription(); | |
660 if (generated_sdp_munger_) { | |
661 generated_sdp_munger_(description->description()); | |
662 } | |
663 return description; | |
664 } | |
665 | |
666 // Returns null on failure. | |
667 std::unique_ptr<SessionDescriptionInterface> CreateOffer() { | |
668 return CreateOfferOrAnswer(true); | |
669 } | |
670 | |
671 // Returns null on failure. | |
672 std::unique_ptr<SessionDescriptionInterface> CreateAnswer() { | |
673 return CreateOfferOrAnswer(false); | |
pthatcher1
2017/03/20 18:23:17
It seems like this would be a little more clear as
Taylor Brandstetter
2017/03/23 04:46:25
Done.
| |
674 } | |
675 | |
676 // Setting the local description and sending the SDP message over the fake | |
677 // signaling channel are combined into the same method because the SDP | |
678 // message needs to be sent as soon as SetLocalDescription finishes, without | |
679 // waiting for the observer to be called. This ensures that ICE candidates | |
680 // don't outrace the description. | |
681 bool SetLocalDescriptionAndSendSdpMessage( | |
682 std::unique_ptr<SessionDescriptionInterface> desc) { | |
683 rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( | |
684 new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); | |
685 LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage"; | |
686 std::string type = desc->type(); | |
687 std::string sdp; | |
688 EXPECT_TRUE(desc->ToString(&sdp)); | |
689 pc()->SetLocalDescription(observer, desc.release()); | |
690 // As mentioned above, we need to send the message immediately after | |
691 // SetLocalDescription. | |
692 SendSdpMessage(type, sdp); | |
693 EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); | |
694 return true; | |
695 } | |
696 | |
697 bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) { | |
698 rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( | |
699 new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); | |
700 LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription"; | |
701 pc()->SetRemoteDescription(observer, desc.release()); | |
702 EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); | |
703 return observer->result(); | |
704 } | |
705 | |
706 // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by | |
707 // default). | |
708 void SendSdpMessage(const std::string& type, const std::string& msg) { | |
709 invoker_.AsyncInvokeDelayed<void>( | |
710 RTC_FROM_HERE, rtc::Thread::Current(), | |
711 rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists, this, | |
712 type, msg), | |
713 signaling_delay_ms_); | |
714 } | |
715 | |
716 void RelaySdpMessageIfReceiverExists(const std::string& type, | |
717 const std::string& msg) { | |
718 if (signaling_message_receiver_) { | |
719 signaling_message_receiver_->ReceiveSdpMessage(type, msg); | |
720 } | |
721 } | |
722 | |
723 // Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by | |
724 // default). | |
725 void SendIceMessage(const std::string& sdp_mid, | |
726 int sdp_mline_index, | |
727 const std::string& msg) { | |
728 invoker_.AsyncInvokeDelayed<void>( | |
729 RTC_FROM_HERE, rtc::Thread::Current(), | |
730 rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists, this, | |
731 sdp_mid, sdp_mline_index, msg), | |
732 signaling_delay_ms_); | |
733 } | |
734 | |
735 void RelayIceMessageIfReceiverExists(const std::string& sdp_mid, | |
736 int sdp_mline_index, | |
737 const std::string& msg) { | |
738 if (signaling_message_receiver_) { | |
739 signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index, | |
740 msg); | |
741 } | |
742 } | |
743 | |
744 // SignalingMessageReceiver callbacks. | |
745 void ReceiveSdpMessage(const std::string& type, | |
746 const std::string& msg) override { | |
747 if (type == webrtc::SessionDescriptionInterface::kOffer) { | |
748 HandleIncomingOffer(msg); | |
749 } else { | |
750 HandleIncomingAnswer(msg); | |
751 } | |
752 } | |
753 | |
754 void ReceiveIceMessage(const std::string& sdp_mid, | |
755 int sdp_mline_index, | |
756 const std::string& msg) override { | |
757 LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage"; | |
758 std::unique_ptr<webrtc::IceCandidateInterface> candidate( | |
759 webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr)); | |
760 EXPECT_TRUE(pc()->AddIceCandidate(candidate.get())); | |
761 } | |
762 | |
763 // PeerConnectionObserver callbacks. | |
764 void OnSignalingChange( | |
765 webrtc::PeerConnectionInterface::SignalingState new_state) override { | |
766 EXPECT_EQ(pc()->signaling_state(), new_state); | |
767 } | |
768 void OnAddStream( | |
769 rtc::scoped_refptr<MediaStreamInterface> media_stream) override { | |
770 media_stream->RegisterObserver(this); | |
771 for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) { | |
772 const std::string id = media_stream->GetVideoTracks()[i]->id(); | |
773 ASSERT_TRUE(fake_video_renderers_.find(id) == | |
774 fake_video_renderers_.end()); | |
775 fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer( | |
776 media_stream->GetVideoTracks()[i])); | |
777 } | |
778 } | |
779 void OnRemoveStream( | |
780 rtc::scoped_refptr<MediaStreamInterface> media_stream) override {} | |
781 void OnRenegotiationNeeded() override {} | |
782 void OnIceConnectionChange( | |
783 webrtc::PeerConnectionInterface::IceConnectionState new_state) override { | |
784 EXPECT_EQ(pc()->ice_connection_state(), new_state); | |
785 } | |
786 void OnIceGatheringChange( | |
787 webrtc::PeerConnectionInterface::IceGatheringState new_state) override { | |
788 if (new_state == PeerConnectionInterface::kIceGatheringGathering) { | |
789 ++transitions_to_gathering_state_; | |
790 } | |
791 EXPECT_EQ(pc()->ice_gathering_state(), new_state); | |
792 } | |
793 void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { | |
794 LOG(LS_INFO) << debug_name_ << ": OnIceCandidate"; | |
795 | |
796 std::string ice_sdp; | |
797 EXPECT_TRUE(candidate->ToString(&ice_sdp)); | |
798 if (signaling_message_receiver_ == nullptr) { | |
799 // Remote party may be deleted. | |
800 return; | |
801 } | |
802 SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp); | |
803 } | |
804 void OnDataChannel( | |
805 rtc::scoped_refptr<DataChannelInterface> data_channel) override { | |
806 LOG(LS_INFO) << debug_name_ << ": OnDataChannel"; | |
807 data_channel_ = data_channel; | |
808 data_observer_.reset(new MockDataChannelObserver(data_channel)); | |
809 } | |
810 | |
811 // MediaStreamInterface callback | |
812 void OnChanged() override { | |
813 // Track added or removed from MediaStream, so update our renderers. | |
814 rtc::scoped_refptr<StreamCollectionInterface> remote_streams = | |
815 pc()->remote_streams(); | |
816 // Remove renderers for tracks that were removed. | |
817 for (auto it = fake_video_renderers_.begin(); | |
818 it != fake_video_renderers_.end();) { | |
819 if (remote_streams->FindVideoTrack(it->first) == nullptr) { | |
820 auto to_remove = it++; | |
821 removed_fake_video_renderers_.push_back(std::move(to_remove->second)); | |
822 fake_video_renderers_.erase(to_remove); | |
823 } else { | |
824 ++it; | |
825 } | |
826 } | |
827 // Create renderers for new video tracks. | |
828 for (size_t stream_index = 0; stream_index < remote_streams->count(); | |
829 ++stream_index) { | |
830 MediaStreamInterface* remote_stream = remote_streams->at(stream_index); | |
831 for (size_t track_index = 0; | |
832 track_index < remote_stream->GetVideoTracks().size(); | |
833 ++track_index) { | |
834 const std::string id = | |
835 remote_stream->GetVideoTracks()[track_index]->id(); | |
836 if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) { | |
837 continue; | |
838 } | |
839 fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer( | |
840 remote_stream->GetVideoTracks()[track_index])); | |
841 } | |
842 } | |
843 } | |
844 | |
845 std::string debug_name_; | |
846 | |
847 std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_; | |
848 | |
849 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | |
850 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | |
851 peer_connection_factory_; | |
852 | |
853 // Needed to keep track of number of frames sent. | |
854 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | |
855 // Needed to keep track of number of frames received. | |
856 std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>> | |
857 fake_video_renderers_; | |
858 // Needed to ensure frames aren't received for removed tracks. | |
859 std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>> | |
860 removed_fake_video_renderers_; | |
861 // Needed to keep track of number of frames received when external decoder | |
862 // used. | |
863 FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr; | |
864 FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr; | |
865 bool video_decoder_factory_enabled_ = false; | |
866 | |
867 // For remote peer communication. | |
868 SignalingMessageReceiver* signaling_message_receiver_ = nullptr; | |
869 int signaling_delay_ms_ = 0; | |
870 | |
871 // Store references to the video capturers we've created, so that we can stop | |
872 // them, if required. | |
873 std::vector<cricket::FakeVideoCapturer*> video_capturers_; | |
874 // |local_video_renderer_| attached to the first created local video track. | |
875 std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_; | |
876 | |
877 PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_; | |
878 std::function<void(cricket::SessionDescription*)> received_sdp_munger_; | |
879 std::function<void(cricket::SessionDescription*)> generated_sdp_munger_; | |
880 | |
881 rtc::scoped_refptr<DataChannelInterface> data_channel_; | |
882 std::unique_ptr<MockDataChannelObserver> data_observer_; | |
883 | |
884 std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_; | |
885 | |
886 int transitions_to_gathering_state_ = 0; | |
887 | |
888 rtc::AsyncInvoker invoker_; | |
889 | |
890 friend class PeerConnectionIntegrationTest; | |
891 }; | |
892 | |
893 // Tests two PeerConnections connecting to each other end-to-end, using a | |
894 // virtual network, fake A/V capture and fake encoder/decoders. The | |
895 // PeerConnections share the threads/socket servers, but use separate versions | |
896 // of everything else (including "PeerConnectionFactory"s). | |
897 class PeerConnectionIntegrationTest : public testing::Test { | |
898 public: | |
899 PeerConnectionIntegrationTest() | |
900 : pss_(new rtc::PhysicalSocketServer), | |
901 ss_(new rtc::VirtualSocketServer(pss_.get())), | |
902 network_thread_(new rtc::Thread(ss_.get())), | |
903 worker_thread_(rtc::Thread::Create()) { | |
904 RTC_CHECK(network_thread_->Start()); | |
905 RTC_CHECK(worker_thread_->Start()); | |
906 } | |
907 | |
908 ~PeerConnectionIntegrationTest() { | |
909 if (caller_pc_wrapper_) { | |
910 caller_pc_wrapper_->set_signaling_message_receiver(nullptr); | |
911 } | |
912 if (callee_pc_wrapper_) { | |
913 callee_pc_wrapper_->set_signaling_message_receiver(nullptr); | |
914 } | |
915 } | |
916 | |
917 bool SignalingStateStable() { | |
918 return caller_pc_wrapper_->SignalingStateStable() && | |
919 callee_pc_wrapper_->SignalingStateStable(); | |
920 } | |
921 | |
922 bool CreatePeerConnectionWrappers() { | |
923 return CreatePeerConnectionWrappersWithConfig( | |
924 PeerConnectionInterface::RTCConfiguration(), | |
925 PeerConnectionInterface::RTCConfiguration()); | |
926 } | |
927 | |
928 bool CreatePeerConnectionWrappersWithConstraints( | |
929 MediaConstraintsInterface* caller_constraints, | |
930 MediaConstraintsInterface* callee_constraints) { | |
931 caller_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithConstraints( | |
932 "Caller", caller_constraints, network_thread_.get(), | |
933 worker_thread_.get())); | |
934 callee_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithConstraints( | |
935 "Callee", callee_constraints, network_thread_.get(), | |
936 worker_thread_.get())); | |
937 return caller_pc_wrapper_ && callee_pc_wrapper_; | |
938 } | |
939 | |
940 bool CreatePeerConnectionWrappersWithConfig( | |
941 const PeerConnectionInterface::RTCConfiguration& caller_config, | |
942 const PeerConnectionInterface::RTCConfiguration& callee_config) { | |
943 caller_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithConfig( | |
944 "Caller", caller_config, network_thread_.get(), worker_thread_.get())); | |
945 callee_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithConfig( | |
946 "Callee", callee_config, network_thread_.get(), worker_thread_.get())); | |
947 return caller_pc_wrapper_ && callee_pc_wrapper_; | |
948 } | |
949 | |
950 bool CreatePeerConnectionWrappersWithOptions( | |
951 const PeerConnectionFactory::Options& caller_options, | |
952 const PeerConnectionFactory::Options& callee_options) { | |
953 caller_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithOptions( | |
954 "Caller", caller_options, network_thread_.get(), worker_thread_.get())); | |
955 callee_pc_wrapper_.reset(PeerConnectionWrapper::CreateWithOptions( | |
956 "Callee", callee_options, network_thread_.get(), worker_thread_.get())); | |
957 return caller_pc_wrapper_ && callee_pc_wrapper_; | |
958 } | |
959 | |
960 PeerConnectionWrapper* CreatePeerConnectionWrapperWithAlternateKey() { | |
961 std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( | |
962 new FakeRTCCertificateGenerator()); | |
963 cert_generator->use_alternate_key(); | |
964 | |
965 // Make sure the new client is using a different certificate. | |
966 return PeerConnectionWrapper::CreateWithDtlsIdentityStore( | |
967 "New Peer", std::move(cert_generator), network_thread_.get(), | |
968 worker_thread_.get()); | |
969 } | |
970 | |
971 // Once called, SDP blobs and ICE candidates will be automatically signaled | |
972 // between PeerConnections. | |
973 void ConnectFakeSignaling() { | |
974 caller_pc_wrapper_->set_signaling_message_receiver( | |
975 callee_pc_wrapper_.get()); | |
976 callee_pc_wrapper_->set_signaling_message_receiver( | |
977 caller_pc_wrapper_.get()); | |
978 } | |
979 | |
