Upstream version 9.38.198.0
[platform/framework/web/crosswalk.git] / src / third_party / libjingle / source / talk / app / webrtc / objctests / RTCPeerConnectionTest.mm
1 /*
2  * libjingle
3  * Copyright 2013, Google Inc.
4  *
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7  *
8  *  1. Redistributions of source code must retain the above copyright notice,
9  *     this list of conditions and the following disclaimer.
10  *  2. Redistributions in binary form must reproduce the above copyright notice,
11  *     this list of conditions and the following disclaimer in the documentation
12  *     and/or other materials provided with the distribution.
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14  *     derived from this software without specific prior written permission.
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19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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24  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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26  */
27
28 #import <Foundation/Foundation.h>
29
30 #import "RTCICEServer.h"
31 #import "RTCMediaConstraints.h"
32 #import "RTCMediaStream.h"
33 #import "RTCPair.h"
34 #import "RTCPeerConnection.h"
35 #import "RTCPeerConnectionFactory.h"
36 #import "RTCPeerConnectionSyncObserver.h"
37 #import "RTCSessionDescription.h"
38 #import "RTCSessionDescriptionSyncObserver.h"
39 #import "RTCVideoRenderer.h"
40 #import "RTCVideoTrack.h"
41
42 #include "webrtc/base/gunit.h"
43 #include "webrtc/base/ssladapter.h"
44
45 #if !defined(__has_feature) || !__has_feature(objc_arc)
46 #error "This file requires ARC support."
47 #endif
48
49 @interface RTCPeerConnectionTest : NSObject
50
51 // Returns whether the two sessions are of the same type.
52 + (BOOL)isSession:(RTCSessionDescription*)session1
53     ofSameTypeAsSession:(RTCSessionDescription*)session2;
54
55 // Create and add tracks to pc, with the given source, label, and IDs
56 - (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
57                                  withFactory:(RTCPeerConnectionFactory*)factory
58                                  videoSource:(RTCVideoSource*)videoSource
59                                  streamLabel:(NSString*)streamLabel
60                                 videoTrackID:(NSString*)videoTrackID
61                                 audioTrackID:(NSString*)audioTrackID;
62
63 - (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory;
64
65 @end
66
67 @implementation RTCPeerConnectionTest
68
69 + (BOOL)isSession:(RTCSessionDescription*)session1
70     ofSameTypeAsSession:(RTCSessionDescription*)session2 {
71   return [session1.type isEqual:session2.type];
72 }
73
74 - (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
75                                  withFactory:(RTCPeerConnectionFactory*)factory
76                                  videoSource:(RTCVideoSource*)videoSource
77                                  streamLabel:(NSString*)streamLabel
78                                 videoTrackID:(NSString*)videoTrackID
79                                 audioTrackID:(NSString*)audioTrackID {
80   RTCMediaStream* localMediaStream = [factory mediaStreamWithLabel:streamLabel];
81   RTCVideoTrack* videoTrack =
82       [factory videoTrackWithID:videoTrackID source:videoSource];
83   RTCVideoRenderer* videoRenderer =
84       [[RTCVideoRenderer alloc] initWithDelegate:nil];
85   [videoTrack addRenderer:videoRenderer];
86   [localMediaStream addVideoTrack:videoTrack];
87   // Test that removal/re-add works.
88   [localMediaStream removeVideoTrack:videoTrack];
89   [localMediaStream addVideoTrack:videoTrack];
90   RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID];
91   [localMediaStream addAudioTrack:audioTrack];
92   RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
93   [pc addStream:localMediaStream constraints:constraints];
94   return localMediaStream;
95 }
96
97 - (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory {
98   NSArray* mandatory = @[
99     [[RTCPair alloc] initWithKey:@"DtlsSrtpKeyAgreement" value:@"true"],
100     [[RTCPair alloc] initWithKey:@"internalSctpDataChannels" value:@"true"],
101   ];
102   RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
103   RTCMediaConstraints* pcConstraints =
104       [[RTCMediaConstraints alloc] initWithMandatoryConstraints:mandatory
105                                             optionalConstraints:nil];
106
107   RTCPeerConnectionSyncObserver* offeringExpectations =
108       [[RTCPeerConnectionSyncObserver alloc] init];
109   RTCPeerConnection* pcOffer =
110       [factory peerConnectionWithICEServers:nil
111                                 constraints:pcConstraints
112                                    delegate:offeringExpectations];
113
114   RTCPeerConnectionSyncObserver* answeringExpectations =
115       [[RTCPeerConnectionSyncObserver alloc] init];
116
117   RTCPeerConnection* pcAnswer =
118       [factory peerConnectionWithICEServers:nil
119                                 constraints:pcConstraints
120                                    delegate:answeringExpectations];
121   // TODO(hughv): Create video capturer
122   RTCVideoCapturer* capturer = nil;
123   RTCVideoSource* videoSource =
124       [factory videoSourceWithCapturer:capturer constraints:constraints];
125
126   // Here and below, "oLMS" refers to offerer's local media stream, and "aLMS"
127   // refers to the answerer's local media stream, with suffixes of "a0" and "v0"
128   // for audio and video tracks, resp.  These mirror chrome historical naming.
