385a722646
- Removes a strong-reference cycle between RTCPeerConnection and RTCPeerConnectionObserver - Gives RTCPeerConnectionObserver a virtual dtor - Ensures RTCPeerConnectionTest tears down correctly - Ensures AppRTCDemo tears down correctly This is the talk/ half; the webrtc/ half is in https://webrtc-codereview.appspot.com/10539005 BUG=3054,3055,3100 R=noahric@google.com Review URL: https://webrtc-codereview.appspot.com/10499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5771 4adac7df-926f-26a2-2b94-8c16560cd09d
261 lines
12 KiB
Plaintext
261 lines
12 KiB
Plaintext
/*
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* libjingle
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* Copyright 2013, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#import <Foundation/Foundation.h>
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#import "RTCICEServer.h"
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#import "RTCMediaConstraints.h"
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#import "RTCMediaStream.h"
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#import "RTCPeerConnection.h"
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#import "RTCPeerConnectionFactory.h"
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#import "RTCPeerConnectionSyncObserver.h"
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#import "RTCSessionDescription.h"
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#import "RTCSessionDescriptionSyncObserver.h"
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#import "RTCVideoRenderer.h"
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#import "RTCVideoTrack.h"
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#include "talk/base/gunit.h"
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#include "talk/base/ssladapter.h"
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#if !defined(__has_feature) || !__has_feature(objc_arc)
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#error "This file requires ARC support."
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#endif
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@interface RTCPeerConnectionTest : NSObject
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// Returns whether the two sessions are of the same type.
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+ (BOOL)isSession:(RTCSessionDescription*)session1
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ofSameTypeAsSession:(RTCSessionDescription*)session2;
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// Create and add tracks to pc, with the given source, label, and IDs
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- (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
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withFactory:(RTCPeerConnectionFactory*)factory
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videoSource:(RTCVideoSource*)videoSource
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streamLabel:(NSString*)streamLabel
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videoTrackID:(NSString*)videoTrackID
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audioTrackID:(NSString*)audioTrackID;
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- (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory;
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@end
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@implementation RTCPeerConnectionTest
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+ (BOOL)isSession:(RTCSessionDescription*)session1
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ofSameTypeAsSession:(RTCSessionDescription*)session2 {
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return [session1.type isEqual:session2.type];
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}
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- (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
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withFactory:(RTCPeerConnectionFactory*)factory
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videoSource:(RTCVideoSource*)videoSource
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streamLabel:(NSString*)streamLabel
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videoTrackID:(NSString*)videoTrackID
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audioTrackID:(NSString*)audioTrackID {
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RTCMediaStream* localMediaStream = [factory mediaStreamWithLabel:streamLabel];
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RTCVideoTrack* videoTrack =
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[factory videoTrackWithID:videoTrackID source:videoSource];
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RTCVideoRenderer* videoRenderer =
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[[RTCVideoRenderer alloc] initWithDelegate:nil];
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[videoTrack addRenderer:videoRenderer];
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[localMediaStream addVideoTrack:videoTrack];
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// Test that removal/re-add works.
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[localMediaStream removeVideoTrack:videoTrack];
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[localMediaStream addVideoTrack:videoTrack];
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RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID];
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[localMediaStream addAudioTrack:audioTrack];
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RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
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[pc addStream:localMediaStream constraints:constraints];
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return localMediaStream;
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}
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- (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory {
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RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
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RTCPeerConnectionSyncObserver* offeringExpectations =
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[[RTCPeerConnectionSyncObserver alloc] init];
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RTCPeerConnection* pcOffer =
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[factory peerConnectionWithICEServers:nil
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constraints:constraints
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delegate:offeringExpectations];
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RTCPeerConnectionSyncObserver* answeringExpectations =
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[[RTCPeerConnectionSyncObserver alloc] init];
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RTCPeerConnection* pcAnswer =
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[factory peerConnectionWithICEServers:nil
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constraints:constraints
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delegate:answeringExpectations];
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// TODO(hughv): Create video capturer
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RTCVideoCapturer* capturer = nil;
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RTCVideoSource* videoSource =
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[factory videoSourceWithCapturer:capturer constraints:constraints];
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// Here and below, "oLMS" refers to offerer's local media stream, and "aLMS"
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// refers to the answerer's local media stream, with suffixes of "a0" and "v0"
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// for audio and video tracks, resp. These mirror chrome historical naming.