980 void SetSignalingDelayMs(int delay_ms) { | |
981 caller_pc_wrapper_->set_signaling_delay_ms(delay_ms); | |
982 callee_pc_wrapper_->set_signaling_delay_ms(delay_ms); | |
983 } | |
984 | |
985 void EnableVideoDecoderFactory() { | |
986 caller_pc_wrapper_->EnableVideoDecoderFactory(); | |
987 callee_pc_wrapper_->EnableVideoDecoderFactory(); | |
988 } | |
989 | |
990 // Messages may get lost on the unreliable DataChannel, so we send multiple | |
991 // times to avoid test flakiness. | |
992 void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc, | |
993 const std::string& data, | |
994 int retries) { | |
995 for (int i = 0; i < retries; ++i) { | |
996 dc->Send(DataBuffer(data)); | |
997 } | |
998 } | |
999 | |
1000 rtc::Thread* network_thread() { return network_thread_.get(); } | |
1001 | |
1002 rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); } | |
1003 | |
1004 PeerConnectionWrapper* caller_pc_wrapper() { | |
1005 return caller_pc_wrapper_.get(); | |
1006 } | |
1007 | |
1008 // Set the |caller_pc_wrapper_| to the |wrapper| passed in and return the | |
1009 // original |caller_pc_wrapper_|. | |
1010 PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent( | |
1011 PeerConnectionWrapper* wrapper) { | |
1012 PeerConnectionWrapper* old = caller_pc_wrapper_.release(); | |
1013 caller_pc_wrapper_.reset(wrapper); | |
1014 return old; | |
1015 } | |
1016 | |
1017 PeerConnectionWrapper* callee_pc_wrapper() { | |
1018 return callee_pc_wrapper_.get(); | |
1019 } | |
1020 | |
1021 // Set the |callee_pc_wrapper_| to the |wrapper| passed in and return the | |
1022 // original |callee_pc_wrapper_|. | |
1023 PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent( | |
1024 PeerConnectionWrapper* wrapper) { | |
1025 PeerConnectionWrapper* old = callee_pc_wrapper_.release(); | |
1026 callee_pc_wrapper_.reset(wrapper); | |
1027 return old; | |
1028 } | |
1029 | |
1030 void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled, | |
1031 bool remote_gcm_enabled, | |
1032 int expected_cipher_suite) { | |
1033 PeerConnectionFactory::Options caller_options; | |
1034 caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled; | |
1035 PeerConnectionFactory::Options callee_options; | |
1036 callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled; | |
1037 ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options, | |
1038 callee_options)); | |
1039 rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = | |
1040 new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
1041 caller_pc_wrapper()->pc()->RegisterUMAObserver(caller_observer); | |
1042 ConnectFakeSignaling(); | |
1043 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1044 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1045 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1046 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1047 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite), | |
1048 caller_pc_wrapper()->GetStats()->SrtpCipher(), | |
1049 kDefaultTimeout); | |
1050 EXPECT_EQ( | |
1051 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
1052 expected_cipher_suite)); | |
1053 caller_pc_wrapper()->pc()->RegisterUMAObserver(nullptr); | |
1054 } | |
1055 | |
1056 private: | |
1057 // |ss_| is used by |network_thread_| so it must be destroyed later. | |
1058 std::unique_ptr<rtc::PhysicalSocketServer> pss_; | |
1059 std::unique_ptr<rtc::VirtualSocketServer> ss_; | |
1060 // |network_thread_| and |worker_thread_| are used by both | |
1061 // |caller_pc_wrapper_| and |callee_pc_wrapper_| so they must be destroyed | |
1062 // later. | |
1063 std::unique_ptr<rtc::Thread> network_thread_; | |
1064 std::unique_ptr<rtc::Thread> worker_thread_; | |
1065 std::unique_ptr<PeerConnectionWrapper> caller_pc_wrapper_; | |
1066 std::unique_ptr<PeerConnectionWrapper> callee_pc_wrapper_; | |
pthatcher1
2017/03/20 18:23:17
I think you ought to just call this caller_ and ca
Taylor Brandstetter
2017/03/23 04:46:25
Done.
| |
1067 }; | |
1068 | |
1069 // Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This | |
1070 // includes testing that the callback is invoked if an observer is connected | |
1071 // after the first packet has already been received. | |
1072 TEST_F(PeerConnectionIntegrationTest, | |
1073 RtpReceiverObserverOnFirstPacketReceived) { | |
1074 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1075 ConnectFakeSignaling(); | |
1076 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1077 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1078 // Start offer/answer exchange and wait for it to complete. | |
1079 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1080 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1081 // Should be one receiver each for audio/video. | |
1082 EXPECT_EQ(2, caller_pc_wrapper()->rtp_receiver_observers().size()); | |
1083 EXPECT_EQ(2, callee_pc_wrapper()->rtp_receiver_observers().size()); | |
1084 // Wait for all "first packet received" callbacks to be fired. | |
1085 EXPECT_TRUE_WAIT( | |
1086 std::all_of(caller_pc_wrapper()->rtp_receiver_observers().begin(), | |
1087 caller_pc_wrapper()->rtp_receiver_observers().end(), | |
1088 [](const std::unique_ptr<MockRtpReceiverObserver>& o) { | |
1089 return o->first_packet_received(); | |
1090 }), | |
pthatcher1
2017/03/20 18:23:17
This is repeated many times. Does it make sense t
Taylor Brandstetter
2017/03/23 04:46:25
I did something similar, see new patchset.
| |
1091 kMaxWaitForFramesMs); | |
1092 EXPECT_TRUE_WAIT( | |
1093 std::all_of(callee_pc_wrapper()->rtp_receiver_observers().begin(), | |
1094 callee_pc_wrapper()->rtp_receiver_observers().end(), | |
1095 [](const std::unique_ptr<MockRtpReceiverObserver>& o) { | |
1096 return o->first_packet_received(); | |
1097 }), | |
1098 kMaxWaitForFramesMs); | |
1099 // If new observers are set after the first packet was already received, the | |
1100 // callback should still be invoked. | |
1101 caller_pc_wrapper()->ResetRtpReceiverObservers(); | |
1102 callee_pc_wrapper()->ResetRtpReceiverObservers(); | |
1103 EXPECT_EQ(2, caller_pc_wrapper()->rtp_receiver_observers().size()); | |
1104 EXPECT_EQ(2, callee_pc_wrapper()->rtp_receiver_observers().size()); | |
1105 EXPECT_TRUE( | |
1106 std::all_of(caller_pc_wrapper()->rtp_receiver_observers().begin(), | |
1107 caller_pc_wrapper()->rtp_receiver_observers().end(), | |
1108 [](const std::unique_ptr<MockRtpReceiverObserver>& o) { | |
1109 return o->first_packet_received(); | |
1110 })); | |
1111 EXPECT_TRUE( | |
1112 std::all_of(callee_pc_wrapper()->rtp_receiver_observers().begin(), | |
1113 callee_pc_wrapper()->rtp_receiver_observers().end(), | |
1114 [](const std::unique_ptr<MockRtpReceiverObserver>& o) { | |
1115 return o->first_packet_received(); | |
1116 })); | |
1117 } | |
1118 | |
1119 class DummyDtmfObserver : public DtmfSenderObserverInterface { | |
1120 public: | |
1121 DummyDtmfObserver() : completed_(false) {} | |
1122 | |
1123 // Implements DtmfSenderObserverInterface. | |
1124 void OnToneChange(const std::string& tone) override { | |
1125 tones_.push_back(tone); | |
1126 if (tone.empty()) { | |
1127 completed_ = true; | |
1128 } | |
1129 } | |
1130 | |
1131 const std::vector<std::string>& tones() const { return tones_; } | |
1132 bool completed() const { return completed_; } | |
1133 | |
1134 private: | |
1135 bool completed_; | |
1136 std::vector<std::string> tones_; | |
1137 }; | |
1138 | |
1139 // Assumes |sender| already has an audio track added and the offer/answer | |
1140 // exchange is done. | |
1141 void TestDtmfBetween(PeerConnectionWrapper* sender, | |
1142 PeerConnectionWrapper* receiver) { | |
pthatcher1
2017/03/20 18:23:17
Could probably just call it "TestDtmf". I'm not s
Taylor Brandstetter
2017/03/23 04:46:25
I'll call it TestDtmfFromSenderToReceiver to hint
| |
1143 DummyDtmfObserver observer; | |
1144 rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender; | |
1145 | |
1146 // We should be able to create a DTMF sender from a local track. | |
1147 webrtc::AudioTrackInterface* localtrack = | |
1148 sender->local_streams()->at(0)->GetAudioTracks()[0]; | |
1149 dtmf_sender = sender->pc()->CreateDtmfSender(localtrack); | |
1150 ASSERT_NE(nullptr, dtmf_sender.get()); | |
1151 dtmf_sender->RegisterObserver(&observer); | |
1152 | |
1153 // Test the DtmfSender object just created. | |
1154 EXPECT_TRUE(dtmf_sender->CanInsertDtmf()); | |
1155 EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50)); | |
1156 | |
1157 EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout); | |
1158 std::vector<std::string> tones = {"1", "a", ""}; | |
1159 EXPECT_EQ(tones, observer.tones()); | |
1160 dtmf_sender->UnregisterObserver(); | |
1161 // TODO(deadbeef): Verify the tones were actually received end-to-end. | |
1162 } | |
1163 | |
1164 // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each | |
1165 // direction). | |
1166 TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) { | |
1167 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1168 ConnectFakeSignaling(); | |
1169 // Only need audio for DTMF. | |
1170 caller_pc_wrapper()->AddAudioOnlyMediaStream(); | |
1171 callee_pc_wrapper()->AddAudioOnlyMediaStream(); | |
1172 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1173 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1174 TestDtmfBetween(caller_pc_wrapper(), callee_pc_wrapper()); | |
1175 TestDtmfBetween(callee_pc_wrapper(), caller_pc_wrapper()); | |
1176 } | |
1177 | |
1178 // Basic end-to-end test, verifying media can be encoded/transmitted/decoded | |
1179 // between two connections, using DTLS-SRTP. | |
1180 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls) { | |
1181 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1182 ConnectFakeSignaling(); | |
1183 // Do normal offer/answer and wait for some frames to be received in each | |
1184 // direction. | |
1185 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1186 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1187 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1188 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1189 EXPECT_TRUE_WAIT( | |
1190 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1191 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1192 kEndVideoFrameCount) && | |
1193 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1194 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1195 kEndVideoFrameCount), | |
1196 kMaxWaitForFramesMs); | |
1197 } | |
1198 | |
1199 // Uses SDES instead of DTLS for key agreement. | |
1200 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSdes) { | |
1201 PeerConnectionInterface::RTCConfiguration sdes_config; | |
1202 sdes_config.enable_dtls_srtp.emplace(false); | |
1203 ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config)); | |
1204 ConnectFakeSignaling(); | |
1205 | |
1206 // Do normal offer/answer and wait for some frames to be received in each | |
1207 // direction. | |
1208 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1209 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1210 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1211 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1212 EXPECT_TRUE_WAIT( | |
1213 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1214 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1215 kEndVideoFrameCount) && | |
1216 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1217 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1218 kEndVideoFrameCount), | |
1219 kMaxWaitForFramesMs); | |
1220 } | |
1221 | |
1222 // This test sets up a call between two parties (using DTLS) and tests that we | |
1223 // can get a video aspect ratio of 16:9. | |
1224 TEST_F(PeerConnectionIntegrationTest, SendAndReceive16To9AspectRatio) { | |
1225 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1226 ConnectFakeSignaling(); | |
1227 | |
1228 // Add video tracks with 16:9 constraint. | |
1229 FakeConstraints constraints; | |
1230 double requested_ratio = 16.0 / 9; | |
1231 constraints.SetMandatoryMinAspectRatio(requested_ratio); | |
1232 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1233 nullptr, | |
1234 caller_pc_wrapper()->CreateLocalVideoTrackWithConstraints(constraints)); | |
1235 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1236 nullptr, | |
1237 callee_pc_wrapper()->CreateLocalVideoTrackWithConstraints(constraints)); | |
1238 | |
1239 // Do normal offer/answer and wait for at least one frame to be received in | |
1240 // each direction. | |
1241 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1242 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1243 kMaxWaitForFramesMs); | |
1244 EXPECT_TRUE_WAIT(callee_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1245 kMaxWaitForFramesMs); | |
1246 | |
1247 // Check rendered aspect ratio. | |
1248 EXPECT_EQ(requested_ratio, | |
1249 caller_pc_wrapper()->local_rendered_aspect_ratio()); | |
1250 EXPECT_EQ(requested_ratio, caller_pc_wrapper()->rendered_aspect_ratio()); | |
1251 EXPECT_EQ(requested_ratio, | |
1252 callee_pc_wrapper()->local_rendered_aspect_ratio()); | |
1253 EXPECT_EQ(requested_ratio, callee_pc_wrapper()->rendered_aspect_ratio()); | |
1254 } | |
1255 | |
1256 // This test sets up a call between two parties with a source resolution of | |
1257 // 1280x720 and verifies that a 16:9 aspect ratio is received. | |
1258 TEST_F(PeerConnectionIntegrationTest, | |
1259 Send1280By720ResolutionAndReceive16To9AspectRatio) { | |
1260 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1261 ConnectFakeSignaling(); | |
1262 | |
1263 // Similar to above test, but uses MandatoryMin[Width/Height] constraint | |
1264 // instead of aspect ratio constraint. | |
1265 FakeConstraints constraints; | |
1266 constraints.SetMandatoryMinWidth(1280); | |
1267 constraints.SetMandatoryMinHeight(720); | |
1268 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1269 nullptr, | |
1270 caller_pc_wrapper()->CreateLocalVideoTrackWithConstraints(constraints)); | |
1271 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1272 nullptr, | |
