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/*
* libjingle
* Copyright 2013 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#import <Foundation/Foundation.h>
#import "RTCICEServer.h"
#import "RTCMediaConstraints.h"
#import "RTCMediaStream.h"
#import "RTCPair.h"
#import "RTCPeerConnection.h"
#import "RTCPeerConnectionFactory.h"
#import "RTCPeerConnectionSyncObserver.h"
#import "RTCSessionDescription.h"
#import "RTCSessionDescriptionSyncObserver.h"
#import "RTCVideoRenderer.h"
#import "RTCVideoTrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/ssladapter.h"
#if !defined(__has_feature) || !__has_feature(objc_arc)
#error "This file requires ARC support."
#endif
const NSTimeInterval kRTCPeerConnectionTestTimeout = 20;
@interface RTCFakeRenderer : NSObject <RTCVideoRenderer>
@end
@implementation RTCFakeRenderer
- (void)setSize:(CGSize)size {}
- (void)renderFrame:(RTCI420Frame*)frame {}
@end
@interface RTCPeerConnectionTest : NSObject
// Returns whether the two sessions are of the same type.
+ (BOOL)isSession:(RTCSessionDescription*)session1
ofSameTypeAsSession:(RTCSessionDescription*)session2;
// Create and add tracks to pc, with the given source, label, and IDs
- (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
withFactory:(RTCPeerConnectionFactory*)factory
videoSource:(RTCVideoSource*)videoSource
streamLabel:(NSString*)streamLabel
videoTrackID:(NSString*)videoTrackID
audioTrackID:(NSString*)audioTrackID;
- (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory;
@end
@implementation RTCPeerConnectionTest
+ (BOOL)isSession:(RTCSessionDescription*)session1
ofSameTypeAsSession:(RTCSessionDescription*)session2 {
return [session1.type isEqual:session2.type];
}
- (RTCMediaStream*)addTracksToPeerConnection:(RTCPeerConnection*)pc
withFactory:(RTCPeerConnectionFactory*)factory
videoSource:(RTCVideoSource*)videoSource
streamLabel:(NSString*)streamLabel
videoTrackID:(NSString*)videoTrackID
audioTrackID:(NSString*)audioTrackID {
RTCMediaStream* localMediaStream = [factory mediaStreamWithLabel:streamLabel];
RTCVideoTrack* videoTrack =
[factory videoTrackWithID:videoTrackID source:videoSource];
RTCFakeRenderer* videoRenderer = [[RTCFakeRenderer alloc] init];
[videoTrack addRenderer:videoRenderer];
[localMediaStream addVideoTrack:videoTrack];
// Test that removal/re-add works.
[localMediaStream removeVideoTrack:videoTrack];
[localMediaStream addVideoTrack:videoTrack];
RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID];
[localMediaStream addAudioTrack:audioTrack];
[pc addStream:localMediaStream];
return localMediaStream;
}
- (void)testCompleteSessionWithFactory:(RTCPeerConnectionFactory*)factory {
NSArray* mandatory = @[
[[RTCPair alloc] initWithKey:@"DtlsSrtpKeyAgreement" value:@"true"],
[[RTCPair alloc] initWithKey:@"internalSctpDataChannels" value:@"true"],
];
RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init];
RTCMediaConstraints* pcConstraints =
[[RTCMediaConstraints alloc] initWithMandatoryConstraints:mandatory
optionalConstraints:nil];
RTCPeerConnectionSyncObserver* offeringExpectations =
[[RTCPeerConnectionSyncObserver alloc] init];
RTCPeerConnection* pcOffer =
[factory peerConnectionWithICEServers:nil
constraints:pcConstraints
delegate:offeringExpectations];
RTCPeerConnectionSyncObserver* answeringExpectations =
[[RTCPeerConnectionSyncObserver alloc] init];
RTCPeerConnection* pcAnswer =
[factory peerConnectionWithICEServers:nil
constraints:pcConstraints
delegate:answeringExpectations];
// TODO(hughv): Create video capturer
RTCVideoCapturer* capturer = nil;
RTCVideoSource* videoSource =
[factory videoSourceWithCapturer:capturer constraints:constraints];
// Here and below, "oLMS" refers to offerer's local media stream, and "aLMS"
// refers to the answerer's local media stream, with suffixes of "a0" and "v0"
// for audio and video tracks, resp. These mirror chrome historical naming.
