cc: Make picture pile base thread safe.
[chromium-blink-merge.git] / content / renderer / media / rtc_peer_connection_handler_unittest.cc
blobaac53423f2f6f18f49e56f429d9fbf7ca89daf41
1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #include <string>
6 #include <vector>
8 #include "base/memory/scoped_ptr.h"
9 #include "base/message_loop/message_loop.h"
10 #include "base/strings/utf_string_conversions.h"
11 #include "base/values.h"
12 #include "content/child/child_process.h"
13 #include "content/renderer/media/media_stream.h"
14 #include "content/renderer/media/media_stream_audio_source.h"
15 #include "content/renderer/media/media_stream_source.h"
16 #include "content/renderer/media/media_stream_video_track.h"
17 #include "content/renderer/media/mock_media_stream_video_source.h"
18 #include "content/renderer/media/mock_peer_connection_impl.h"
19 #include "content/renderer/media/mock_web_rtc_peer_connection_handler_client.h"
20 #include "content/renderer/media/peer_connection_tracker.h"
21 #include "content/renderer/media/rtc_media_constraints.h"
22 #include "content/renderer/media/rtc_peer_connection_handler.h"
23 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
24 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
25 #include "content/renderer/media/webrtc_audio_capturer.h"
26 #include "testing/gmock/include/gmock/gmock.h"
27 #include "testing/gtest/include/gtest/gtest.h"
28 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
29 #include "third_party/WebKit/public/platform/WebMediaStream.h"
30 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
31 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
32 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
33 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h"
34 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h"
35 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
36 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
37 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h"
38 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
39 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
40 #include "third_party/WebKit/public/platform/WebRTCStatsRequest.h"
41 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
42 #include "third_party/WebKit/public/platform/WebURL.h"
43 #include "third_party/WebKit/public/web/WebHeap.h"
44 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
46 static const char kDummySdp[] = "dummy sdp";
47 static const char kDummySdpType[] = "dummy type";
49 using blink::WebRTCPeerConnectionHandlerClient;
50 using testing::NiceMock;
51 using testing::_;
52 using testing::Ref;
54 namespace content {
56 class MockRTCStatsResponse : public LocalRTCStatsResponse {
57 public:
58 MockRTCStatsResponse()
59 : report_count_(0),
60 statistic_count_(0) {
63 size_t addReport(blink::WebString type,
64 blink::WebString id,
65 double timestamp) override {
66 ++report_count_;
67 return report_count_;
70 void addStatistic(size_t report,
71 blink::WebString name,
72 blink::WebString value) override {
73 ++statistic_count_;
75 int report_count() const { return report_count_; }
77 private:
78 int report_count_;
79 int statistic_count_;
82 // Mocked wrapper for blink::WebRTCStatsRequest
83 class MockRTCStatsRequest : public LocalRTCStatsRequest {
84 public:
85 MockRTCStatsRequest()
86 : has_selector_(false),
87 request_succeeded_called_(false) {}
89 bool hasSelector() const override { return has_selector_; }
90 blink::WebMediaStreamTrack component() const override { return component_; }
91 scoped_refptr<LocalRTCStatsResponse> createResponse() override {
92 DCHECK(!response_.get());
93 response_ = new rtc::RefCountedObject<MockRTCStatsResponse>();
94 return response_;
97 void requestSucceeded(const LocalRTCStatsResponse* response) override {
98 EXPECT_EQ(response, response_.get());
99 request_succeeded_called_ = true;
102 // Function for setting whether or not a selector is available.
103 void setSelector(const blink::WebMediaStreamTrack& component) {
104 has_selector_ = true;
105 component_ = component;
108 // Function for inspecting the result of a stats request.
