Sign in
webkit
/
WebKit
/
6ada3cd2e0d9b9e450b5e021e0e5526f55f59df8
/
LayoutTests
/
webrtc
5e7f476
Remove testRunner.setWebRTCUnifiedPlanEnabled
by youenn@apple.com
· 5 years ago
af79611
Simplify data channel buffer amount low tests
by youenn@apple.com
· 5 years ago
9066559
Allow to suspend RTCPeerConnection when not connected
by youenn@apple.com
· 5 years ago
9631a4f
Update testharness.js from upstream
by cdumez@apple.com
· 5 years ago
66e0533
Source/ThirdParty/libwebrtc:
by youenn@apple.com
· 5 years ago
9a6c2ca
Make mock libwebrtc tests run with unified plan
by youenn@apple.com
· 6 years ago
e8703fd
WebRTC: got incorrect `this` in negotiationneeded event
by youenn@apple.com
· 6 years ago
c3598d4
Enable a debug WebRTC mode without any encryption
by youenn@apple.com
· 6 years ago
43434f0
Filter SDP c lines https://bugs.webkit.org/show_bug.cgi?id=199791
by youenn@apple.com
· 6 years ago
0bb9d6d
[iOS] Local capture MediaStreamTrack does not render in portrait mode
by youenn@apple.com
· 6 years ago
ae0ae88
Call was negotiated with H264 Base Profile 42e01f but encoded in High Profile
by youenn@apple.com
· 6 years ago
9122c2d
createAnswer() SDP Rejected by setLocalDescription()
by youenn@apple.com
· 6 years ago
3b8185c
Multiple videos (with audios) with autoplay & playinline not working. Only one video play at a time.
by youenn@apple.com
· 6 years ago
143e485
Video stream freeze on front camera orientation changing
by youenn@apple.com
· 6 years ago
a2cd207
RTCTrackEvent should be delayed until the whole remote description is set
by youenn@apple.com
· 6 years ago
5dfc3fe
Running RTCRtpSender.getCapabilities("video") before initial offer breaks VP8
by youenn@apple.com
· 6 years ago
5455c2e
A track source should be unmuted whenever reenabled after setDirection changes
by youenn@apple.com
· 6 years ago
faeabf8
RTCRtpSender.setParameters() does set active parameter
by youenn@apple.com
· 6 years ago
2554d50
[ MacOS iOS ] Layout Test webrtc/no-port-zero-in-upd-candidates.html is flaky timeout
by youenn@apple.com
· 6 years ago
8e5433b
Make RTCRtpSender.setParameters to activate specific encodings
by youenn@apple.com
· 6 years ago
3789c2a1
IDB should store RTCCertificate https://bugs.webkit.org/show_bug.cgi?id=192599
by youenn@apple.com
· 6 years ago
6d97381
Recycling the m section should work if it was rejected remotely
by youenn@apple.com
· 6 years ago
43a5511
DataChannels created asynchronously never open and are unusable
by youenn@apple.com
· 6 years ago
80e8583
[WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs
by commit-queue@webkit.org
· 6 years ago
1979f5a
A sender created through addTransceiver and populated using addTrack should have its source set
by youenn@apple.com
· 6 years ago
d36e6c4
Unreviewed, rolling out r238680.
by ryanhaddad@apple.com
· 6 years ago
6f45bda
A sender created through addTransceiver and populated using addTrack should have its source set
by youenn@apple.com
· 6 years ago
eca99bb
Add support for transport and peerConnection stats
by youenn@apple.com
· 6 years ago
898cc7a
RealtimeOutgoing A/V sources should observe their sources only if having a sink
by youenn@apple.com
· 6 years ago
dd10619
Calling sender.replaceTrack() twice produces a new transceiver and its corresponding m= section
by youenn@apple.com
· 6 years ago
53ad071
Make mDNS ICE Candidate an experimental flag again
by youenn@apple.com
· 6 years ago
09fc949
Make sure RTCIceCandidateStats address is undefined for host and peer reflexive case
by youenn@apple.com
· 6 years ago
7f8b82f
LibWebRTCRtpReceiverBackend::getSynchronizationSources should use Vector::append
by youenn@apple.com
· 6 years ago
c9985fb
Invalid ssrc value in the stats of type 'inbound-rtp'
by youenn@apple.com
· 6 years ago
9c8d578
Add support for IceCandidate stats
by youenn@apple.com
· 6 years ago
3725e13
Add VP8 support to WebRTC https://bugs.webkit.org/show_bug.cgi?id=189976
by youenn@apple.com
· 6 years ago
48452c8
Enable H264 simulcast https://bugs.webkit.org/show_bug.cgi?id=190167
by youenn@apple.com
· 6 years ago
ba4911a5
[MediaStream] Clean up RealtimeMediaSource settings change handling
by eric.carlson@apple.com
· 6 years ago
13f2af9
Unreviewed, rolling out r236557.