1273 callee_pc_wrapper()->CreateLocalVideoTrackWithConstraints(constraints)); | |
1274 | |
1275 // Do normal offer/answer and wait for at least one frame to be received in | |
1276 // each direction. | |
1277 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1278 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedVideoFramesForEachTrack(1) && | |
1279 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1280 kMaxWaitForFramesMs); | |
1281 | |
1282 // Check rendered aspect ratio. | |
1283 EXPECT_EQ(16.0 / 9, caller_pc_wrapper()->local_rendered_aspect_ratio()); | |
1284 EXPECT_EQ(16.0 / 9, caller_pc_wrapper()->rendered_aspect_ratio()); | |
1285 EXPECT_EQ(16.0 / 9, callee_pc_wrapper()->local_rendered_aspect_ratio()); | |
1286 EXPECT_EQ(16.0 / 9, callee_pc_wrapper()->rendered_aspect_ratio()); | |
1287 } | |
1288 | |
1289 // This test sets up an one-way call, with media only from caller to | |
1290 // callee. | |
1291 TEST_F(PeerConnectionIntegrationTest, OneWayMediaCall) { | |
1292 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1293 ConnectFakeSignaling(); | |
1294 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1295 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1296 EXPECT_TRUE_WAIT( | |
1297 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1298 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1299 kEndVideoFrameCount), | |
1300 kMaxWaitForFramesMs); | |
1301 } | |
1302 | |
1303 // This test sets up a audio call initially, with the callee rejecting video | |
1304 // initially. Then later the callee decides to upgrade to audio/video, and | |
1305 // initiates a new offer/answer exchange. | |
1306 TEST_F(PeerConnectionIntegrationTest, AudioToVideoUpgrade) { | |
1307 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1308 ConnectFakeSignaling(); | |
1309 // Initially, offer an audio/video stream from the caller, but refuse to | |
1310 // send/receive video on the callee side. | |
1311 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1312 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1313 callee_pc_wrapper()->CreateLocalAudioTrack(), nullptr); | |
1314 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
1315 options.offer_to_receive_video = 0; | |
1316 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
1317 // Do offer/answer and make sure audio is still received end-to-end. | |
1318 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1319 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1320 EXPECT_TRUE_WAIT( | |
1321 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1322 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount), | |
1323 kMaxWaitForFramesMs); | |
1324 // Sanity check that the callee's description has a rejected video section. | |
1325 ASSERT_NE(nullptr, callee_pc_wrapper()->pc()->local_description()); | |
1326 const ContentInfo* callee_video_content = GetFirstVideoContent( | |
1327 callee_pc_wrapper()->pc()->local_description()->description()); | |
1328 ASSERT_NE(nullptr, callee_video_content); | |
1329 EXPECT_TRUE(callee_video_content->rejected); | |
1330 // Now negotiate with video and ensure negotiation succeeds, with video | |
1331 // frames and additional audio frames being received. | |
1332 callee_pc_wrapper()->AddMediaStreamFromTracksWithLabel( | |
1333 nullptr, callee_pc_wrapper()->CreateLocalVideoTrack(), | |
1334 "video_only_stream"); | |
1335 options.offer_to_receive_video = 1; | |
1336 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
1337 callee_pc_wrapper()->CreateSetAndSignalOffer(); | |
1338 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1339 int last_caller_audio_frames = caller_pc_wrapper()->audio_frames_received(); | |
1340 int last_callee_audio_frames = callee_pc_wrapper()->audio_frames_received(); | |
1341 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedAudioFrames( | |
1342 kEndAudioFrameCount + last_caller_audio_frames) && | |
1343 callee_pc_wrapper()->ReceivedAudioFrames( | |
1344 kEndAudioFrameCount + last_callee_audio_frames) && | |
1345 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1346 kEndVideoFrameCount) && | |
1347 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1348 kEndVideoFrameCount), | |
1349 kMaxWaitForFramesMs); | |
1350 } | |
1351 | |
1352 // This test sets up a call that's transferred to a new caller with a different | |
1353 // DTLS fingerprint. | |
1354 TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) { | |
1355 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1356 ConnectFakeSignaling(); | |
1357 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1358 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1359 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1360 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1361 | |
1362 // Keep the original peer around which will still send packets to the | |
1363 // receiving client. These SRTP packets will be dropped. | |
1364 std::unique_ptr<PeerConnectionWrapper> original_peer( | |
1365 SetCallerPcWrapperAndReturnCurrent( | |
1366 CreatePeerConnectionWrapperWithAlternateKey())); | |
1367 // TODO(deadbeef): Why do we call Close here? That goes against the comment | |
1368 // directly above. | |
1369 original_peer->pc()->Close(); | |
1370 | |
1371 // Store the last frame counts so we can ensure additional frames are | |
1372 // received from the new peer. | |
1373 int last_audio_frames = callee_pc_wrapper()->audio_frames_received(); | |
1374 int last_video_frames = callee_pc_wrapper()->video_frames_received(); | |
1375 | |
1376 ConnectFakeSignaling(); | |
1377 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1378 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1379 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1380 // Wait for some additional frames to be transmitted end-to-end. | |
1381 EXPECT_TRUE_WAIT( | |
1382 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1383 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1384 kEndVideoFrameCount) && | |
1385 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount + | |
1386 last_audio_frames) && | |
1387 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1388 kEndVideoFrameCount + last_video_frames), | |
1389 kMaxWaitForFramesMs); | |
1390 } | |
1391 | |
1392 // This test sets up a call that's transferred to a new callee with a different | |
1393 // DTLS fingerprint. | |
1394 TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) { | |
1395 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1396 ConnectFakeSignaling(); | |
1397 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1398 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1399 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1400 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1401 | |
1402 // Keep the original peer around which will still send packets to the | |
1403 // receiving client. These SRTP packets will be dropped. | |
1404 std::unique_ptr<PeerConnectionWrapper> original_peer( | |
1405 SetCalleePcWrapperAndReturnCurrent( | |
1406 CreatePeerConnectionWrapperWithAlternateKey())); | |
1407 // TODO(deadbeef): Why do we call Close here? That goes against the comment | |
1408 // directly above. | |
1409 original_peer->pc()->Close(); | |
1410 | |
1411 // Store the last frame counts so we can ensure additional frames are | |
1412 // received from the new peer. | |
1413 int last_audio_frames = caller_pc_wrapper()->audio_frames_received(); | |
1414 int last_video_frames = caller_pc_wrapper()->video_frames_received(); | |
1415 | |
1416 ConnectFakeSignaling(); | |
1417 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1418 caller_pc_wrapper()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); | |
1419 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1420 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1421 // Wait for some additional frames to be transmitted end-to-end. | |
1422 EXPECT_TRUE_WAIT( | |
1423 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount + | |
1424 last_audio_frames) && | |
1425 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1426 kEndVideoFrameCount + last_video_frames) && | |
1427 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1428 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1429 kEndVideoFrameCount), | |
1430 kMaxWaitForFramesMs); | |
1431 } | |
1432 | |
1433 // This test sets up a non-bundled call and negotiates bundling at the same | |
1434 // time as starting an ICE restart. When bundling is in effect in the restart, | |
1435 // the DTLS-SRTP context should be successfully reset. | |
1436 TEST_F(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) { | |
1437 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1438 ConnectFakeSignaling(); | |
1439 | |
1440 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1441 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1442 // Remove the bundle group from the SDP received by the callee. | |
1443 callee_pc_wrapper()->SetReceivedSdpMunger( | |
1444 [](cricket::SessionDescription* desc) { | |
1445 desc->RemoveGroupByName("BUNDLE"); | |
1446 }); | |
1447 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1448 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1449 EXPECT_TRUE_WAIT( | |
1450 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1451 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1452 kEndVideoFrameCount) && | |
1453 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1454 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1455 kEndVideoFrameCount), | |
pthatcher1
2017/03/20 18:23:17
This seems like another duplicate that could be mo
| |
1456 kMaxWaitForFramesMs); | |
1457 | |
1458 // Now stop removing the BUNDLE group, and trigger an ICE restart. | |
1459 callee_pc_wrapper()->SetReceivedSdpMunger(nullptr); | |
1460 caller_pc_wrapper()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); | |
1461 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1462 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1463 EXPECT_TRUE_WAIT( | |
1464 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1465 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1466 kEndVideoFrameCount) && | |
1467 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1468 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1469 kEndVideoFrameCount), | |
1470 kMaxWaitForFramesMs); | |
1471 } | |
1472 | |
1473 // Test CVO (Coordination of Video Orientation). If a video source is rotated | |
1474 // and both peers support the CVO RTP header extension, the actual video frames | |
1475 // don't need to be encoded in different resolutions, since the rotation is | |
1476 // communicated through the RTP header extension. | |
1477 TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) { | |
1478 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1479 ConnectFakeSignaling(); | |
1480 // Add rotated video tracks. | |
1481 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1482 nullptr, caller_pc_wrapper()->CreateLocalVideoTrackWithRotation( | |
1483 webrtc::kVideoRotation_90)); | |
1484 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1485 nullptr, callee_pc_wrapper()->CreateLocalVideoTrackWithRotation( | |
1486 webrtc::kVideoRotation_270)); | |
1487 | |
1488 // Wait for video frames to be received by both sides. | |
1489 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1490 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1491 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1492 kMaxWaitForFramesMs); | |
1493 EXPECT_TRUE_WAIT(callee_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1494 kMaxWaitForFramesMs); | |
1495 | |
1496 // Ensure that the aspect ratio is unmodified. | |
1497 // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, | |
1498 // not just assumed. | |
pthatcher1
2017/03/20 18:23:17
Perhaps make constants for kUnrotatedAspectRatio a
Taylor Brandstetter
2017/03/23 04:46:25
I don't think that helps, since ultimately the con
| |
1499 EXPECT_EQ(4.0 / 3, caller_pc_wrapper()->local_rendered_aspect_ratio()); | |
1500 EXPECT_EQ(4.0 / 3, caller_pc_wrapper()->rendered_aspect_ratio()); | |
1501 EXPECT_EQ(4.0 / 3, callee_pc_wrapper()->local_rendered_aspect_ratio()); | |
1502 EXPECT_EQ(4.0 / 3, callee_pc_wrapper()->rendered_aspect_ratio()); | |
1503 // Ensure that the CVO bits were surfaced to the renderer. | |
1504 EXPECT_EQ(webrtc::kVideoRotation_270, | |
1505 caller_pc_wrapper()->rendered_rotation()); | |
1506 EXPECT_EQ(webrtc::kVideoRotation_90, | |
1507 callee_pc_wrapper()->rendered_rotation()); | |
1508 } | |
1509 | |
1510 // Test that when the CVO extension isn't supported, video is rotated the | |
1511 // old-fashioned way, by encoding rotated frames. | |
1512 TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) { | |
1513 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1514 ConnectFakeSignaling(); | |
1515 // Add rotated video tracks. | |
1516 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1517 nullptr, caller_pc_wrapper()->CreateLocalVideoTrackWithRotation( | |
1518 webrtc::kVideoRotation_90)); | |
1519 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1520 nullptr, callee_pc_wrapper()->CreateLocalVideoTrackWithRotation( | |
1521 webrtc::kVideoRotation_270)); | |
1522 | |
1523 // Remove the CVO extension from the offered SDP. | |
1524 callee_pc_wrapper()->SetReceivedSdpMunger( | |
1525 [](cricket::SessionDescription* desc) { | |
1526 cricket::VideoContentDescription* video = | |
1527 GetFirstVideoContentDescription(desc); | |
1528 video->ClearRtpHeaderExtensions(); | |
1529 }); | |
1530 // Wait for video frames to be received by both sides. | |
1531 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1532 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1533 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1534 kMaxWaitForFramesMs); | |
1535 EXPECT_TRUE_WAIT(callee_pc_wrapper()->ReceivedVideoFramesForEachTrack(1), | |
1536 kMaxWaitForFramesMs); | |
1537 | |
1538 // Expect that the aspect ratio is inversed to account for the 90/270 degree | |