RTCMediaStream* oLMSUnused = [self addTracksToPeerConnection:pcOffer
withFactory:factory
videoSource:videoSource
streamLabel:@"oLMS"
videoTrackID:@"oLMSv0"
audioTrackID:@"oLMSa0"];
RTCDataChannel* offerDC =
[pcOffer createDataChannelWithLabel:@"offerDC"
config:[[RTCDataChannelInit alloc] init]];
EXPECT_TRUE([offerDC.label isEqual:@"offerDC"]);
offerDC.delegate = offeringExpectations;
offeringExpectations.dataChannel = offerDC;
RTCSessionDescriptionSyncObserver* sdpObserver =
[[RTCSessionDescriptionSyncObserver alloc] init];
[pcOffer createOfferWithDelegate:sdpObserver constraints:constraints];
[sdpObserver wait];
EXPECT_TRUE(sdpObserver.success);
RTCSessionDescription* offerSDP = sdpObserver.sessionDescription;
EXPECT_EQ([@"offer" compare:offerSDP.type options:NSCaseInsensitiveSearch],
NSOrderedSame);
EXPECT_GT([offerSDP.description length], 0);
sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
[answeringExpectations expectSignalingChange:RTCSignalingHaveRemoteOffer];
[answeringExpectations expectAddStream:@"oLMS"];
[pcAnswer setRemoteDescriptionWithDelegate:sdpObserver
sessionDescription:offerSDP];
[sdpObserver wait];
RTCMediaStream* aLMSUnused = [self addTracksToPeerConnection:pcAnswer
withFactory:factory
videoSource:videoSource
streamLabel:@"aLMS"
videoTrackID:@"aLMSv0"
audioTrackID:@"aLMSa0"];
sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
[pcAnswer createAnswerWithDelegate:sdpObserver constraints:constraints];
[sdpObserver wait];
EXPECT_TRUE(sdpObserver.success);
RTCSessionDescription* answerSDP = sdpObserver.sessionDescription;
EXPECT_EQ([@"answer" compare:answerSDP.type options:NSCaseInsensitiveSearch],
NSOrderedSame);
EXPECT_GT([answerSDP.description length], 0);
[offeringExpectations expectICECandidates:2];
// It's possible to only have 1 ICE candidate for the answerer, since we use
// BUNDLE and rtcp-mux by default, and don't provide any ICE servers in this
// test.
[answeringExpectations expectICECandidates:1];
sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
[answeringExpectations expectSignalingChange:RTCSignalingStable];
[pcAnswer setLocalDescriptionWithDelegate:sdpObserver
sessionDescription:answerSDP];
[sdpObserver wait];
EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
[offeringExpectations expectSignalingChange:RTCSignalingHaveLocalOffer];
[pcOffer setLocalDescriptionWithDelegate:sdpObserver
sessionDescription:offerSDP];
[sdpObserver wait];
EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
[offeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
[offeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
// TODO(fischman): figure out why this is flaky and re-introduce (and remove
// special-casing from the observer!).
// [offeringExpectations expectICEConnectionChange:RTCICEConnectionCompleted];
[answeringExpectations expectICEConnectionChange:RTCICEConnectionChecking];
[answeringExpectations expectICEConnectionChange:RTCICEConnectionConnected];
[offeringExpectations expectStateChange:kRTCDataChannelStateOpen];
[answeringExpectations expectDataChannel:@"offerDC"];
[answeringExpectations expectStateChange:kRTCDataChannelStateOpen];
[offeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
[answeringExpectations expectICEGatheringChange:RTCICEGatheringComplete];
sdpObserver = [[RTCSessionDescriptionSyncObserver alloc] init];
[offeringExpectations expectSignalingChange:RTCSignalingStable];
[offeringExpectations expectAddStream:@"aLMS"];
[pcOffer setRemoteDescriptionWithDelegate:sdpObserver
sessionDescription:answerSDP];
[sdpObserver wait];
EXPECT_TRUE(sdpObserver.sessionDescription == NULL);
EXPECT_TRUE([offerSDP.type isEqual:pcOffer.localDescription.type]);
EXPECT_TRUE([answerSDP.type isEqual:pcOffer.remoteDescription.type]);
EXPECT_TRUE([offerSDP.type isEqual:pcAnswer.remoteDescription.type]);
EXPECT_TRUE([answerSDP.type isEqual:pcAnswer.localDescription.type]);
for (RTCICECandidate* candidate in offeringExpectations
.releaseReceivedICECandidates) {