109 MockRTCStatsResponse* result() {
110 if (request_succeeded_called_) {
111 return response_.get();
112 } else {
113 return NULL;
117 private:
118 bool has_selector_;
119 blink::WebMediaStreamTrack component_;
120 scoped_refptr<MockRTCStatsResponse> response_;
121 bool request_succeeded_called_;
124 class MockPeerConnectionTracker : public PeerConnectionTracker {
125 public:
126 MOCK_METHOD1(UnregisterPeerConnection,
127 void(RTCPeerConnectionHandler* pc_handler));
128 // TODO(jiayl): add coverage for the following methods
129 MOCK_METHOD2(TrackCreateOffer,
130 void(RTCPeerConnectionHandler* pc_handler,
131 const RTCMediaConstraints& constraints));
132 MOCK_METHOD2(TrackCreateAnswer,
133 void(RTCPeerConnectionHandler* pc_handler,
134 const RTCMediaConstraints& constraints));
135 MOCK_METHOD3(TrackSetSessionDescription,
136 void(RTCPeerConnectionHandler* pc_handler,
137 const blink::WebRTCSessionDescription& desc,
138 Source source));
139 MOCK_METHOD3(
140 TrackUpdateIce,
141 void(RTCPeerConnectionHandler* pc_handler,
142 const webrtc::PeerConnectionInterface::RTCConfiguration& config,
143 const RTCMediaConstraints& options));
144 MOCK_METHOD4(TrackAddIceCandidate,
145 void(RTCPeerConnectionHandler* pc_handler,
146 const blink::WebRTCICECandidate& candidate,
147 Source source,
148 bool succeeded));
149 MOCK_METHOD3(TrackAddStream,
150 void(RTCPeerConnectionHandler* pc_handler,
151 const blink::WebMediaStream& stream,
152 Source source));
153 MOCK_METHOD3(TrackRemoveStream,
154 void(RTCPeerConnectionHandler* pc_handler,
155 const blink::WebMediaStream& stream,
156 Source source));
157 MOCK_METHOD1(TrackOnIceComplete,
158 void(RTCPeerConnectionHandler* pc_handler));
159 MOCK_METHOD3(TrackCreateDataChannel,
160 void(RTCPeerConnectionHandler* pc_handler,
161 const webrtc::DataChannelInterface* data_channel,
162 Source source));
163 MOCK_METHOD1(TrackStop, void(RTCPeerConnectionHandler* pc_handler));
164 MOCK_METHOD2(TrackSignalingStateChange,
165 void(RTCPeerConnectionHandler* pc_handler,
166 WebRTCPeerConnectionHandlerClient::SignalingState state));
167 MOCK_METHOD2(
168 TrackIceConnectionStateChange,
169 void(RTCPeerConnectionHandler* pc_handler,
170 WebRTCPeerConnectionHandlerClient::ICEConnectionState state));
171 MOCK_METHOD2(
172 TrackIceGatheringStateChange,
173 void(RTCPeerConnectionHandler* pc_handler,
174 WebRTCPeerConnectionHandlerClient::ICEGatheringState state));
175 MOCK_METHOD1(TrackOnRenegotiationNeeded,
176 void(RTCPeerConnectionHandler* pc_handler));
177 MOCK_METHOD2(TrackCreateDTMFSender,
178 void(RTCPeerConnectionHandler* pc_handler,
179 const blink::WebMediaStreamTrack& track));
182 class RTCPeerConnectionHandlerUnderTest : public RTCPeerConnectionHandler {
183 public:
184 RTCPeerConnectionHandlerUnderTest(
185 WebRTCPeerConnectionHandlerClient* client,
186 PeerConnectionDependencyFactory* dependency_factory)
187 : RTCPeerConnectionHandler(client, dependency_factory) {
190 MockPeerConnectionImpl* native_peer_connection() {
191 return static_cast<MockPeerConnectionImpl*>(
192 RTCPeerConnectionHandler::native_peer_connection());
196 class RTCPeerConnectionHandlerTest : public ::testing::Test {
197 public:
198 RTCPeerConnectionHandlerTest() : mock_peer_connection_(NULL) {
199 child_process_.reset(new ChildProcess());
202 virtual void SetUp() {
203 mock_client_.reset(new NiceMock<MockWebRTCPeerConnectionHandlerClient>());
204 mock_dependency_factory_.reset(new MockPeerConnectionDependencyFactory());
205 pc_handler_.reset(
206 new RTCPeerConnectionHandlerUnderTest(mock_client_.get(),
207 mock_dependency_factory_.get()));
208 mock_tracker_.reset(new NiceMock<MockPeerConnectionTracker>());
209 blink::WebRTCConfiguration config;
210 blink::WebMediaConstraints constraints;
211 EXPECT_TRUE(pc_handler_->InitializeForTest(config, constraints,
212 mock_tracker_.get()));
214 mock_peer_connection_ = pc_handler_->native_peer_connection();
215 ASSERT_TRUE(mock_peer_connection_);
218 virtual void TearDown() {
219 pc_handler_.reset();
220 mock_tracker_.reset();
221 mock_dependency_factory_.reset();
222 mock_client_.reset();
223 blink::WebHeap::collectAllGarbageForTesting();
226 // Creates a WebKit local MediaStream.