129   RTCMediaStream* oLMSUnused = [self addTracksToPeerConnection:pcOffer
130                                                    withFactory:factory
131                                                    videoSource:videoSource
132                                                    streamLabel:@"oLMS"
133                                                   videoTrackID:@"oLMSv0"
134                                                   audioTrackID:@"oLMSa0"];
135
136   RTCDataChannel* offerDC =
137       [pcOffer createDataChannelWithLabel:@"offerDC"
138                                    config:[[RTCDataChannelInit alloc] init]];
139   EXPECT_TRUE([offerDC.label isEqual:@"offerDC"]);
140   offerDC.delegate = offeringExpectations;
141   offeringExpectations.dataChannel = offerDC;
142
143   RTCSessionDescriptionSyncObserver* sdpObserver =
144       [[RTCSessionDescriptionSyncObserver alloc] init];
145   [pcOffer createOfferWithDelegate:sdpObserver constraints:constraints];
146   [sdpObserver wait];
147   EXPECT_TRUE(sdpObserver.success);
148   RTCSessionDescription* offerSDP = sdpObserver.sessionDescription;
149   EXPECT_EQ([@"offer" compare:offerSDP.type options:NSCaseInsensitiveSearch],
150             NSOrderedSame);
151   EXPECT_GT([offerSDP.description length], 0);
152
153   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
154   [answeringExpectations expectSignalingChange:RTCSignalingHaveRemoteOffer];
155   [answeringExpectations expectAddStream:@"oLMS"];
156   [pcAnswer setRemoteDescriptionWithDelegate:sdpObserver
157                           sessionDescription:offerSDP];
158   [sdpObserver wait];
159
160   RTCMediaStream* aLMSUnused = [self addTracksToPeerConnection:pcAnswer
161                                                    withFactory:factory
162                                                    videoSource:videoSource
163                                                    streamLabel:@"aLMS"
164                                                   videoTrackID:@"aLMSv0"
165                                                   audioTrackID:@"aLMSa0"];
166
167   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
168   [pcAnswer createAnswerWithDelegate:sdpObserver constraints:constraints];
169   [sdpObserver wait];
170   EXPECT_TRUE(sdpObserver.success);
171   RTCSessionDescription* answerSDP = sdpObserver.sessionDescription;
172   EXPECT_EQ([@"answer" compare:answerSDP.type options:NSCaseInsensitiveSearch],
173             NSOrderedSame);
174   EXPECT_GT([answerSDP.description length], 0);
175
176   [offeringExpectations expectICECandidates:2];
177   [answeringExpectations expectICECandidates:2];
178
179   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
180   [answeringExpectations expectSignalingChange:RTCSignalingStable];
181   [pcAnswer setLocalDescriptionWithDelegate:sdpObserver
182                          sessionDescription:answerSDP];
183   [sdpObserver wait];
184   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
185
186   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
187   [offeringExpectations expectSignalingChange:RTCSignalingHaveLocalOffer];
188   [pcOffer setLocalDescriptionWithDelegate:sdpObserver
189                         sessionDescription:offerSDP];
190   [sdpObserver wait];
191   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
192
193   [offeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
194   [offeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
195   // TODO(fischman): figure out why this is flaky and re-introduce (and remove
196   // special-casing from the observer!).