[pcAnswer addICECandidate:candidate];
}
for (RTCICECandidate* candidate in answeringExpectations
.releaseReceivedICECandidates) {
[pcOffer addICECandidate:candidate];
}
EXPECT_TRUE(
[offeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
EXPECT_TRUE(
[answeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
EXPECT_EQ(pcOffer.signalingState, RTCSignalingStable);
EXPECT_EQ(pcAnswer.signalingState, RTCSignalingStable);
// Test send and receive UTF-8 text
NSString* text = @"你好";
NSData* textData = [text dataUsingEncoding:NSUTF8StringEncoding];
RTCDataBuffer* buffer =
[[RTCDataBuffer alloc] initWithData:textData isBinary:NO];
[answeringExpectations expectMessage:[textData copy] isBinary:NO];
EXPECT_TRUE([offeringExpectations.dataChannel sendData:buffer]);
EXPECT_TRUE(
[answeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
// Test send and receive binary data
const size_t byteLength = 5;
char bytes[byteLength] = {1, 2, 3, 4, 5};
NSData* byteData = [NSData dataWithBytes:bytes length:byteLength];
buffer = [[RTCDataBuffer alloc] initWithData:byteData isBinary:YES];
[answeringExpectations expectMessage:[byteData copy] isBinary:YES];
EXPECT_TRUE([offeringExpectations.dataChannel sendData:buffer]);
EXPECT_TRUE(
[answeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
[offeringExpectations expectStateChange:kRTCDataChannelStateClosing];
[answeringExpectations expectStateChange:kRTCDataChannelStateClosing];
[offeringExpectations expectStateChange:kRTCDataChannelStateClosed];
[answeringExpectations expectStateChange:kRTCDataChannelStateClosed];
[answeringExpectations.dataChannel close];
[offeringExpectations.dataChannel close];
EXPECT_TRUE(
[offeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
EXPECT_TRUE(
[answeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
// Don't need to listen to further state changes.
// TODO(tkchin): figure out why Closed->Closing without this.
offeringExpectations.dataChannel.delegate = nil;
answeringExpectations.dataChannel.delegate = nil;
// Let the audio feedback run for 2s to allow human testing and to ensure
// things stabilize. TODO(fischman): replace seconds with # of video frames,
// when we have video flowing.
[[NSRunLoop currentRunLoop]
runUntilDate:[NSDate dateWithTimeIntervalSinceNow:2]];
[offeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
[answeringExpectations expectICEConnectionChange:RTCICEConnectionClosed];
[offeringExpectations expectSignalingChange:RTCSignalingClosed];
[answeringExpectations expectSignalingChange:RTCSignalingClosed];
[pcOffer close];
[pcAnswer close];
EXPECT_TRUE(
[offeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
EXPECT_TRUE(
[answeringExpectations waitForAllExpectationsToBeSatisfiedWithTimeout:
kRTCPeerConnectionTestTimeout]);
capturer = nil;
videoSource = nil;
pcOffer = nil;
pcAnswer = nil;
// TODO(fischman): be stricter about shutdown checks; ensure thread
// counts return to where they were before the test kicked off, and
// that all objects have in fact shut down.
}
@end
// TODO(fischman): move {Initialize,Cleanup}SSL into alloc/dealloc of
// RTCPeerConnectionTest and avoid the appearance of RTCPeerConnectionTest being
// a TestBase since it's not.
TEST(RTCPeerConnectionTest, SessionTest) {
@autoreleasepool {
rtc::InitializeSSL();
// Since |factory| will own the signaling & worker threads, it's important
// that it outlive the created PeerConnections since they self-delete on the
// signaling thread, and if |factory| is freed first then a last refcount on
// the factory will expire during this teardown, causing the signaling
// thread to try to Join() with itself. This is a hack to ensure that the
// factory outlives RTCPeerConnection:dealloc.
// See https://code.google.com/p/webrtc/issues/detail?id=3100.
RTCPeerConnectionFactory* factory = [[RTCPeerConnectionFactory alloc] init];
@autoreleasepool {
RTCPeerConnectionTest* pcTest = [[RTCPeerConnectionTest alloc] init];
[pcTest testCompleteSessionWithFactory:factory];
}
rtc::CleanupSSL();
}
}