227 blink::WebMediaStream CreateLocalMediaStream(
228 const std::string& stream_label) {
229 std::string video_track_label("video-label");
230 std::string audio_track_label("audio-label");
232 blink::WebMediaStreamSource audio_source;
233 audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
234 blink::WebMediaStreamSource::TypeAudio,
235 blink::WebString::fromUTF8("audio_track"));
236 audio_source.setExtraData(new MediaStreamAudioSource());
237 blink::WebMediaStreamSource video_source;
238 video_source.initialize(blink::WebString::fromUTF8(video_track_label),
239 blink::WebMediaStreamSource::TypeVideo,
240 blink::WebString::fromUTF8("video_track"));
241 MockMediaStreamVideoSource* native_video_source =
242 new MockMediaStreamVideoSource(false);
243 video_source.setExtraData(native_video_source);
245 blink::WebVector<blink::WebMediaStreamTrack> audio_tracks(
246 static_cast<size_t>(1));
247 audio_tracks[0].initialize(audio_source.id(), audio_source);
248 audio_tracks[0].setExtraData(
249 new MediaStreamTrack(
250 WebRtcLocalAudioTrackAdapter::Create(audio_track_label,
251 NULL),
252 true));
253 blink::WebVector<blink::WebMediaStreamTrack> video_tracks(
254 static_cast<size_t>(1));
255 blink::WebMediaConstraints constraints;
256 constraints.initialize();
257 video_tracks[0] = MediaStreamVideoTrack::CreateVideoTrack(
258 native_video_source, constraints,
259 MediaStreamVideoSource::ConstraintsCallback(), true);
261 blink::WebMediaStream local_stream;
262 local_stream.initialize(base::UTF8ToUTF16(stream_label), audio_tracks,
263 video_tracks);
264 local_stream.setExtraData(
265 new MediaStream(local_stream));
266 return local_stream;
269 // Creates a remote MediaStream and adds it to the mocked native
270 // peer connection.
271 scoped_refptr<webrtc::MediaStreamInterface>
272 AddRemoteMockMediaStream(const std::string& stream_label,
273 const std::string& video_track_label,
274 const std::string& audio_track_label) {
275 scoped_refptr<webrtc::MediaStreamInterface> stream(
276 mock_dependency_factory_->CreateLocalMediaStream(stream_label));
277 if (!video_track_label.empty()) {
278 webrtc::VideoSourceInterface* source = NULL;
279 scoped_refptr<webrtc::VideoTrackInterface> video_track(
280 mock_dependency_factory_->CreateLocalVideoTrack(
281 video_track_label, source));
282 stream->AddTrack(video_track.get());
284 if (!audio_track_label.empty()) {
285 scoped_refptr<WebRtcAudioCapturer> capturer;
286 scoped_refptr<webrtc::AudioTrackInterface> audio_track(
287 WebRtcLocalAudioTrackAdapter::Create(audio_track_label, NULL));
288 stream->AddTrack(audio_track.get());
290 mock_peer_connection_->AddRemoteStream(stream.get());
291 return stream;
294 base::MessageLoop message_loop_;
295 scoped_ptr<ChildProcess> child_process_;
296 scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_;
297 scoped_ptr<MockPeerConnectionDependencyFactory> mock_dependency_factory_;
298 scoped_ptr<NiceMock<MockPeerConnectionTracker> > mock_tracker_;
299 scoped_ptr<RTCPeerConnectionHandlerUnderTest> pc_handler_;
301 // Weak reference to the mocked native peer connection implementation.
302 MockPeerConnectionImpl* mock_peer_connection_;
305 TEST_F(RTCPeerConnectionHandlerTest, Destruct) {
306 EXPECT_CALL(*mock_tracker_.get(), UnregisterPeerConnection(pc_handler_.get()))
307 .Times(1);
308 pc_handler_.reset(NULL);
311 TEST_F(RTCPeerConnectionHandlerTest, CreateOffer) {
312 blink::WebRTCSessionDescriptionRequest request;
313 blink::WebMediaConstraints options;
314 EXPECT_CALL(*mock_tracker_.get(), TrackCreateOffer(pc_handler_.get(), _));
316 // TODO(perkj): Can blink::WebRTCSessionDescriptionRequest be changed so
317 // the |reqest| requestSucceeded can be tested? Currently the |request| object
318 // can not be initialized from a unit test.
319 EXPECT_FALSE(mock_peer_connection_->created_session_description() != NULL);
320 pc_handler_->createOffer(request, options);
321 EXPECT_TRUE(mock_peer_connection_->created_session_description() != NULL);
324 TEST_F(RTCPeerConnectionHandlerTest, CreateAnswer) {
325 blink::WebRTCSessionDescriptionRequest request;
326 blink::WebMediaConstraints options;
327 EXPECT_CALL(*mock_tracker_.get(), TrackCreateAnswer(pc_handler_.get(), _));
328 // TODO(perkj): Can blink::WebRTCSessionDescriptionRequest be changed so
329 // the |reqest| requestSucceeded can be tested? Currently the |request| object
330 // can not be initialized from a unit test.