1539 // rotation. | |
1540 // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, | |
1541 // not just assumed. | |
1542 EXPECT_EQ(3.0 / 4, caller_pc_wrapper()->local_rendered_aspect_ratio()); | |
1543 EXPECT_EQ(3.0 / 4, caller_pc_wrapper()->rendered_aspect_ratio()); | |
1544 EXPECT_EQ(3.0 / 4, callee_pc_wrapper()->local_rendered_aspect_ratio()); | |
1545 EXPECT_EQ(3.0 / 4, callee_pc_wrapper()->rendered_aspect_ratio()); | |
1546 // Expect that each endpoint is unaware of the rotation of the other endpoint. | |
1547 EXPECT_EQ(webrtc::kVideoRotation_0, caller_pc_wrapper()->rendered_rotation()); | |
1548 EXPECT_EQ(webrtc::kVideoRotation_0, callee_pc_wrapper()->rendered_rotation()); | |
1549 } | |
1550 | |
1551 // TODO(deadbeef): The tests below rely on RTCOfferAnswerOptions to reject an | |
1552 // m= section. When we implement Unified Plan SDP, the right way to do this | |
1553 // would be by stopping an RtpTransceiver. | |
1554 | |
1555 // Test that if the answerer rejects the audio m= section, no audio is sent or | |
1556 // received, but video still can be. | |
1557 TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) { | |
1558 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1559 ConnectFakeSignaling(); | |
1560 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1561 // Only add video track for callee, and set offer_to_receive_audio to 0, so | |
1562 // it will reject the audio m= section completely. | |
1563 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
1564 options.offer_to_receive_audio = 0; | |
1565 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
1566 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1567 nullptr, callee_pc_wrapper()->CreateLocalVideoTrack()); | |
1568 // Do offer/answer and wait for successful end-to-end video frames. | |
1569 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1570 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1571 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1572 kEndVideoFrameCount) && | |
1573 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1574 kEndVideoFrameCount), | |
1575 kMaxWaitForFramesMs); | |
1576 // Shouldn't have received audio frames at any point. | |
1577 EXPECT_EQ(0, caller_pc_wrapper()->audio_frames_received()); | |
1578 EXPECT_EQ(0, callee_pc_wrapper()->audio_frames_received()); | |
1579 // Sanity check that the callee's description has a rejected audio section. | |
1580 ASSERT_NE(nullptr, callee_pc_wrapper()->pc()->local_description()); | |
1581 const ContentInfo* callee_audio_content = GetFirstAudioContent( | |
1582 callee_pc_wrapper()->pc()->local_description()->description()); | |
1583 ASSERT_NE(nullptr, callee_audio_content); | |
1584 EXPECT_TRUE(callee_audio_content->rejected); | |
1585 } | |
1586 | |
1587 // Test that if the answerer rejects the video m= section, no video is sent or | |
1588 // received, but audio still can be. | |
1589 TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) { | |
1590 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1591 ConnectFakeSignaling(); | |
1592 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1593 // Only add audio track for callee, and set offer_to_receive_video to 0, so | |
1594 // it will reject the video m= section completely. | |
1595 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
1596 options.offer_to_receive_video = 0; | |
1597 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
1598 callee_pc_wrapper()->AddMediaStreamFromTracks( | |
1599 callee_pc_wrapper()->CreateLocalAudioTrack(), nullptr); | |
1600 // Do offer/answer and wait for successful end-to-end audio frames. | |
1601 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1602 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1603 EXPECT_TRUE_WAIT( | |
1604 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1605 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount), | |
1606 kMaxWaitForFramesMs); | |
1607 // Shouldn't have received video frames at any point. | |
1608 EXPECT_EQ(0, caller_pc_wrapper()->video_frames_received()); | |
1609 EXPECT_EQ(0, callee_pc_wrapper()->video_frames_received()); | |
1610 // Sanity check that the callee's description has a rejected video section. | |
1611 ASSERT_NE(nullptr, callee_pc_wrapper()->pc()->local_description()); | |
1612 const ContentInfo* callee_video_content = GetFirstVideoContent( | |
1613 callee_pc_wrapper()->pc()->local_description()->description()); | |
1614 ASSERT_NE(nullptr, callee_video_content); | |
1615 EXPECT_TRUE(callee_video_content->rejected); | |
1616 } | |
1617 | |
1618 // Test that if the answerer rejects both audio and video m= sections, nothing | |
1619 // bad happens. | |
1620 // TODO(deadbeef): Test that a data channel still works. Currently this doesn't | |
1621 // test anything but the fact that negotiation succeeds, which doesn't mean | |
1622 // much. | |
1623 TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) { | |
1624 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1625 ConnectFakeSignaling(); | |
1626 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1627 // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it | |
1628 // will reject both audio and video m= sections. | |
1629 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
1630 options.offer_to_receive_audio = 0; | |
1631 options.offer_to_receive_video = 0; | |
1632 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
1633 // Do offer/answer and wait for stable signaling state. | |
1634 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1635 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1636 // Sanity check that the callee's description has rejected m= sections. | |
1637 ASSERT_NE(nullptr, callee_pc_wrapper()->pc()->local_description()); | |
1638 const ContentInfo* callee_audio_content = GetFirstAudioContent( | |
1639 callee_pc_wrapper()->pc()->local_description()->description()); | |
1640 ASSERT_NE(nullptr, callee_audio_content); | |
1641 EXPECT_TRUE(callee_audio_content->rejected); | |
1642 const ContentInfo* callee_video_content = GetFirstVideoContent( | |
1643 callee_pc_wrapper()->pc()->local_description()->description()); | |
1644 ASSERT_NE(nullptr, callee_video_content); | |
1645 EXPECT_TRUE(callee_video_content->rejected); | |
1646 } | |
1647 | |
1648 // This test sets up an audio and video call between two parties. After the | |
1649 // call runs for a while, the caller sends an updated offer with video being | |
1650 // rejected. Once the re-negotiation is done, the video flow should stop and | |
1651 // the audio flow should continue. | |
1652 TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) { | |
1653 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1654 ConnectFakeSignaling(); | |
1655 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1656 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1657 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1658 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1659 EXPECT_TRUE_WAIT( | |
1660 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1661 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1662 kEndVideoFrameCount) && | |
1663 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1664 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1665 kEndVideoFrameCount), | |
1666 kMaxWaitForFramesMs); | |
1667 | |
1668 // Renegotiate, rejecting the video m= section. | |
1669 // TODO(deadbeef): When an RtpTransceiver API is available, use that to | |
1670 // reject the video m= section. | |
1671 caller_pc_wrapper()->SetGeneratedSdpMunger( | |
1672 [](cricket::SessionDescription* description) { | |
1673 for (cricket::ContentInfo& content : description->contents()) { | |
1674 if (cricket::IsVideoContent(&content)) { | |
1675 content.rejected = true; | |
1676 } | |
1677 } | |
1678 }); | |
1679 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1680 ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); | |
1681 | |
1682 // Sanity check that the caller's description has a rejected video section. | |
1683 ASSERT_NE(nullptr, caller_pc_wrapper()->pc()->local_description()); | |
1684 const ContentInfo* caller_video_content = GetFirstVideoContent( | |
1685 caller_pc_wrapper()->pc()->local_description()->description()); | |
1686 ASSERT_NE(nullptr, caller_video_content); | |
1687 EXPECT_TRUE(caller_video_content->rejected); | |
1688 | |
1689 int caller_audio_received = caller_pc_wrapper()->audio_frames_received(); | |
1690 int caller_video_received = caller_pc_wrapper()->video_frames_received(); | |
1691 int callee_audio_received = callee_pc_wrapper()->audio_frames_received(); | |
1692 int callee_video_received = callee_pc_wrapper()->video_frames_received(); | |
1693 | |
1694 // Wait for some additional audio frames to be received. | |
1695 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedAudioFrames( | |
1696 caller_audio_received + kEndAudioFrameCount) && | |
1697 callee_pc_wrapper()->ReceivedAudioFrames( | |
1698 callee_audio_received + kEndAudioFrameCount), | |
1699 kMaxWaitForFramesMs); | |
1700 | |
1701 // During this time, we shouldn't have received any additional video frames | |
1702 // for the rejected video tracks. | |
1703 EXPECT_EQ(caller_video_received, | |
1704 caller_pc_wrapper()->video_frames_received()); | |
1705 EXPECT_EQ(callee_video_received, | |
1706 callee_pc_wrapper()->video_frames_received()); | |
1707 } | |
1708 | |
1709 // Basic end-to-end test, but without SSRC/MSID signaling. This functionality | |
1710 // is needed to support legacy endpoints. | |
1711 // TODO(deadbeef): When we support the MID extension and demuxing on MID, also | |
1712 // add a test for an end-to-end test without MID signaling either (basically, | |
1713 // the minimum acceptable SDP). | |
1714 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) { | |
1715 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1716 ConnectFakeSignaling(); | |
1717 // Add audio and video, testing that packets can be demuxed on payload type. | |
1718 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1719 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1720 // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic" | |
1721 // attribute from received SDP, simulating a legacy endpoint. | |
1722 callee_pc_wrapper()->SetReceivedSdpMunger( | |
1723 [](cricket::SessionDescription* desc) { | |
1724 for (ContentInfo& content : desc->contents()) { | |
1725 MediaContentDescription* media_desc = | |
1726 static_cast<MediaContentDescription*>(content.description); | |
1727 media_desc->mutable_streams().clear(); | |
1728 } | |
1729 desc->set_msid_supported(false); | |
1730 }); | |
1731 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1732 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1733 EXPECT_TRUE_WAIT( | |
1734 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1735 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1736 kEndVideoFrameCount) && | |
1737 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1738 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1739 kEndVideoFrameCount), | |
1740 kMaxWaitForFramesMs); | |
1741 } | |
1742 | |
1743 // Test that if two video tracks are sent (from caller to callee, in this test), | |
1744 // they're transmitted correctly end-to-end. | |
1745 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) { | |
1746 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1747 ConnectFakeSignaling(); | |
1748 // Add one audio/video stream, and one video-only stream. | |
1749 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1750 caller_pc_wrapper()->AddMediaStreamFromTracksWithLabel( | |
1751 nullptr, caller_pc_wrapper()->CreateLocalVideoTrackWithId("extra_track"), | |
1752 "extra_stream"); | |
1753 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1754 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1755 ASSERT_EQ(2u, callee_pc_wrapper()->number_of_remote_streams()); | |
1756 EXPECT_TRUE_WAIT( | |
1757 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1758 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1759 kEndVideoFrameCount), | |
1760 kMaxWaitForFramesMs); | |
1761 } | |
1762 | |
1763 // Test that if applying a true "max bundle" offer, which uses ports of 0, | |
1764 // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and | |
1765 // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes | |
1766 // successfully and media flows. | |
1767 // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works. | |
1768 // TODO(deadbeef): Won't need this test once we start generating actual | |
1769 // standards-compliant SDP. | |
1770 TEST_F(PeerConnectionIntegrationTest, | |
1771 EndToEndCallWithSpecCompliantMaxBundleOffer) { | |
1772 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1773 ConnectFakeSignaling(); | |
1774 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1775 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1776 // Do the equivalent of setting the port to 0, adding a=bundle-only, and | |
1777 // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all | |
1778 // but the first m= section. | |
1779 callee_pc_wrapper()->SetReceivedSdpMunger( | |
1780 [](cricket::SessionDescription* desc) { | |
1781 bool first = true; | |
1782 for (cricket::ContentInfo& content : desc->contents()) { | |
1783 if (first) { | |
1784 first = false; | |
1785 continue; | |
1786 } | |
1787 content.bundle_only = true; | |
1788 } | |
1789 first = true; | |
1790 for (cricket::TransportInfo& transport : desc->transport_infos()) { | |
1791 if (first) { | |
1792 first = false; | |
1793 continue; | |
1794 } | |
1795 transport.description.ice_ufrag.clear(); | |
1796 transport.description.ice_pwd.clear(); | |
1797 transport.description.connection_role = cricket::CONNECTIONROLE_NONE; | |
1798 transport.description.identity_fingerprint.reset(nullptr); | |
1799 } | |
1800 }); | |
pthatcher1
2017/03/20 18:23:17
Would it be worth giving this a named method? cal
Taylor Brandstetter
2017/03/23 04:46:25
Done.