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RTCMediaStream* oLMSUnused = [self addTracksToPeerConnection:pcOffer
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withFactory:factory
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videoSource:videoSource
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streamLabel:@"oLMS"
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videoTrackID:@"oLMSv0"
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audioTrackID:@"oLMSa0"];
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RTCSessionDescriptionSyncObserver* sdpObserver =
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[[RTCSessionDescriptionSyncObserver alloc] init];
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[pcOffer createOfferWithDelegate:sdpObserver constraints:constraints];
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[sdpObserver wait];
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EXPECT_TRUE(sdpObserver.success);
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RTCSessionDescription* offerSDP = sdpObserver.sessionDescription;
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EXPECT_EQ([@"offer" compare:offerSDP.type options:NSCaseInsensitiveSearch],
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NSOrderedSame);
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EXPECT_GT([offerSDP.description length], 0);
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sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
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[answeringExpectations expectSignalingChange:RTCSignalingHaveRemoteOffer];
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[answeringExpectations expectAddStream:@"oLMS"];
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[pcAnswer setRemoteDescriptionWithDelegate:sdpObserver
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sessionDescription:offerSDP];
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[sdpObserver wait];
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RTCMediaStream* aLMSUnused = [self addTracksToPeerConnection:pcAnswer
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withFactory:factory
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videoSource:videoSource
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streamLabel:@"aLMS"
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videoTrackID:@"aLMSv0"
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audioTrackID:@"aLMSa0"];
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sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
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[pcAnswer createAnswerWithDelegate:sdpObserver constraints:constraints];
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[sdpObserver wait];
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EXPECT_TRUE(sdpObserver.success);
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RTCSessionDescription* answerSDP = sdpObserver.sessionDescription;
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EXPECT_EQ([@"answer" compare:answerSDP.type options:NSCaseInsensitiveSearch],
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NSOrderedSame);
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EXPECT_GT([answerSDP.description length], 0);
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[offeringExpectations expectICECandidates:2];
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[answeringExpectations expectICECandidates:2];
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sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
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[answeringExpectations expectSignalingChange:RTCSignalingStable];
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[pcAnswer setLocalDescriptionWithDelegate:sdpObserver
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sessionDescription:answerSDP];
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[sdpObserver wait];
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EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
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sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
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[offeringExpectations expectSignalingChange:RTCSignalingHaveLocalOffer];
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[pcOffer setLocalDescriptionWithDelegate:sdpObserver
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sessionDescription:offerSDP];
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[sdpObserver wait];
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EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
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[offeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
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[offeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
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// TODO(fischman): figure out why this is flaky and re-introduce (and remove
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// special-casing from the observer!).
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// [offeringExpectations expectICEConnectionChange:RTCICEConnectionCompleted];
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[answeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
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[answeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
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[offeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
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[answeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
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sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
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[offeringExpectations expectSignalingChange:RTCSignalingStable];
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[offeringExpectations expectAddStream:@"aLMS"];
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[pcOffer setRemoteDescriptionWithDelegate:sdpObserver
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sessionDescription:answerSDP];
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[sdpObserver wait];
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EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
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EXPECT_TRUE([offerSDP.type isEqual:pcOffer.localDescription.type]);
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EXPECT_TRUE([answerSDP.type isEqual:pcOffer.remoteDescription.type]);
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EXPECT_TRUE([offerSDP.type isEqual:pcAnswer.remoteDescription.type]);
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EXPECT_TRUE([answerSDP.type isEqual:pcAnswer.localDescription.type]);
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for (RTCICECandidate* candidate in offeringExpectations
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.releaseReceivedICECandidates) {
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[pcAnswer addICECandidate:candidate];
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}
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for (RTCICECandidate* candidate in answeringExpectations
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.releaseReceivedICECandidates) {
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[pcOffer addICECandidate:candidate];
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}
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[offeringExpectations waitForAllExpectationsToBeSatisfied];
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[answeringExpectations waitForAllExpectationsToBeSatisfied];
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// Let the audio feedback run for 2s to allow human testing and to ensure
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// things stabilize. TODO(fischman): replace seconds with # of video frames,
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// when we have video flowing.
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[[NSRunLoop currentRunLoop]
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runUntilDate:[NSDate dateWithTimeIntervalSinceNow:2]];
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[offeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
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[answeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
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[offeringExpectations expectSignalingChange:RTCSignalingClosed];
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[answeringExpectations expectSignalingChange:RTCSignalingClosed];
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[pcOffer close];
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[pcAnswer close];
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[offeringExpectations waitForAllExpectationsToBeSatisfied];
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[answeringExpectations waitForAllExpectationsToBeSatisfied];
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capturer = nil;
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videoSource = nil;
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pcOffer = nil;
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pcAnswer = nil;
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// TODO(fischman): be stricter about shutdown checks; ensure thread
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// counts return to where they were before the test kicked off, and
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// that all objects have in fact shut down.
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}
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@end
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// TODO(fischman): move {Initialize,Cleanup}SSL into alloc/dealloc of
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// RTCPeerConnectionTest and avoid the appearance of RTCPeerConnectionTest being
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// a TestBase since it's not.
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TEST(RTCPeerConnectionTest, SessionTest) {
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@autoreleasepool {
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talk_base::InitializeSSL();
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// Since |factory| will own the signaling & worker threads, it's important
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// that it outlive the created PeerConnections since they self-delete on the
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// signaling thread, and if |factory| is freed first then a last refcount on
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// the factory will expire during this teardown, causing the signaling
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// thread to try to Join() with itself. This is a hack to ensure that the
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// factory outlives RTCPeerConnection:dealloc.
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// See https://code.google.com/p/webrtc/issues/detail?id=3100.
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RTCPeerConnectionFactory* factory = [[RTCPeerConnectionFactory alloc] init];
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@autoreleasepool {
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RTCPeerConnectionTest* pcTest = [[RTCPeerConnectionTest alloc] init];
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[pcTest testCompleteSessionWithFactory:factory];
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}
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talk_base::CleanupSSL();
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}
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}
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