331 EXPECT_FALSE(mock_peer_connection_->created_session_description() != NULL);
332 pc_handler_->createAnswer(request, options);
333 EXPECT_TRUE(mock_peer_connection_->created_session_description() != NULL);
336 TEST_F(RTCPeerConnectionHandlerTest, setLocalDescription) {
337 blink::WebRTCVoidRequest request;
338 blink::WebRTCSessionDescription description;
339 description.initialize(kDummySdpType, kDummySdp);
340 // PeerConnectionTracker::TrackSetSessionDescription is expected to be called
341 // before |mock_peer_connection| is called.
342 testing::InSequence sequence;
343 EXPECT_CALL(*mock_tracker_.get(),
344 TrackSetSessionDescription(pc_handler_.get(), Ref(description),
345 PeerConnectionTracker::SOURCE_LOCAL));
346 EXPECT_CALL(*mock_peer_connection_, SetLocalDescription(_, _));
348 pc_handler_->setLocalDescription(request, description);
349 EXPECT_EQ(description.type(), pc_handler_->localDescription().type());
350 EXPECT_EQ(description.sdp(), pc_handler_->localDescription().sdp());
352 std::string sdp_string;
353 ASSERT_TRUE(mock_peer_connection_->local_description() != NULL);
354 EXPECT_EQ(kDummySdpType, mock_peer_connection_->local_description()->type());
355 mock_peer_connection_->local_description()->ToString(&sdp_string);
356 EXPECT_EQ(kDummySdp, sdp_string);
359 TEST_F(RTCPeerConnectionHandlerTest, setRemoteDescription) {
360 blink::WebRTCVoidRequest request;
361 blink::WebRTCSessionDescription description;
362 description.initialize(kDummySdpType, kDummySdp);
364 // PeerConnectionTracker::TrackSetSessionDescription is expected to be called
365 // before |mock_peer_connection| is called.
366 testing::InSequence sequence;
367 EXPECT_CALL(*mock_tracker_.get(),
368 TrackSetSessionDescription(pc_handler_.get(), Ref(description),
369 PeerConnectionTracker::SOURCE_REMOTE));
370 EXPECT_CALL(*mock_peer_connection_, SetRemoteDescription(_, _));
372 pc_handler_->setRemoteDescription(request, description);
373 EXPECT_EQ(description.type(), pc_handler_->remoteDescription().type());
374 EXPECT_EQ(description.sdp(), pc_handler_->remoteDescription().sdp());
376 std::string sdp_string;
377 ASSERT_TRUE(mock_peer_connection_->remote_description() != NULL);
378 EXPECT_EQ(kDummySdpType, mock_peer_connection_->remote_description()->type());
379 mock_peer_connection_->remote_description()->ToString(&sdp_string);
380 EXPECT_EQ(kDummySdp, sdp_string);
383 TEST_F(RTCPeerConnectionHandlerTest, updateICE) {
384 blink::WebRTCConfiguration config;
385 blink::WebMediaConstraints constraints;
387 EXPECT_CALL(*mock_tracker_.get(), TrackUpdateIce(pc_handler_.get(), _, _));
388 // TODO(perkj): Test that the parameters in |config| can be translated when a
389 // WebRTCConfiguration can be constructed. It's WebKit class and can't be
390 // initialized from a test.
