- add third_party src.
[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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6  * modification, are permitted provided that the following conditions are met:
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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17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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23  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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 "RTCPeerConnection.h"
34 #import "RTCPeerConnectionFactory.h"
35 #import "RTCPeerConnectionSyncObserver.h"
36 #import "RTCSessionDescription.h"
37 #import "RTCSessionDescriptionSyncObserver.h"
38 #import "RTCVideoRenderer.h"
39 #import "RTCVideoTrack.h"
40
41 #include "talk/base/gunit.h"
42
43 #if !defined(__has_feature) || !__has_feature(objc_arc)
44 #error "This file requires ARC support."
45 #endif
46
47 @interface RTCPeerConnectionTest : NSObject
48
49 // Returns whether the two sessions are of the same type.
50 + (BOOL)isSession:(RTCSessionDescription *)session1
51     ofSameTypeAsSession:(RTCSessionDescription *)session2;
52
53 // Create and add tracks to pc, with the given source, label, and IDs
54 - (RTCMediaStream *)
55     addTracksToPeerConnection:(RTCPeerConnection *)pc
56                   withFactory:(RTCPeerConnectionFactory *)factory
57                   videoSource:(RTCVideoSource *)videoSource
58                   streamLabel:(NSString *)streamLabel
59                  videoTrackID:(NSString *)videoTrackID
60                  audioTrackID:(NSString *)audioTrackID;
61
62 - (void)testCompleteSession;
63
64 @end
65
66 @implementation RTCPeerConnectionTest
67
68 + (BOOL)isSession:(RTCSessionDescription *)session1
69     ofSameTypeAsSession:(RTCSessionDescription *)session2 {
70   return [session1.type isEqual:session2.type];
71 }
72
73 - (RTCMediaStream *)
74     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)testCompleteSession {
98   RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
99   NSString *stunURL = @"stun:stun.l.google.com:19302";
100   RTCICEServer *stunServer =
101       [[RTCICEServer alloc] initWithURI:[NSURL URLWithString:stunURL]
102                                username:@""
103                                password:@""];
104   NSArray *iceServers = @[stunServer];
105
106   RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc] init];
107   RTCPeerConnectionSyncObserver *offeringExpectations =
108       [[RTCPeerConnectionSyncObserver alloc] init];
109   RTCPeerConnection *pcOffer =
110       [factory peerConnectionWithICEServers:iceServers
111                                 constraints:constraints
112                                    delegate:offeringExpectations];
113
114   RTCPeerConnectionSyncObserver *answeringExpectations =
115       [[RTCPeerConnectionSyncObserver alloc] init];
116   RTCPeerConnection *pcAnswer =
117       [factory peerConnectionWithICEServers:iceServers
118                                 constraints:constraints
119                                    delegate:answeringExpectations];
120
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 =
130       [self addTracksToPeerConnection:pcOffer
131                           withFactory:factory
132                           videoSource:videoSource
133                           streamLabel:@"oLMS"
134                          videoTrackID:@"oLMSv0"
135                          audioTrackID:@"oLMSa0"];
136   RTCSessionDescriptionSyncObserver *sdpObserver =
137       [[RTCSessionDescriptionSyncObserver alloc] init];
138   [pcOffer createOfferWithDelegate:sdpObserver constraints:constraints];
139   [sdpObserver wait];
140   EXPECT_TRUE(sdpObserver.success);
141   RTCSessionDescription *offerSDP = sdpObserver.sessionDescription;
142   EXPECT_EQ([@"offer" compare:offerSDP.type options:NSCaseInsensitiveSearch],
143             NSOrderedSame);
144   EXPECT_GT([offerSDP.description length], 0);
145
146   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
147   [answeringExpectations
148       expectSignalingChange:RTCSignalingHaveRemoteOffer];
149   [answeringExpectations expectAddStream:@"oLMS"];
150   [pcAnswer setRemoteDescriptionWithDelegate:sdpObserver
151                           sessionDescription:offerSDP];
152   [sdpObserver wait];
153
154   RTCMediaStream *aLMSUnused =
155       [self addTracksToPeerConnection:pcAnswer
156                           withFactory:factory
157                           videoSource:videoSource
158                           streamLabel:@"aLMS"
159                          videoTrackID:@"aLMSv0"
160                          audioTrackID:@"aLMSa0"];
161
162   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
163   [pcAnswer createAnswerWithDelegate:sdpObserver constraints:constraints];
164   [sdpObserver wait];
165   EXPECT_TRUE(sdpObserver.success);
166   RTCSessionDescription *answerSDP = sdpObserver.sessionDescription;
167   EXPECT_EQ([@"answer" compare:answerSDP.type options:NSCaseInsensitiveSearch],
168             NSOrderedSame);
169   EXPECT_GT([answerSDP.description length], 0);
170
171   [offeringExpectations expectICECandidates:2];
172   [answeringExpectations expectICECandidates:2];
173
174   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
175   [answeringExpectations expectSignalingChange:RTCSignalingStable];
176   [pcAnswer setLocalDescriptionWithDelegate:sdpObserver
177                          sessionDescription:answerSDP];
178   [sdpObserver wait];
179   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
180
181   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
182   [offeringExpectations expectSignalingChange:RTCSignalingHaveLocalOffer];
183   [pcOffer setLocalDescriptionWithDelegate:sdpObserver
184                         sessionDescription:offerSDP];
185   [sdpObserver wait];
186   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
187
188   [offeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
189   [offeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
190   [answeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
191   [answeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
192
193   [offeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
194   [answeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
195
196   sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
197   [offeringExpectations expectSignalingChange:RTCSignalingStable];
198   [offeringExpectations expectAddStream:@"aLMS"];
199   [pcOffer setRemoteDescriptionWithDelegate:sdpObserver
200                          sessionDescription:answerSDP];
201   [sdpObserver wait];
202   EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
203
204   EXPECT_TRUE([offerSDP.type isEqual:pcOffer.localDescription.type]);
205   EXPECT_TRUE([answerSDP.type isEqual:pcOffer.remoteDescription.type]);
206   EXPECT_TRUE([offerSDP.type isEqual:pcAnswer.remoteDescription.type]);
207   EXPECT_TRUE([answerSDP.type isEqual:pcAnswer.localDescription.type]);
208
209   for (RTCICECandidate *candidate in
210        offeringExpectations.releaseReceivedICECandidates) {
211     [pcAnswer addICECandidate:candidate];
212   }
213   for (RTCICECandidate *candidate in
214        answeringExpectations.releaseReceivedICECandidates) {
215     [pcOffer addICECandidate:candidate];
216   }
217
218   [offeringExpectations waitForAllExpectationsToBeSatisfied];
219   [answeringExpectations waitForAllExpectationsToBeSatisfied];
220
221   // Let the audio feedback run for 10s to allow human testing and to ensure
222   // things stabilize.  TODO(fischman): replace seconds with # of video frames,
223   // when we have video flowing.
224   [[NSRunLoop currentRunLoop]
225       runUntilDate:[NSDate dateWithTimeIntervalSinceNow:10]];
226
227   // TODO(hughv): Implement orderly shutdown.
228 }
229
230 @end
231
232
233 TEST(RTCPeerConnectionTest, SessionTest) {
234   RTCPeerConnectionTest *pcTest = [[RTCPeerConnectionTest alloc] init];
235   [pcTest testCompleteSession];
236 }