| |
1801 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1802 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1803 EXPECT_TRUE_WAIT( | |
1804 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1805 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1806 kEndVideoFrameCount) && | |
1807 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1808 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1809 kEndVideoFrameCount), | |
1810 kMaxWaitForFramesMs); | |
1811 } | |
1812 | |
1813 // Test that we can receive the audio output level from a remote audio track. | |
1814 // TODO(deadbeef): Use a fake audio source and verify that the output level is | |
1815 // exactly what the source on the other side was configured with. | |
1816 TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStats) { | |
1817 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1818 ConnectFakeSignaling(); | |
1819 // Just add an audio track. | |
1820 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1821 caller_pc_wrapper()->CreateLocalAudioTrack(), nullptr); | |
1822 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1823 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1824 | |
1825 // Get the audio output level stats. Note that the level is not available | |
1826 // until an RTCP packet has been received. | |
1827 EXPECT_TRUE_WAIT(callee_pc_wrapper()->GetStats()->AudioOutputLevel() > 0, | |
1828 kMaxWaitForFramesMs); | |
1829 } | |
1830 | |
1831 // Test that an audio input level is reported. | |
1832 // TODO(deadbeef): Use a fake audio source and verify that the input level is | |
1833 // exactly what the source was configured with. | |
1834 TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStats) { | |
1835 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1836 ConnectFakeSignaling(); | |
1837 // Just add an audio track. | |
1838 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
1839 caller_pc_wrapper()->CreateLocalAudioTrack(), nullptr); | |
1840 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1841 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1842 | |
1843 // Get the audio input level stats. The level should be available very | |
1844 // soon after the test starts. | |
1845 EXPECT_TRUE_WAIT(caller_pc_wrapper()->GetStats()->AudioInputLevel() > 0, | |
1846 kMaxWaitForStatsMs); | |
1847 } | |
1848 | |
1849 // Test that we can get incoming byte counts from both audio and video tracks. | |
1850 TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStats) { | |
1851 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1852 ConnectFakeSignaling(); | |
1853 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1854 // Do offer/answer, wait for the callee to receive some frames. | |
1855 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1856 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1857 EXPECT_TRUE_WAIT( | |
1858 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1859 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1860 kEndVideoFrameCount), | |
1861 kMaxWaitForFramesMs); | |
1862 | |
1863 // Get a handle to the remote tracks created, so they can be used as GetStats | |
1864 // filters. | |
1865 StreamCollectionInterface* remote_streams = | |
1866 callee_pc_wrapper()->remote_streams(); | |
1867 ASSERT_EQ(1u, remote_streams->count()); | |
1868 ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size()); | |
1869 ASSERT_EQ(1u, remote_streams->at(0)->GetVideoTracks().size()); | |
1870 MediaStreamTrackInterface* remote_audio_track = | |
1871 remote_streams->at(0)->GetAudioTracks()[0]; | |
1872 MediaStreamTrackInterface* remote_video_track = | |
1873 remote_streams->at(0)->GetVideoTracks()[0]; | |
1874 | |
1875 // We received frames, so we definitely should have nonzero "received bytes" | |
1876 // stats at this point. | |
1877 EXPECT_GT(callee_pc_wrapper() | |
1878 ->GetStatsForTrack(remote_audio_track) | |
1879 ->BytesReceived(), | |
1880 0); | |
1881 EXPECT_GT(callee_pc_wrapper() | |
1882 ->GetStatsForTrack(remote_video_track) | |
1883 ->BytesReceived(), | |
1884 0); | |
1885 } | |
1886 | |
1887 // Test that we can get outgoing byte counts from both audio and video tracks. | |
1888 TEST_F(PeerConnectionIntegrationTest, GetBytesSentStats) { | |
1889 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
1890 ConnectFakeSignaling(); | |
1891 auto audio_track = caller_pc_wrapper()->CreateLocalAudioTrack(); | |
1892 auto video_track = caller_pc_wrapper()->CreateLocalVideoTrack(); | |
1893 caller_pc_wrapper()->AddMediaStreamFromTracks(audio_track, video_track); | |
1894 // Do offer/answer, wait for the callee to receive some frames. | |
1895 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1896 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1897 EXPECT_TRUE_WAIT( | |
1898 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1899 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1900 kEndVideoFrameCount), | |
1901 kMaxWaitForFramesMs); | |
1902 | |
1903 // The callee received frames, so we definitely should have nonzero "sent | |
1904 // bytes" stats at this point. | |
1905 EXPECT_GT(caller_pc_wrapper()->GetStatsForTrack(audio_track)->BytesSent(), 0); | |
1906 EXPECT_GT(caller_pc_wrapper()->GetStatsForTrack(video_track)->BytesSent(), 0); | |
1907 } | |
1908 | |
1909 // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. | |
1910 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) { | |
1911 PeerConnectionFactory::Options dtls_10_options; | |
1912 dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
1913 ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, | |
1914 dtls_10_options)); | |
1915 ConnectFakeSignaling(); | |
1916 // Do normal offer/answer and wait for some frames to be received in each | |
1917 // direction. | |
1918 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1919 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1920 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1921 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1922 EXPECT_TRUE_WAIT( | |
1923 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1924 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1925 kEndVideoFrameCount) && | |
1926 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
1927 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
1928 kEndVideoFrameCount), | |
1929 kMaxWaitForFramesMs); | |
1930 } | |
1931 | |
1932 // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated. | |
1933 TEST_F(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) { | |
1934 PeerConnectionFactory::Options dtls_10_options; | |
1935 dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
1936 ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, | |
1937 dtls_10_options)); | |
1938 ConnectFakeSignaling(); | |
1939 // Register UMA observer before signaling begins. | |
1940 rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = | |
1941 new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
1942 caller_pc_wrapper()->pc()->RegisterUMAObserver(caller_observer); | |
1943 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1944 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1945 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1946 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1947 EXPECT_TRUE_WAIT( | |
1948 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
1949 caller_pc_wrapper()->GetStats()->DtlsCipher(), rtc::KT_DEFAULT), | |
1950 kDefaultTimeout); | |
1951 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
1952 caller_pc_wrapper()->GetStats()->SrtpCipher(), | |
1953 kDefaultTimeout); | |
1954 EXPECT_EQ(1, | |
1955 caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
1956 kDefaultSrtpCryptoSuite)); | |
1957 } | |
1958 | |
1959 // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated. | |
1960 TEST_F(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) { | |
1961 PeerConnectionFactory::Options dtls_12_options; | |
1962 dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
1963 ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options, | |
1964 dtls_12_options)); | |
1965 ConnectFakeSignaling(); | |
1966 // Register UMA observer before signaling begins. | |
1967 rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer = | |
1968 new rtc::RefCountedObject<webrtc::FakeMetricsObserver>(); | |
1969 caller_pc_wrapper()->pc()->RegisterUMAObserver(caller_observer); | |
1970 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1971 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
1972 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
1973 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
1974 EXPECT_TRUE_WAIT( | |
1975 rtc::SSLStreamAdapter::IsAcceptableCipher( | |
1976 caller_pc_wrapper()->GetStats()->DtlsCipher(), rtc::KT_DEFAULT), | |
1977 kDefaultTimeout); | |
1978 EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), | |
1979 caller_pc_wrapper()->GetStats()->SrtpCipher(), | |
1980 kDefaultTimeout); | |
1981 EXPECT_EQ(1, | |
1982 caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher, | |
1983 kDefaultSrtpCryptoSuite)); | |
1984 } | |
1985 | |
1986 // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the | |
1987 // callee only supports 1.0. | |
1988 TEST_F(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) { | |
1989 PeerConnectionFactory::Options caller_options; | |
1990 caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
1991 PeerConnectionFactory::Options callee_options; | |
1992 callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
1993 ASSERT_TRUE( | |
1994 CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); | |
1995 ConnectFakeSignaling(); | |
1996 // Do normal offer/answer and wait for some frames to be received in each | |
1997 // direction. | |
1998 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
1999 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2000 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2001 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2002 EXPECT_TRUE_WAIT( | |
2003 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2004 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2005 kEndVideoFrameCount) && | |
2006 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2007 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2008 kEndVideoFrameCount), | |
2009 kMaxWaitForFramesMs); | |
2010 } | |
2011 | |
2012 // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the | |
2013 // callee supports 1.2. | |
2014 TEST_F(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) { | |
2015 PeerConnectionFactory::Options caller_options; | |
2016 caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; | |
2017 PeerConnectionFactory::Options callee_options; | |
2018 callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; | |
2019 ASSERT_TRUE( | |
2020 CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); | |
2021 ConnectFakeSignaling(); | |
2022 // Do normal offer/answer and wait for some frames to be received in each | |
2023 // direction. | |
2024 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2025 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2026 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2027 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2028 EXPECT_TRUE_WAIT( | |
2029 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2030 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2031 kEndVideoFrameCount) && | |
2032 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2033 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2034 kEndVideoFrameCount), | |
2035 kMaxWaitForFramesMs); | |
2036 } | |
2037 | |
2038 // Test that a non-GCM cipher is used if both sides only support non-GCM. | |
2039 TEST_F(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) { | |
2040 bool local_gcm_enabled = false; | |
2041 bool remote_gcm_enabled = false; | |
2042 int expected_cipher_suite = kDefaultSrtpCryptoSuite; | |
2043 TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, | |
2044 expected_cipher_suite); | |
2045 } | |
2046 | |
2047 // Test that a GCM cipher is used if both ends support it. | |
2048 TEST_F(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) { | |
2049 bool local_gcm_enabled = true; | |
2050 bool remote_gcm_enabled = true; | |
2051 int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm; | |
2052 TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, | |
2053 expected_cipher_suite); | |
2054 } | |
2055 | |
2056 // Test that GCM isn't used if only the offerer supports it. | |
2057 TEST_F(PeerConnectionIntegrationTest, | |
2058 NonGcmCipherUsedWhenOnlyCallerSupportsGcm) { | |
2059 bool local_gcm_enabled = true; | |
2060 bool remote_gcm_enabled = false; | |
2061 int expected_cipher_suite = kDefaultSrtpCryptoSuite; | |
2062 TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, | |
2063 expected_cipher_suite); | |
2064 } | |
2065 | |
2066 // Test that GCM isn't used if only the answerer supports it. | |
2067 TEST_F(PeerConnectionIntegrationTest, | |
2068 NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) { | |
2069 bool local_gcm_enabled = false; | |
2070 bool remote_gcm_enabled = true; | |
2071 int expected_cipher_suite = kDefaultSrtpCryptoSuite; | |
2072 TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, | |
2073 expected_cipher_suite); | |
2074 } | |
2075 | |