391 EXPECT_TRUE(pc_handler_->updateICE(config, constraints));
394 TEST_F(RTCPeerConnectionHandlerTest, addICECandidate) {
395 blink::WebRTCICECandidate candidate;
396 candidate.initialize(kDummySdp, "sdpMid", 1);
398 EXPECT_CALL(*mock_tracker_.get(),
399 TrackAddIceCandidate(pc_handler_.get(),
400 testing::Ref(candidate),
401 PeerConnectionTracker::SOURCE_REMOTE,
402 true));
403 EXPECT_TRUE(pc_handler_->addICECandidate(candidate));
404 EXPECT_EQ(kDummySdp, mock_peer_connection_->ice_sdp());
405 EXPECT_EQ(1, mock_peer_connection_->sdp_mline_index());
406 EXPECT_EQ("sdpMid", mock_peer_connection_->sdp_mid());
409 TEST_F(RTCPeerConnectionHandlerTest, addAndRemoveStream) {
410 std::string stream_label = "local_stream";
411 blink::WebMediaStream local_stream(
412 CreateLocalMediaStream(stream_label));
413 blink::WebMediaConstraints constraints;
415 EXPECT_CALL(*mock_tracker_.get(),
416 TrackAddStream(pc_handler_.get(),
417 testing::Ref(local_stream),
418 PeerConnectionTracker::SOURCE_LOCAL));
419 EXPECT_CALL(*mock_tracker_.get(),
420 TrackRemoveStream(pc_handler_.get(),
421 testing::Ref(local_stream),
422 PeerConnectionTracker::SOURCE_LOCAL));
423 EXPECT_TRUE(pc_handler_->addStream(local_stream, constraints));
424 EXPECT_EQ(stream_label, mock_peer_connection_->stream_label());
425 EXPECT_EQ(1u,
426 mock_peer_connection_->local_streams()->at(0)->GetAudioTracks().size());
427 EXPECT_EQ(1u,
428 mock_peer_connection_->local_streams()->at(0)->GetVideoTracks().size());
430 EXPECT_FALSE(pc_handler_->addStream(local_stream, constraints));
432 pc_handler_->removeStream(local_stream);
433 EXPECT_EQ(0u, mock_peer_connection_->local_streams()->count());
436 TEST_F(RTCPeerConnectionHandlerTest, addStreamWithStoppedAudioAndVideoTrack) {
437 std::string stream_label = "local_stream";
438 blink::WebMediaStream local_stream(
439 CreateLocalMediaStream(stream_label));
440 blink::WebMediaConstraints constraints;
442 blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
443 local_stream.audioTracks(audio_tracks);
444 MediaStreamAudioSource* native_audio_source =
445 static_cast<MediaStreamAudioSource*>(
446 audio_tracks[0].source().extraData());
447 native_audio_source->StopSource();
449 blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
450 local_stream.videoTracks(video_tracks);
451 MediaStreamVideoSource* native_video_source =
452 static_cast<MediaStreamVideoSource*>(
453 video_tracks[0].source().extraData());
454 native_video_source->StopSource();
456 EXPECT_TRUE(pc_handler_->addStream(local_stream, constraints));
457 EXPECT_EQ(stream_label, mock_peer_connection_->stream_label());
458 EXPECT_EQ(
460 mock_peer_connection_->local_streams()->at(0)->GetAudioTracks().size());
461 EXPECT_EQ(
463 mock_peer_connection_->local_streams()->at(0)->GetVideoTracks().size());
466 TEST_F(RTCPeerConnectionHandlerTest, GetStatsNoSelector) {
467 scoped_refptr<MockRTCStatsRequest> request(
468 new rtc::RefCountedObject<MockRTCStatsRequest>());
469 pc_handler_->getStats(request.get());
470 // Note that callback gets executed synchronously by mock.
471 ASSERT_TRUE(request->result());
472 EXPECT_LT(1, request->result()->report_count());
475 TEST_F(RTCPeerConnectionHandlerTest, GetStatsAfterClose) {
476 scoped_refptr<MockRTCStatsRequest> request(
477 new rtc::RefCountedObject<MockRTCStatsRequest>());
478 pc_handler_->stop();
479 pc_handler_->getStats(request.get());
480 // Note that callback gets executed synchronously by mock.
481 ASSERT_TRUE(request->result());
482 EXPECT_LT(1, request->result()->report_count());
485 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithLocalSelector) {
486 blink::WebMediaStream local_stream(
487 CreateLocalMediaStream("local_stream"));
488 blink::WebMediaConstraints constraints;
489 pc_handler_->addStream(local_stream, constraints);
490 blink::WebVector<blink::WebMediaStreamTrack> tracks;
491 local_stream.audioTracks(tracks);
492 ASSERT_LE(1ul, tracks.size());
494 scoped_refptr<MockRTCStatsRequest> request(
495 new rtc::RefCountedObject<MockRTCStatsRequest>());
496 request->setSelector(tracks[0]);
497 pc_handler_->getStats(request.get());
498 EXPECT_EQ(1, request->result()->report_count());
501 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithRemoteSelector) {
502 scoped_refptr<webrtc::MediaStreamInterface> stream(
503 AddRemoteMockMediaStream("remote_stream", "video", "audio"));
504 pc_handler_->OnAddStream(stream.get());
505 const blink::WebMediaStream& remote_stream = mock_client_->remote_stream();
507 blink::WebVector<blink::WebMediaStreamTrack> tracks;
508 remote_stream.audioTracks(tracks);
509 ASSERT_LE(1ul, tracks.size());
511 scoped_refptr<MockRTCStatsRequest> request(
512 new rtc::RefCountedObject<MockRTCStatsRequest>());
513 request->setSelector(tracks[0]);
514 pc_handler_->getStats(request.get());
515 EXPECT_EQ(1, request->result()->report_count());
518 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithBadSelector) {
519 // The setup is the same as GetStatsWithLocalSelector, but the stream is not
520 // added to the PeerConnection.