197   // [offeringExpectations expectICEConnectionChange:RTCICEConnectionCompleted];
198   [answeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
199   [answeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
200
201   [offeringExpectations expectStateChange:kRTCDataChannelStateOpen];
202   [answeringExpectations expectDataChannel:@"offerDC"];
203   [answeringExpectations expectStateChange:kRTCDataChannelStateOpen];
204
205   [offeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
206   [answeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
207
208   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
209   [offeringExpectations expectSignalingChange:RTCSignalingStable];
210   [offeringExpectations expectAddStream:@"aLMS"];
211   [pcOffer setRemoteDescriptionWithDelegate:sdpObserver
212                          sessionDescription:answerSDP];
213   [sdpObserver wait];
214   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
215
216   EXPECT_TRUE([offerSDP.type isEqual:pcOffer.localDescription.type]);
217   EXPECT_TRUE([answerSDP.type isEqual:pcOffer.remoteDescription.type]);
218   EXPECT_TRUE([offerSDP.type isEqual:pcAnswer.remoteDescription.type]);
219   EXPECT_TRUE([answerSDP.type isEqual:pcAnswer.localDescription.type]);
220
221   for (RTCICECandidate* candidate in offeringExpectations
222            .releaseReceivedICECandidates) {
223     [pcAnswer addICECandidate:candidate];
224   }
225   for (RTCICECandidate* candidate in answeringExpectations
226            .releaseReceivedICECandidates) {
227     [pcOffer addICECandidate:candidate];
228   }
229
230   [offeringExpectations waitForAllExpectationsToBeSatisfied];
231   [answeringExpectations waitForAllExpectationsToBeSatisfied];
232
233   EXPECT_EQ(pcOffer.signalingState, RTCSignalingStable);
234   EXPECT_EQ(pcAnswer.signalingState, RTCSignalingStable);
235
236   // Test send and receive UTF-8 text
237   NSString* text = @"你好";
238   NSData* textData = [text dataUsingEncoding:NSUTF8StringEncoding];
239   RTCDataBuffer* buffer =
240       [[RTCDataBuffer alloc] initWithData:textData isBinary:NO];
241   [answeringExpectations expectMessage:[textData copy] isBinary:NO];
242   EXPECT_TRUE([offeringExpectations.dataChannel sendData:buffer]);
243   [answeringExpectations waitForAllExpectationsToBeSatisfied];
244
245   // Test send and receive binary data
246   const size_t byteLength = 5;
247   char bytes[byteLength] = {1, 2, 3, 4, 5};
248   NSData* byteData = [NSData dataWithBytes:bytes length:byteLength];
249   buffer = [[RTCDataBuffer alloc] initWithData:byteData isBinary:YES];
250   [answeringExpectations expectMessage:[byteData copy] isBinary:YES];
251   EXPECT_TRUE([offeringExpectations.dataChannel sendData:buffer]);
252   [answeringExpectations waitForAllExpectationsToBeSatisfied];
253
254   [offeringExpectations expectStateChange:kRTCDataChannelStateClosing];
255   [answeringExpectations expectStateChange:kRTCDataChannelStateClosing];
256   [offeringExpectations expectStateChange:kRTCDataChannelStateClosed];
257   [answeringExpectations expectStateChange:kRTCDataChannelStateClosed];
258
259   [answeringExpectations.dataChannel close];
260   [offeringExpectations.dataChannel close];
261
262   [offeringExpectations waitForAllExpectationsToBeSatisfied];
263   [answeringExpectations waitForAllExpectationsToBeSatisfied];
264   // Don't need to listen to further state changes.
265   // TODO(tkchin): figure out why Closed->Closing without this.
266   offeringExpectations.dataChannel.delegate = nil;
267   answeringExpectations.dataChannel.delegate = nil;
268
269   // Let the audio feedback run for 2s to allow human testing and to ensure
270   // things stabilize.  TODO(fischman): replace seconds with # of video frames,
271   // when we have video flowing.
272   [[NSRunLoop currentRunLoop]
273       runUntilDate:[NSDate dateWithTimeIntervalSinceNow:2]];
274
275   [offeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
276   [answeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
277   [offeringExpectations expectSignalingChange:RTCSignalingClosed];
278   [answeringExpectations expectSignalingChange:RTCSignalingClosed];
279
280   [pcOffer close];
281   [pcAnswer close];
282
283   [offeringExpectations waitForAllExpectationsToBeSatisfied];
284   [answeringExpectations waitForAllExpectationsToBeSatisfied];
285
286   capturer = nil;
287   videoSource = nil;
288   pcOffer = nil;
289   pcAnswer = nil;
290   // TODO(fischman): be stricter about shutdown checks; ensure thread
291   // counts return to where they were before the test kicked off, and
292   // that all objects have in fact shut down.
293 }
294
295 @end
296
297 // TODO(fischman): move {Initialize,Cleanup}SSL into alloc/dealloc of
298 // RTCPeerConnectionTest and avoid the appearance of RTCPeerConnectionTest being
299 // a TestBase since it's not.
300 TEST(RTCPeerConnectionTest, SessionTest) {
301   @autoreleasepool {
302     rtc::InitializeSSL();
303     // Since |factory| will own the signaling & worker threads, it's important
304     // that it outlive the created PeerConnections since they self-delete on the
305     // signaling thread, and if |factory| is freed first then a last refcount on
306     // the factory will expire during this teardown, causing the signaling
307     // thread to try to Join() with itself.  This is a hack to ensure that the
308     // factory outlives RTCPeerConnection:dealloc.
309     // See https://code.google.com/p/webrtc/issues/detail?id=3100.
310     RTCPeerConnectionFactory* factory = [[RTCPeerConnectionFactory alloc] init];
311     @autoreleasepool {
312       RTCPeerConnectionTest* pcTest = [[RTCPeerConnectionTest alloc] init];
313       [pcTest testCompleteSessionWithFactory:factory];
314     }
315     rtc::CleanupSSL();
316   }
317 }