2076 // This test sets up a call between two parties with audio, video and an RTP | |
2077 // data channel. | |
2078 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) { | |
2079 FakeConstraints setup_constraints; | |
2080 setup_constraints.SetAllowRtpDataChannels(); | |
2081 ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, | |
2082 &setup_constraints)); | |
2083 ConnectFakeSignaling(); | |
2084 // Expect that data channel created on caller side will show up for callee as | |
2085 // well. | |
2086 caller_pc_wrapper()->CreateDataChannel(); | |
2087 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2088 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2089 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2090 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2091 // Ensure the existence of the RTP data channel didn't impede audio/video. | |
2092 EXPECT_TRUE_WAIT( | |
2093 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2094 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2095 kEndVideoFrameCount) && | |
2096 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2097 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2098 kEndVideoFrameCount), | |
2099 kMaxWaitForFramesMs); | |
2100 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2101 ASSERT_NE(nullptr, callee_pc_wrapper()->data_channel()); | |
2102 EXPECT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2103 kDefaultTimeout); | |
2104 EXPECT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2105 kDefaultTimeout); | |
2106 | |
2107 // Ensure data can be sent in both directions. | |
2108 std::string data = "hello world"; | |
2109 SendRtpDataWithRetries(caller_pc_wrapper()->data_channel(), data, 5); | |
2110 EXPECT_EQ_WAIT(data, callee_pc_wrapper()->data_observer()->last_message(), | |
2111 kDefaultTimeout); | |
2112 SendRtpDataWithRetries(callee_pc_wrapper()->data_channel(), data, 5); | |
2113 EXPECT_EQ_WAIT(data, caller_pc_wrapper()->data_observer()->last_message(), | |
2114 kDefaultTimeout); | |
2115 } | |
2116 | |
2117 // Ensure that an RTP data channel is signaled as closed for the caller when | |
2118 // the callee rejects it in a subsequent offer. | |
2119 TEST_F(PeerConnectionIntegrationTest, | |
2120 RtpDataChannelSignaledClosedInCalleeOffer) { | |
2121 // Same procedure as above test. | |
2122 FakeConstraints setup_constraints; | |
2123 setup_constraints.SetAllowRtpDataChannels(); | |
2124 ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, | |
2125 &setup_constraints)); | |
2126 ConnectFakeSignaling(); | |
2127 caller_pc_wrapper()->CreateDataChannel(); | |
2128 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2129 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2130 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2131 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2132 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2133 ASSERT_NE(nullptr, callee_pc_wrapper()->data_channel()); | |
2134 ASSERT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2135 kDefaultTimeout); | |
2136 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2137 kDefaultTimeout); | |
2138 | |
2139 // Close the data channel on the callee, and do an updated offer/answer. | |
2140 callee_pc_wrapper()->data_channel()->Close(); | |
2141 callee_pc_wrapper()->CreateSetAndSignalOffer(); | |
2142 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2143 EXPECT_FALSE(caller_pc_wrapper()->data_observer()->IsOpen()); | |
2144 EXPECT_FALSE(callee_pc_wrapper()->data_observer()->IsOpen()); | |
2145 } | |
2146 | |
2147 // Tests that data is buffered in an RTP data channel until an observer is | |
2148 // registered for it. | |
2149 // | |
2150 // NOTE: RTP data channels can receive data before the underlying | |
2151 // transport has detected that a channel is writable and thus data can be | |
2152 // received before the data channel state changes to open. That is hard to test | |
2153 // but the same buffering is expected to be used in that case. | |
2154 TEST_F(PeerConnectionIntegrationTest, | |
2155 DataBufferedUntilRtpDataChannelObserverRegistered) { | |
2156 // Use fake clock and simulated network delay so that we predictably can wait | |
2157 // until an SCTP message has been delivered without "sleep()"ing. | |
2158 rtc::ScopedFakeClock fake_clock; | |
2159 // Some things use a time of "0" as a special value, so we need to start out | |
2160 // the fake clock at a nonzero time. | |
2161 // TODO(deadbeef): Fix this. | |
2162 fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1)); | |
2163 virtual_socket_server()->set_delay_mean(5); // 5 ms per hop. | |
2164 virtual_socket_server()->UpdateDelayDistribution(); | |
2165 | |
2166 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2167 ConnectFakeSignaling(); | |
2168 caller_pc_wrapper()->CreateDataChannel(); | |
2169 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2170 ASSERT_TRUE(caller_pc_wrapper()->data_channel() != nullptr); | |
2171 ASSERT_TRUE_SIMULATED_WAIT(callee_pc_wrapper()->data_channel() != nullptr, | |
2172 kDefaultTimeout, fake_clock); | |
2173 ASSERT_TRUE_SIMULATED_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2174 kDefaultTimeout, fake_clock); | |
2175 ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen, | |
2176 callee_pc_wrapper()->data_channel()->state(), | |
2177 kDefaultTimeout, fake_clock); | |
2178 | |
2179 // Unregister the observer which is normally automatically registered. | |
2180 callee_pc_wrapper()->data_channel()->UnregisterObserver(); | |
2181 // Send data and advance fake clock until it should have been received. | |
2182 std::string data = "hello world"; | |
2183 caller_pc_wrapper()->data_channel()->Send(DataBuffer(data)); | |
2184 SIMULATED_WAIT(false, 50, fake_clock); | |
2185 | |
2186 // Attach data channel and expect data to be received immediately. Note that | |
2187 // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any | |
2188 // further, but data can be received even if the callback is asynchronous. | |
2189 MockDataChannelObserver new_observer(callee_pc_wrapper()->data_channel()); | |
2190 EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout, | |
2191 fake_clock); | |
2192 } | |
2193 | |
2194 // This test sets up a call between two parties with audio, video and but only | |
2195 // the caller client supports RTP data channels. | |
2196 TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) { | |
2197 FakeConstraints setup_constraints_1; | |
2198 setup_constraints_1.SetAllowRtpDataChannels(); | |
2199 // Must disable DTLS to make negotiation succeed. | |
2200 setup_constraints_1.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
2201 false); | |
2202 FakeConstraints setup_constraints_2; | |
2203 setup_constraints_2.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, | |
2204 false); | |
2205 ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints( | |
2206 &setup_constraints_1, &setup_constraints_2)); | |
2207 ConnectFakeSignaling(); | |
2208 caller_pc_wrapper()->CreateDataChannel(); | |
2209 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2210 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2211 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2212 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2213 // The caller should still have a data channel, but it should be closed, and | |
2214 // one should ever have been created for the callee. | |
2215 EXPECT_TRUE(caller_pc_wrapper()->data_channel() != nullptr); | |
2216 EXPECT_FALSE(caller_pc_wrapper()->data_observer()->IsOpen()); | |
2217 EXPECT_EQ(nullptr, callee_pc_wrapper()->data_channel()); | |
2218 } | |
2219 | |
2220 // This test sets up a call between two parties with audio, and video. When | |
2221 // audio and video is setup and flowing, an RTP data channel is negotiated. | |
2222 TEST_F(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) { | |
2223 FakeConstraints setup_constraints; | |
2224 setup_constraints.SetAllowRtpDataChannels(); | |
2225 ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints, | |
2226 &setup_constraints)); | |
2227 ConnectFakeSignaling(); | |
2228 // Do initial offer/answer with audio/video. | |
2229 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2230 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2231 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2232 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2233 // Create data channel and do new offer and answer. | |
2234 caller_pc_wrapper()->CreateDataChannel(); | |
2235 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2236 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2237 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2238 ASSERT_NE(nullptr, callee_pc_wrapper()->data_channel()); | |
2239 EXPECT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2240 kDefaultTimeout); | |
2241 EXPECT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2242 kDefaultTimeout); | |
2243 // Ensure data can be sent in both directions. | |
2244 std::string data = "hello world"; | |
2245 SendRtpDataWithRetries(caller_pc_wrapper()->data_channel(), data, 5); | |
2246 EXPECT_EQ_WAIT(data, callee_pc_wrapper()->data_observer()->last_message(), | |
2247 kDefaultTimeout); | |
2248 SendRtpDataWithRetries(callee_pc_wrapper()->data_channel(), data, 5); | |
2249 EXPECT_EQ_WAIT(data, caller_pc_wrapper()->data_observer()->last_message(), | |
2250 kDefaultTimeout); | |
2251 } | |
2252 | |
2253 #ifdef HAVE_SCTP | |
2254 | |
2255 // This test sets up a call between two parties with audio, video and an SCTP | |
2256 // data channel. | |
2257 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) { | |
2258 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2259 ConnectFakeSignaling(); | |
2260 // Expect that data channel created on caller side will show up for callee as | |
2261 // well. | |
2262 caller_pc_wrapper()->CreateDataChannel(); | |
2263 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2264 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2265 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2266 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2267 // Ensure the existence of the SCTP data channel didn't impede audio/video. | |
2268 EXPECT_TRUE_WAIT( | |
2269 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2270 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2271 kEndVideoFrameCount) && | |
2272 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2273 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2274 kEndVideoFrameCount), | |
2275 kMaxWaitForFramesMs); | |
2276 // Caller data channel should already exist (it created one). Callee data | |
2277 // channel may not exist yet, since negotiation happens in-band, not in SDP. | |
2278 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2279 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_channel() != nullptr, | |
2280 kDefaultTimeout); | |
2281 EXPECT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2282 kDefaultTimeout); | |
2283 EXPECT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2284 kDefaultTimeout); | |
2285 | |
2286 // Ensure data can be sent in both directions. | |
2287 std::string data = "hello world"; | |
2288 caller_pc_wrapper()->data_channel()->Send(DataBuffer(data)); | |
2289 EXPECT_EQ_WAIT(data, callee_pc_wrapper()->data_observer()->last_message(), | |
2290 kDefaultTimeout); | |
2291 callee_pc_wrapper()->data_channel()->Send(DataBuffer(data)); | |
2292 EXPECT_EQ_WAIT(data, caller_pc_wrapper()->data_observer()->last_message(), | |
2293 kDefaultTimeout); | |
2294 } | |
2295 | |
2296 // Ensure that when the callee closes an SCTP data channel, the closing | |
2297 // procedure results in the data channel being closed for the caller as well. | |
2298 TEST_F(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) { | |
2299 // Same procedure as above test. | |
2300 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2301 ConnectFakeSignaling(); | |
2302 caller_pc_wrapper()->CreateDataChannel(); | |
2303 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2304 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2305 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2306 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2307 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2308 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_channel() != nullptr, | |
2309 kDefaultTimeout); | |
2310 ASSERT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2311 kDefaultTimeout); | |
2312 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2313 kDefaultTimeout); | |
2314 | |
2315 // Close the data channel on the callee side, and wait for it to reach the | |
2316 // "closed" state on both sides. | |
2317 callee_pc_wrapper()->data_channel()->Close(); | |
2318 EXPECT_TRUE_WAIT(!caller_pc_wrapper()->data_observer()->IsOpen(), | |
2319 kDefaultTimeout); | |
2320 EXPECT_TRUE_WAIT(!callee_pc_wrapper()->data_observer()->IsOpen(), | |
2321 kDefaultTimeout); | |
2322 } | |
2323 | |
2324 // Test usrsctp's ability to process unordered data stream, where data actually | |
2325 // arrives out of order using simulated delays. Previously there have been some | |
2326 // bugs in this area. | |
2327 TEST_F(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) { | |
2328 // Introduce random network delays. | |