521 blink::WebMediaStream local_stream(
522 CreateLocalMediaStream("local_stream_2"));
523 blink::WebMediaConstraints constraints;
524 blink::WebVector<blink::WebMediaStreamTrack> tracks;
526 local_stream.audioTracks(tracks);
527 blink::WebMediaStreamTrack component = tracks[0];
528 mock_peer_connection_->SetGetStatsResult(false);
530 scoped_refptr<MockRTCStatsRequest> request(
531 new rtc::RefCountedObject<MockRTCStatsRequest>());
532 request->setSelector(component);
533 pc_handler_->getStats(request.get());
534 EXPECT_EQ(0, request->result()->report_count());
537 TEST_F(RTCPeerConnectionHandlerTest, OnSignalingChange) {
538 testing::InSequence sequence;
540 webrtc::PeerConnectionInterface::SignalingState new_state =
541 webrtc::PeerConnectionInterface::kHaveRemoteOffer;
542 EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
543 pc_handler_.get(),
544 WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemoteOffer));
545 EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
546 WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemoteOffer));
547 pc_handler_->OnSignalingChange(new_state);
549 new_state = webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
550 EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
551 pc_handler_.get(),
552 WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalPrAnswer));
553 EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
554 WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalPrAnswer));
555 pc_handler_->OnSignalingChange(new_state);
557 new_state = webrtc::PeerConnectionInterface::kHaveLocalOffer;
558 EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
559 pc_handler_.get(),
560 WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalOffer));
561 EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
562 WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalOffer));
563 pc_handler_->OnSignalingChange(new_state);
565 new_state = webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
566 EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
567 pc_handler_.get(),
568 WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemotePrAnswer));
569 EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
570 WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemotePrAnswer));
571 pc_handler_->OnSignalingChange(new_state);
573 new_state = webrtc::PeerConnectionInterface::kClosed;
574 EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
575 pc_handler_.get(),
576 WebRTCPeerConnectionHandlerClient::SignalingStateClosed));
577 EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
578 WebRTCPeerConnectionHandlerClient::SignalingStateClosed));
579 pc_handler_->OnSignalingChange(new_state);
582 TEST_F(RTCPeerConnectionHandlerTest, OnIceConnectionChange) {
583 testing::InSequence sequence;
585 webrtc::PeerConnectionInterface::IceConnectionState new_state =
586 webrtc::PeerConnectionInterface::kIceConnectionNew;
587 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
588 pc_handler_.get(),
589 WebRTCPeerConnectionHandlerClient::ICEConnectionStateStarting));
590 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
591 WebRTCPeerConnectionHandlerClient::ICEConnectionStateStarting));
592 pc_handler_->OnIceConnectionChange(new_state);
594 new_state = webrtc::PeerConnectionInterface::kIceConnectionChecking;
595 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
596 pc_handler_.get(),
597 WebRTCPeerConnectionHandlerClient::ICEConnectionStateChecking));
598 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
599 WebRTCPeerConnectionHandlerClient::ICEConnectionStateChecking));
600 pc_handler_->OnIceConnectionChange(new_state);
602 new_state = webrtc::PeerConnectionInterface::kIceConnectionConnected;
603 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
604 pc_handler_.get(),
605 WebRTCPeerConnectionHandlerClient::ICEConnectionStateConnected));
606 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
607 WebRTCPeerConnectionHandlerClient::ICEConnectionStateConnected));
608 pc_handler_->OnIceConnectionChange(new_state);
610 new_state = webrtc::PeerConnectionInterface::kIceConnectionCompleted;
611 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
612 pc_handler_.get(),
613 WebRTCPeerConnectionHandlerClient::ICEConnectionStateCompleted));
614 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
615 WebRTCPeerConnectionHandlerClient::ICEConnectionStateCompleted));
616 pc_handler_->OnIceConnectionChange(new_state);
618 new_state = webrtc::PeerConnectionInterface::kIceConnectionFailed;
619 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
620 pc_handler_.get(),
621 WebRTCPeerConnectionHandlerClient::ICEConnectionStateFailed));
622 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
623 WebRTCPeerConnectionHandlerClient::ICEConnectionStateFailed));
624 pc_handler_->OnIceConnectionChange(new_state);
626 new_state = webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
627 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
628 pc_handler_.get(),
629 WebRTCPeerConnectionHandlerClient::ICEConnectionStateDisconnected));
630 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
631 WebRTCPeerConnectionHandlerClient::ICEConnectionStateDisconnected));
632 pc_handler_->OnIceConnectionChange(new_state);
634 new_state = webrtc::PeerConnectionInterface::kIceConnectionClosed;
635 EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
636 pc_handler_.get(),
637 WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed));
638 EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
639 WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed));
640 pc_handler_->OnIceConnectionChange(new_state);
643 TEST_F(RTCPeerConnectionHandlerTest, OnIceGatheringChange) {
644 testing::InSequence sequence;
645 EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
646 pc_handler_.get(),
647 WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew));
648 EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
649 WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew));
650 EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
651 pc_handler_.get(),
652 WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering));
653 EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
654 WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering));
655 EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
656 pc_handler_.get(),
657 WebRTCPeerConnectionHandlerClient::ICEGatheringStateComplete));
658 EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
659 WebRTCPeerConnectionHandlerClient::ICEGatheringStateComplete));
661 webrtc::PeerConnectionInterface::IceGatheringState new_state =
662 webrtc::PeerConnectionInterface::kIceGatheringNew;
663 pc_handler_->OnIceGatheringChange(new_state);
665 new_state = webrtc::PeerConnectionInterface::kIceGatheringGathering;
666 pc_handler_->OnIceGatheringChange(new_state);
668 new_state = webrtc::PeerConnectionInterface::kIceGatheringComplete;
669 pc_handler_->OnIceGatheringChange(new_state);
671 // Check NULL candidate after ice gathering is completed.