2329 // Otherwise it's not a true "unordered" test. | |
2330 virtual_socket_server()->set_delay_mean(20); | |
2331 virtual_socket_server()->set_delay_stddev(5); | |
2332 virtual_socket_server()->UpdateDelayDistribution(); | |
2333 // Normal procedure, but with unordered data channel config. | |
2334 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2335 ConnectFakeSignaling(); | |
2336 webrtc::DataChannelInit init; | |
2337 init.ordered = false; | |
2338 caller_pc_wrapper()->CreateDataChannel(&init); | |
2339 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2340 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2341 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2342 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_channel() != nullptr, | |
2343 kDefaultTimeout); | |
2344 ASSERT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2345 kDefaultTimeout); | |
2346 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2347 kDefaultTimeout); | |
2348 | |
2349 static constexpr int kNumMessages = 100; | |
2350 // Deliberately chosen to be larger than the MTU so messages get fragmented. | |
2351 static constexpr size_t kMaxMessageSize = 4096; | |
2352 // Create and send random messages. | |
2353 std::vector<std::string> sent_messages; | |
2354 for (int i = 0; i < kNumMessages; ++i) { | |
2355 size_t length = | |
2356 (rand() % kMaxMessageSize) + 1; // NOLINT (rand_r instead of rand) | |
2357 std::string message; | |
2358 ASSERT_TRUE(rtc::CreateRandomString(length, &message)); | |
2359 caller_pc_wrapper()->data_channel()->Send(DataBuffer(message)); | |
2360 callee_pc_wrapper()->data_channel()->Send(DataBuffer(message)); | |
2361 sent_messages.push_back(message); | |
2362 } | |
2363 | |
2364 // Wait for all messages to be received. | |
2365 EXPECT_EQ_WAIT(kNumMessages, | |
2366 caller_pc_wrapper()->data_observer()->received_message_count(), | |
2367 kDefaultTimeout); | |
2368 EXPECT_EQ_WAIT(kNumMessages, | |
2369 callee_pc_wrapper()->data_observer()->received_message_count(), | |
2370 kDefaultTimeout); | |
2371 | |
2372 // Sort and compare to make sure none of the messages were corrupted. | |
2373 std::vector<std::string> caller_pc_wrapper_received_messages = | |
2374 caller_pc_wrapper()->data_observer()->messages(); | |
2375 std::vector<std::string> callee_pc_wrapper_received_messages = | |
2376 callee_pc_wrapper()->data_observer()->messages(); | |
2377 std::sort(sent_messages.begin(), sent_messages.end()); | |
2378 std::sort(caller_pc_wrapper_received_messages.begin(), | |
2379 caller_pc_wrapper_received_messages.end()); | |
2380 std::sort(callee_pc_wrapper_received_messages.begin(), | |
2381 callee_pc_wrapper_received_messages.end()); | |
2382 EXPECT_EQ(sent_messages, caller_pc_wrapper_received_messages); | |
2383 EXPECT_EQ(sent_messages, callee_pc_wrapper_received_messages); | |
2384 } | |
2385 | |
2386 // This test sets up a call between two parties with audio, and video. When | |
2387 // audio and video are setup and flowing, an SCTP data channel is negotiated. | |
2388 TEST_F(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) { | |
2389 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2390 ConnectFakeSignaling(); | |
2391 // Do initial offer/answer with audio/video. | |
2392 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2393 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2394 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2395 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2396 // Create data channel and do new offer and answer. | |
2397 caller_pc_wrapper()->CreateDataChannel(); | |
2398 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2399 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2400 // Caller data channel should already exist (it created one). Callee data | |
2401 // channel may not exist yet, since negotiation happens in-band, not in SDP. | |
2402 ASSERT_NE(nullptr, caller_pc_wrapper()->data_channel()); | |
2403 ASSERT_TRUE_WAIT(callee_pc_wrapper()->data_channel() != nullptr, | |
2404 kDefaultTimeout); | |
2405 EXPECT_TRUE_WAIT(caller_pc_wrapper()->data_observer()->IsOpen(), | |
2406 kDefaultTimeout); | |
2407 EXPECT_TRUE_WAIT(callee_pc_wrapper()->data_observer()->IsOpen(), | |
2408 kDefaultTimeout); | |
2409 // Ensure data can be sent in both directions. | |
2410 std::string data = "hello world"; | |
2411 caller_pc_wrapper()->data_channel()->Send(DataBuffer(data)); | |
2412 EXPECT_EQ_WAIT(data, callee_pc_wrapper()->data_observer()->last_message(), | |
2413 kDefaultTimeout); | |
2414 callee_pc_wrapper()->data_channel()->Send(DataBuffer(data)); | |
2415 EXPECT_EQ_WAIT(data, caller_pc_wrapper()->data_observer()->last_message(), | |
2416 kDefaultTimeout); | |
2417 } | |
2418 | |
2419 #endif // HAVE_SCTP | |
2420 | |
2421 // Test that the ICE connection and gathering states eventually reach | |
2422 // "complete". | |
2423 TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) { | |
2424 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2425 ConnectFakeSignaling(); | |
2426 // Do normal offer/answer. | |
2427 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2428 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2429 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2430 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2431 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, | |
2432 caller_pc_wrapper()->ice_gathering_state(), | |
2433 kMaxWaitForFramesMs); | |
2434 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete, | |
2435 callee_pc_wrapper()->ice_gathering_state(), | |
2436 kMaxWaitForFramesMs); | |
2437 // After the best candidate pair is selected and all candidates are signaled, | |
2438 // the ICE connection state should reach "complete". | |
2439 // TODO(deadbeef): Currently, the ICE "controlled" agent (the | |
2440 // answerer/"callee" by default) only reaches "connected". When this is | |
2441 // fixed, this test should be updated. | |
2442 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
2443 caller_pc_wrapper()->ice_connection_state(), kDefaultTimeout); | |
2444 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
2445 callee_pc_wrapper()->ice_connection_state(), kDefaultTimeout); | |
2446 } | |
2447 | |
2448 // This test sets up a call between two parties with audio and video. | |
2449 // During the call, the caller restarts ICE and the test verifies that | |
2450 // new ICE candidates are generated and audio and video still can flow, and the | |
2451 // ICE state reaches completed again. | |
2452 TEST_F(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) { | |
2453 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2454 ConnectFakeSignaling(); | |
2455 // Do normal offer/answer and wait for ICE to complete. | |
2456 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2457 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2458 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2459 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2460 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
2461 caller_pc_wrapper()->ice_connection_state(), | |
2462 kMaxWaitForFramesMs); | |
2463 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
2464 callee_pc_wrapper()->ice_connection_state(), | |
2465 kMaxWaitForFramesMs); | |
2466 | |
2467 // To verify that the ICE restart actually occurs, get | |
2468 // ufrag/password/candidates before and after restart. | |
2469 // Create an SDP string of the first audio candidate for both clients. | |
2470 const webrtc::IceCandidateCollection* audio_candidates_caller = | |
2471 caller_pc_wrapper()->pc()->local_description()->candidates(0); | |
2472 const webrtc::IceCandidateCollection* audio_candidates_callee = | |
2473 callee_pc_wrapper()->pc()->local_description()->candidates(0); | |
2474 ASSERT_GT(audio_candidates_caller->count(), 0u); | |
2475 ASSERT_GT(audio_candidates_callee->count(), 0u); | |
2476 std::string caller_candidate_pre_restart; | |
2477 ASSERT_TRUE( | |
2478 audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart)); | |
2479 std::string callee_candidate_pre_restart; | |
2480 ASSERT_TRUE( | |
2481 audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart)); | |
2482 const cricket::SessionDescription* desc = | |
2483 caller_pc_wrapper()->pc()->local_description()->description(); | |
2484 std::string caller_ufrag_pre_restart = | |
2485 desc->transport_infos()[0].description.ice_ufrag; | |
2486 desc = callee_pc_wrapper()->pc()->local_description()->description(); | |
2487 std::string callee_ufrag_pre_restart = | |
2488 desc->transport_infos()[0].description.ice_ufrag; | |
2489 | |
2490 // Have the caller initiate an ICE restart. | |
2491 caller_pc_wrapper()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); | |
2492 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2493 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2494 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
2495 caller_pc_wrapper()->ice_connection_state(), | |
2496 kMaxWaitForFramesMs); | |
2497 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
2498 callee_pc_wrapper()->ice_connection_state(), | |
2499 kMaxWaitForFramesMs); | |
2500 | |
2501 // Grab the ufrags/candidates again. | |
2502 audio_candidates_caller = | |
2503 caller_pc_wrapper()->pc()->local_description()->candidates(0); | |
2504 audio_candidates_callee = | |
2505 callee_pc_wrapper()->pc()->local_description()->candidates(0); | |
2506 ASSERT_GT(audio_candidates_caller->count(), 0u); | |
2507 ASSERT_GT(audio_candidates_callee->count(), 0u); | |
2508 std::string caller_candidate_post_restart; | |
2509 ASSERT_TRUE( | |
2510 audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart)); | |
2511 std::string callee_candidate_post_restart; | |
2512 ASSERT_TRUE( | |
2513 audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart)); | |
2514 desc = caller_pc_wrapper()->pc()->local_description()->description(); | |
2515 std::string caller_ufrag_post_restart = | |
2516 desc->transport_infos()[0].description.ice_ufrag; | |
2517 desc = callee_pc_wrapper()->pc()->local_description()->description(); | |
2518 std::string callee_ufrag_post_restart = | |
2519 desc->transport_infos()[0].description.ice_ufrag; | |
2520 // Sanity check that an ICE restart was actually negotiated in SDP. | |
2521 ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart); | |
2522 ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart); | |
2523 ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart); | |
2524 ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart); | |
2525 | |
2526 // Ensure that additional frames are received after the ICE restart. | |
2527 int last_caller_audio_frames = caller_pc_wrapper()->audio_frames_received(); | |
2528 int last_caller_video_frames = caller_pc_wrapper()->video_frames_received(); | |
2529 int last_callee_audio_frames = callee_pc_wrapper()->audio_frames_received(); | |
2530 int last_callee_video_frames = callee_pc_wrapper()->video_frames_received(); | |
2531 EXPECT_TRUE_WAIT(caller_pc_wrapper()->ReceivedAudioFrames( | |
2532 kEndAudioFrameCount + last_caller_audio_frames) && | |
2533 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2534 kEndVideoFrameCount + last_caller_video_frames) && | |
2535 callee_pc_wrapper()->ReceivedAudioFrames( | |
2536 kEndAudioFrameCount + last_callee_audio_frames) && | |
2537 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2538 kEndVideoFrameCount + last_callee_video_frames), | |
2539 kMaxWaitForFramesMs); | |
2540 } | |
2541 | |
2542 // Verify that audio/video can be received end-to-end when ICE renomination is | |
2543 // enabled. | |
2544 TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) { | |
2545 PeerConnectionInterface::RTCConfiguration config; | |
2546 config.enable_ice_renomination = true; | |
2547 ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config)); | |
2548 ConnectFakeSignaling(); | |
2549 // Do normal offer/answer and wait for some frames to be received in each | |
2550 // direction. | |
2551 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2552 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2553 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2554 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2555 // Sanity check that ICE renomination was actually negotiated. | |
2556 const cricket::SessionDescription* desc = | |
2557 caller_pc_wrapper()->pc()->local_description()->description(); | |
2558 for (const cricket::TransportInfo& info : desc->transport_infos()) { | |
2559 ASSERT_NE(info.description.transport_options.end(), | |
2560 std::find(info.description.transport_options.begin(), | |
2561 info.description.transport_options.end(), | |
2562 cricket::ICE_RENOMINATION_STR)); | |
2563 } | |
2564 desc = callee_pc_wrapper()->pc()->local_description()->description(); | |
2565 for (const cricket::TransportInfo& info : desc->transport_infos()) { | |
2566 ASSERT_NE(info.description.transport_options.end(), | |
2567 std::find(info.description.transport_options.begin(), | |
2568 info.description.transport_options.end(), | |
2569 cricket::ICE_RENOMINATION_STR)); | |
2570 } | |
2571 EXPECT_TRUE_WAIT( | |
2572 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2573 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2574 kEndVideoFrameCount) && | |
2575 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2576 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2577 kEndVideoFrameCount), | |
2578 kMaxWaitForFramesMs); | |
2579 } | |
2580 | |
2581 // This test sets up a call between two parties with audio and video. It then | |
2582 // renegotiates setting the video m-line to "port 0", then later renegotiates | |
2583 // again, enabling video. | |
2584 TEST_F(PeerConnectionIntegrationTest, | |
2585 VideoFlowsAfterMediaSectionIsRejectedAndRecycled) { | |
2586 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2587 ConnectFakeSignaling(); | |
2588 | |
2589 // Do initial negotiation, only sending media from the caller. Will result in | |
2590 // video and audio recvonly "m=" sections. | |
2591 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2592 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2593 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2594 | |
2595 // Negotiate again, disabling the video "m=" section (the callee will set the | |
2596 // port to 0 due to offer_to_receive_video = 0). | |
2597 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
2598 options.offer_to_receive_video = 0; | |
2599 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
2600 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2601 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2602 // Sanity check that video "m=" section was actually rejected. | |
2603 const ContentInfo* answer_video_content = cricket::GetFirstVideoContent( | |
2604 callee_pc_wrapper()->pc()->local_description()->description()); | |
2605 ASSERT_NE(nullptr, answer_video_content); | |
2606 ASSERT_TRUE(answer_video_content->rejected); | |
2607 | |
2608 // Enable video and do negotiation again, making sure video is received | |
2609 // end-to-end, also adding media stream to callee. | |
2610 options.offer_to_receive_video = 1; | |
2611 callee_pc_wrapper()->SetOfferAnswerOptions(options); | |
2612 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2613 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2614 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2615 // Verify caller receives frames from the newly added stream, and the callee | |
2616 // receives additional frames from the re-enabled video m= section. | |
2617 int last_callee_audio_frames = callee_pc_wrapper()->audio_frames_received(); | |
2618 int last_callee_video_frames = callee_pc_wrapper()->video_frames_received(); | |
2619 EXPECT_TRUE_WAIT( | |
2620 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2621 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2622 kEndVideoFrameCount) && | |
2623 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount + | |
2624 last_callee_audio_frames) && | |
2625 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2626 kEndVideoFrameCount + last_callee_video_frames), | |
2627 kMaxWaitForFramesMs); | |
2628 } | |
2629 | |
2630 // This test sets up a Jsep call between two parties with external | |
2631 // VideoDecoderFactory. | |
2632 // TODO(holmer): Disabled due to sometimes crashing on buildbots. | |
2633 // See issue webrtc/2378. | |
2634 TEST_F(PeerConnectionIntegrationTest, | |
2635 DISABLED_EndToEndCallWithVideoDecoderFactory) { | |
2636 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2637 EnableVideoDecoderFactory(); | |
2638 ConnectFakeSignaling(); | |
2639 caller_pc_wrapper()->AddAudioVideoMediaStream(); | |
2640 callee_pc_wrapper()->AddAudioVideoMediaStream(); | |
2641 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2642 ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); | |
2643 EXPECT_TRUE_WAIT( | |
2644 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2645 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2646 kEndVideoFrameCount) && | |
2647 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2648 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2649 kEndVideoFrameCount), | |
2650 kMaxWaitForFramesMs); | |
2651 } | |
2652 | |
2653 // This tests that if we negotiate after calling CreateSender but before we | |
2654 // have a track, then set a track later, frames from the newly-set track are | |
2655 // received end-to-end. | |
2656 // TODO(deadbeef): Change this test to use AddTransceiver, once that's | |
2657 // implemented. | |
2658 TEST_F(PeerConnectionIntegrationTest, | |
2659 MediaFlowsAfterEarlyWarmupWithCreateSender) { | |
2660 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2661 ConnectFakeSignaling(); | |
2662 auto caller_audio_sender = | |
2663 caller_pc_wrapper()->pc()->CreateSender("audio", "caller_stream"); | |
2664 auto caller_video_sender = | |
2665 caller_pc_wrapper()->pc()->CreateSender("video", "caller_stream"); | |
2666 auto callee_audio_sender = | |
2667 callee_pc_wrapper()->pc()->CreateSender("audio", "callee_stream"); | |
2668 auto callee_video_sender = | |
2669 callee_pc_wrapper()->pc()->CreateSender("video", "callee_stream"); | |
2670 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2671 ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); | |
2672 // Wait for ICE to complete, without any tracks being set. | |
2673 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted, | |
2674 caller_pc_wrapper()->ice_connection_state(), | |
2675 kMaxWaitForFramesMs); | |
2676 EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected, | |
2677 callee_pc_wrapper()->ice_connection_state(), | |
2678 kMaxWaitForFramesMs); | |
2679 // Now set the tracks, and expect frames to immediately start flowing. | |
2680 EXPECT_TRUE(caller_audio_sender->SetTrack( | |
2681 caller_pc_wrapper()->CreateLocalAudioTrack())); | |
2682 EXPECT_TRUE(caller_video_sender->SetTrack( | |
2683 caller_pc_wrapper()->CreateLocalVideoTrack())); | |
2684 EXPECT_TRUE(callee_audio_sender->SetTrack( | |
2685 callee_pc_wrapper()->CreateLocalAudioTrack())); | |
2686 EXPECT_TRUE(callee_video_sender->SetTrack( | |
2687 callee_pc_wrapper()->CreateLocalVideoTrack())); | |
2688 EXPECT_TRUE_WAIT( | |
2689 caller_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2690 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2691 kEndVideoFrameCount) && | |
2692 callee_pc_wrapper()->ReceivedAudioFrames(kEndAudioFrameCount) && | |
2693 callee_pc_wrapper()->ReceivedVideoFramesForEachTrack( | |
2694 kEndVideoFrameCount), | |
2695 kMaxWaitForFramesMs); | |
2696 } | |
2697 | |
2698 // This test verifies that a remote video track can be added via AddStream, | |
2699 // and sent end-to-end. For this particular test, it's simply echoed back | |
2700 // from the caller to the callee, rather than being forwarded to a third | |
2701 // PeerConnection. | |
2702 TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) { | |
2703 ASSERT_TRUE(CreatePeerConnectionWrappers()); | |
2704 ConnectFakeSignaling(); | |
2705 // Just send a video track from the caller. | |
2706 caller_pc_wrapper()->AddMediaStreamFromTracks( | |
2707 nullptr, caller_pc_wrapper()->CreateLocalVideoTrack()); | |
2708 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2709 ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); | |
2710 ASSERT_EQ(1, callee_pc_wrapper()->remote_streams()->count()); | |
2711 | |
2712 // Echo the stream back, and do a new offer/anwer (initiated by callee this | |
2713 // time). | |
2714 callee_pc_wrapper()->pc()->AddStream( | |
2715 callee_pc_wrapper()->remote_streams()->at(0)); | |
2716 callee_pc_wrapper()->CreateSetAndSignalOffer(); | |
2717 ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); | |
2718 | |
2719 EXPECT_TRUE_WAIT( | |
2720 caller_pc_wrapper()->ReceivedVideoFramesForEachTrack(kEndVideoFrameCount), | |
2721 kMaxWaitForFramesMs); | |
2722 } | |
2723 | |
2724 // Test that we achieve the expected end-to-end connection time, using a | |
2725 // fake clock and simulated latency on the media and signaling paths. | |
2726 // We use a TURN<->TURN connection because this is usually the quickest to | |
2727 // set up initially, especially when we're confident the connection will work | |
2728 // and can start sending media before we get a STUN response. | |
2729 // | |
2730 // With various optimizations enabled, here are the network delays we expect to | |
2731 // be on the critical path: | |
2732 // 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then | |
2733 // signaling answer (with DTLS fingerprint). | |
2734 // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when | |
2735 // using TURN<->TURN pair, and DTLS exchange is 4 packets, | |
2736 // the first of which should have arrived before the answer. | |
pthatcher1
2017/03/20 18:23:17
Wow. That's thorough.
| |
2737 TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) { | |
2738 rtc::ScopedFakeClock fake_clock; | |
2739 // Some things use a time of "0" as a special value, so we need to start out | |
2740 // the fake clock at a nonzero time. | |
2741 // TODO(deadbeef): Fix this. | |
2742 fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1)); | |
2743 | |
2744 static constexpr int media_hop_delay_ms = 50; | |
2745 static constexpr int signaling_trip_delay_ms = 500; | |
2746 // For explanation of these values, see comment above. | |
2747 static constexpr int required_media_hops = 9; | |
2748 static constexpr int required_signaling_trips = 2; | |
2749 // For internal delays (such as posting an event asychronously). | |
2750 static constexpr int allowed_internal_delay_ms = 20; | |
2751 static constexpr int total_connection_time_ms = | |
2752 media_hop_delay_ms * required_media_hops + | |
2753 signaling_trip_delay_ms * required_signaling_trips + | |
2754 allowed_internal_delay_ms; | |
2755 | |
2756 static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0", | |
2757 3478}; | |
2758 static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1", | |
2759 0}; | |
2760 static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0", | |
2761 3478}; | |
2762 static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1", | |
2763 0}; | |
2764 cricket::TestTurnServer turn_server_1(network_thread(), | |
2765 turn_server_1_internal_address, | |
2766 turn_server_1_external_address); | |
2767 cricket::TestTurnServer turn_server_2(network_thread(), | |
2768 turn_server_2_internal_address, | |
2769 turn_server_2_external_address); | |
2770 // Bypass permission check on received packets so media can be sent before | |
2771 // the candidate is signaled. | |
2772 turn_server_1.set_enable_permission_checks(false); | |
2773 turn_server_2.set_enable_permission_checks(false); | |
2774 | |
2775 PeerConnectionInterface::RTCConfiguration client_1_config; | |
2776 webrtc::PeerConnectionInterface::IceServer ice_server_1; | |
2777 ice_server_1.urls.push_back("turn:88.88.88.0:3478"); | |
2778 ice_server_1.username = "test"; | |
2779 ice_server_1.password = "test"; | |
2780 client_1_config.servers.push_back(ice_server_1); | |
2781 client_1_config.type = webrtc::PeerConnectionInterface::kRelay; | |
2782 client_1_config.presume_writable_when_fully_relayed = true; | |
2783 | |
2784 PeerConnectionInterface::RTCConfiguration client_2_config; | |
2785 webrtc::PeerConnectionInterface::IceServer ice_server_2; | |
2786 ice_server_2.urls.push_back("turn:99.99.99.0:3478"); | |
2787 ice_server_2.username = "test"; | |
2788 ice_server_2.password = "test"; | |
2789 client_2_config.servers.push_back(ice_server_2); | |
2790 client_2_config.type = webrtc::PeerConnectionInterface::kRelay; | |
2791 client_2_config.presume_writable_when_fully_relayed = true; | |
2792 | |
2793 ASSERT_TRUE( | |
2794 CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config)); | |
2795 // Set up the simulated delays. | |
2796 SetSignalingDelayMs(signaling_trip_delay_ms); | |
2797 ConnectFakeSignaling(); | |
2798 virtual_socket_server()->set_delay_mean(media_hop_delay_ms); | |
2799 virtual_socket_server()->UpdateDelayDistribution(); | |
2800 | |
2801 // Set "offer to receive audio/video" without adding any tracks, so we just | |
2802 // set up ICE/DTLS with no media. | |
2803 PeerConnectionInterface::RTCOfferAnswerOptions options; | |
2804 options.offer_to_receive_audio = 1; | |
2805 options.offer_to_receive_video = 1; | |
2806 caller_pc_wrapper()->SetOfferAnswerOptions(options); | |
2807 caller_pc_wrapper()->CreateSetAndSignalOffer(); | |
2808 // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS | |
2809 // are connected. This is an important distinction. Once we have separate ICE | |
2810 // and DTLS state, this check needs to use the DTLS state. | |
2811 EXPECT_TRUE_SIMULATED_WAIT( | |
2812 (callee_pc_wrapper()->ice_connection_state() == | |
2813 webrtc::PeerConnectionInterface::kIceConnectionConnected || | |
2814 callee_pc_wrapper()->ice_connection_state() == | |
2815 webrtc::PeerConnectionInterface::kIceConnectionCompleted) && | |
2816 (caller_pc_wrapper()->ice_connection_state() == | |
2817 webrtc::PeerConnectionInterface::kIceConnectionConnected || | |
2818 caller_pc_wrapper()->ice_connection_state() == | |
2819 webrtc::PeerConnectionInterface::kIceConnectionCompleted), | |
2820 total_connection_time_ms, fake_clock); | |
2821 // Need to free the clients here since they're using things we created on | |
2822 // the stack. | |
2823 delete SetCallerPcWrapperAndReturnCurrent(nullptr); | |
2824 delete SetCalleePcWrapperAndReturnCurrent(nullptr); | |
2825 } | |
2826 | |
2827 #endif // if !defined(THREAD_SANITIZER) | |
2828 | |
2829 } // namespace | |
OLD | NEW |