672 EXPECT_EQ("", mock_client_->candidate_mid());
673 EXPECT_EQ(-1, mock_client_->candidate_mlineindex());
674 EXPECT_EQ("", mock_client_->candidate_sdp());
677 TEST_F(RTCPeerConnectionHandlerTest, OnAddAndOnRemoveStream) {
678 std::string remote_stream_label("remote_stream");
679 scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
680 AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
682 testing::InSequence sequence;
683 EXPECT_CALL(*mock_tracker_.get(), TrackAddStream(
684 pc_handler_.get(),
685 testing::Property(&blink::WebMediaStream::id,
686 base::UTF8ToUTF16(remote_stream_label)),
687 PeerConnectionTracker::SOURCE_REMOTE));
688 EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
689 testing::Property(&blink::WebMediaStream::id,
690 base::UTF8ToUTF16(remote_stream_label))));
692 EXPECT_CALL(*mock_tracker_.get(), TrackRemoveStream(
693 pc_handler_.get(),
694 testing::Property(&blink::WebMediaStream::id,
695 base::UTF8ToUTF16(remote_stream_label)),
696 PeerConnectionTracker::SOURCE_REMOTE));
697 EXPECT_CALL(*mock_client_.get(), didRemoveRemoteStream(
698 testing::Property(&blink::WebMediaStream::id,
699 base::UTF8ToUTF16(remote_stream_label))));
701 pc_handler_->OnAddStream(remote_stream.get());
702 pc_handler_->OnRemoveStream(remote_stream.get());
705 // This test that WebKit is notified about remote track state changes.
706 TEST_F(RTCPeerConnectionHandlerTest, RemoteTrackState) {
707 std::string remote_stream_label("remote_stream");
708 scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
709 AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
711 testing::InSequence sequence;
712 EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
713 testing::Property(&blink::WebMediaStream::id,
714 base::UTF8ToUTF16(remote_stream_label))));
715 pc_handler_->OnAddStream(remote_stream.get());
716 const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
718 blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
719 webkit_stream.audioTracks(audio_tracks);
720 EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
721 audio_tracks[0].source().readyState());
723 blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
724 webkit_stream.videoTracks(video_tracks);
725 EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
726 video_tracks[0].source().readyState());
728 remote_stream->GetAudioTracks()[0]->set_state(
729 webrtc::MediaStreamTrackInterface::kEnded);
730 EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
731 audio_tracks[0].source().readyState());
733 remote_stream->GetVideoTracks()[0]->set_state(
734 webrtc::MediaStreamTrackInterface::kEnded);
735 EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
736 video_tracks[0].source().readyState());
739 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddAudioTrackFromRemoteStream) {
740 std::string remote_stream_label("remote_stream");
741 scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
742 AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
744 EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
745 testing::Property(&blink::WebMediaStream::id,
746 base::UTF8ToUTF16(remote_stream_label))));
747 pc_handler_->OnAddStream(remote_stream.get());
748 const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
751 // Test in a small scope so that |audio_tracks| don't hold on to destroyed
752 // source later.
753 blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
754 webkit_stream.audioTracks(audio_tracks);
755 EXPECT_EQ(1u, audio_tracks.size());
758 // Remove the Webrtc audio track from the Webrtc MediaStream.
759 scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
760 remote_stream->GetAudioTracks()[0].get();
761 remote_stream->RemoveTrack(webrtc_track.get());
764 blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks1;
765 webkit_stream.audioTracks(modified_audio_tracks1);
766 EXPECT_EQ(0u, modified_audio_tracks1.size());
769 blink::WebHeap::collectGarbageForTesting();
771 // Add the WebRtc audio track again.
772 remote_stream->AddTrack(webrtc_track.get());
773 blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks2;
774 webkit_stream.audioTracks(modified_audio_tracks2);
775 EXPECT_EQ(1u, modified_audio_tracks2.size());
778 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) {
779 std::string remote_stream_label("remote_stream");
780 scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
781 AddRemoteMockMediaStream(remote_stream_label, "video", "video"));
783 EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
784 testing::Property(&blink::WebMediaStream::id,
785 base::UTF8ToUTF16(remote_stream_label))));
786 pc_handler_->OnAddStream(remote_stream.get());
787 const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
790 // Test in a small scope so that |video_tracks| don't hold on to destroyed
791 // source later.
792 blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
793 webkit_stream.videoTracks(video_tracks);
794 EXPECT_EQ(1u, video_tracks.size());
797 // Remove the Webrtc video track from the Webrtc MediaStream.
798 scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
799 remote_stream->GetVideoTracks()[0].get();
800 remote_stream->RemoveTrack(webrtc_track.get());
802 blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks1;
803 webkit_stream.videoTracks(modified_video_tracks1);
804 EXPECT_EQ(0u, modified_video_tracks1.size());
807 blink::WebHeap::collectGarbageForTesting();
809 // Add the WebRtc video track again.
810 remote_stream->AddTrack(webrtc_track.get());
811 blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks2;
812 webkit_stream.videoTracks(modified_video_tracks2);
813 EXPECT_EQ(1u, modified_video_tracks2.size());
816 TEST_F(RTCPeerConnectionHandlerTest, OnIceCandidate) {
817 testing::InSequence sequence;
818 EXPECT_CALL(*mock_tracker_.get(),
819 TrackAddIceCandidate(pc_handler_.get(), _,
820 PeerConnectionTracker::SOURCE_LOCAL, true));
821 EXPECT_CALL(*mock_client_.get(), didGenerateICECandidate(_));
823 scoped_ptr<webrtc::IceCandidateInterface> native_candidate(
824 mock_dependency_factory_->CreateIceCandidate("sdpMid", 1, kDummySdp));
825 pc_handler_->OnIceCandidate(native_candidate.get());
826 EXPECT_EQ("sdpMid", mock_client_->candidate_mid());
827 EXPECT_EQ(1, mock_client_->candidate_mlineindex());
828 EXPECT_EQ(kDummySdp, mock_client_->candidate_sdp());
831 TEST_F(RTCPeerConnectionHandlerTest, OnRenegotiationNeeded) {
832 testing::InSequence sequence;
833 EXPECT_CALL(*mock_tracker_.get(),
834 TrackOnRenegotiationNeeded(pc_handler_.get()));
835 EXPECT_CALL(*mock_client_.get(), negotiationNeeded());
836 pc_handler_->OnRenegotiationNeeded();
839 TEST_F(RTCPeerConnectionHandlerTest, CreateDataChannel) {
840 blink::WebString label = "d1";
841 EXPECT_CALL(*mock_tracker_.get(),
842 TrackCreateDataChannel(pc_handler_.get(),
843 testing::NotNull(),
844 PeerConnectionTracker::SOURCE_LOCAL));
845 scoped_ptr<blink::WebRTCDataChannelHandler> channel(
846 pc_handler_->createDataChannel("d1", blink::WebRTCDataChannelInit()));
847 EXPECT_TRUE(channel.get() != NULL);
848 EXPECT_EQ(label, channel->label());
851 TEST_F(RTCPeerConnectionHandlerTest, CreateDtmfSender) {
852 std::string stream_label = "local_stream";
853 blink::WebMediaStream local_stream(CreateLocalMediaStream(stream_label));
854 blink::WebMediaConstraints constraints;
855 pc_handler_->addStream(local_stream, constraints);
857 blink::WebVector<blink::WebMediaStreamTrack> tracks;
858 local_stream.videoTracks(tracks);
860 ASSERT_LE(1ul, tracks.size());
861 EXPECT_FALSE(pc_handler_->createDTMFSender(tracks[0]));
863 local_stream.audioTracks(tracks);
864 ASSERT_LE(1ul, tracks.size());
866 EXPECT_CALL(*mock_tracker_.get(),
867 TrackCreateDTMFSender(pc_handler_.get(),
868 testing::Ref(tracks[0])));
870 scoped_ptr<blink::WebRTCDTMFSenderHandler> sender(
871 pc_handler_->createDTMFSender(tracks[0]));
872 EXPECT_TRUE(sender.get());
875 } // namespace content