by ryanhaddad@apple.com
· 6 years ago
9fbd3d3
Add VP8 support to WebRTC https://bugs.webkit.org/show_bug.cgi?id=189976
by youenn@apple.com
· 6 years ago
9b80312
LayoutTest webrtc/video-unmute.html is a flaky timeout
by youenn@apple.com
· 6 years ago
eca81d6
Add RTCCodecStats support
by youenn@apple.com
· 6 years ago
f3edf04
Layout Test webrtc/video-mute.html is flaky.
by youenn@apple.com
· 6 years ago
ede11ee
Implement sender/receiver getStats
by youenn@apple.com
· 6 years ago
701517b
Enable Unified Plan by default https://bugs.webkit.org/show_bug.cgi?id=189675
by youenn@apple.com
· 6 years ago
8ac9535
track.onmute isn't called for a remote MediaStreamTrack when its counter part track is removed from the peer connection
by youenn@apple.com
· 6 years ago
ac355e5
Enable VCP for iOS and reenable it for MacOS
by youenn@apple.com
· 6 years ago
c2b4589
Introduce RTCRtpSendParameters https://bugs.webkit.org/show_bug.cgi?id=189563
by youenn@apple.com
· 6 years ago
d1ddef0
ontrack events should be fired even if an existing transceiver exists
by youenn@apple.com
· 6 years ago
87db97d
Add support for unified plan transceivers
by youenn@apple.com
· 6 years ago
2ea0e9c
Expose RTCRtpSender.setParameters https://bugs.webkit.org/show_bug.cgi?id=189307
by youenn@apple.com
· 6 years ago
2ce352b
Mock video devices should only support discrete sizes
by eric.carlson@apple.com
· 6 years ago
96612a9
Remove WebRTC legacy API implementation
by youenn@apple.com
· 6 years ago
2e72d4f
Add a runtime flag for WebRTC unified plan
by youenn@apple.com
· 6 years ago
9d67832
2018-08-23 Youenn Fablet <youenn@apple.com>
by youenn@apple.com
· 6 years ago
cb4613a
Support connecting a MediaStreamAudioDestinationNode to RTCPeerConnection
by youenn@apple.com
· 7 years ago
e6f8cb2
WebRTC MediaStreamTrack Enable / Disable causes video delay / lag
by youenn@apple.com
· 7 years ago
2ad9513
Incoming G722 doesn't work
by youenn@apple.com
· 7 years ago
6dbd5eb
RTCRtpSender.replaceTrack(null) ends current track
by youenn@apple.com
· 7 years ago
67ad567
Layout Test webrtc/addICECandidate-closed.html is a flaky failure
by youenn@apple.com
· 7 years ago
2496f30
Layout Test webrtc/addICECandidate-closed.html is a flaky failure
by youenn@apple.com
· 7 years ago
4536bd0
Layout Test webrtc/addICECandidate-closed.html is a flaky failure
by youenn@apple.com
· 7 years ago
2d6518b
webrtc/addICECandidate-closed.html is timing out
by youenn@apple.com
· 7 years ago
2d12347
PeerConnection should have its connectionState closed even if doing gathering
by youenn@apple.com
· 7 years ago
734fdaa
Throw in case of PeerConnection created for detached documents
by youenn@apple.com
· 7 years ago
6b19492
Preventively expect UTF8 strings from libwebrtc SDP and error messages
by youenn@apple.com
· 7 years ago
2ed85f3
webrtc/datachannel/bufferedAmountLowThreshold tests are failing on WK1
by youenn@apple.com
· 7 years ago
6c1196d
WebRTC data channel only applications require capture permissions for direct connections
by youenn@apple.com
· 7 years ago
e0ee94d
Safari WebKitWebRTCAudioModule crash during <video> tag update when audio track present in MediaStream
by youenn@apple.com
· 7 years ago
36b556d
replaceTrack triggers negotiationneeded
by youenn@apple.com
· 7 years ago
f1511de
Update libwebrtc up to 36af4e9614f707f733eb2340fae66d6325aaac5b
by youenn@apple.com
· 7 years ago
7bc6502
getUserMedia is resolving before the document knows it is capturing
by commit-queue@webkit.org
· 7 years ago
2f820b4
Allow AudioContext to start when getUserMedia is on
by commit-queue@webkit.org
· 7 years ago
801fd69
[mac-wk1] Layout test webrtc/datachannel/bufferedAmountLowThreshold tests are flaky
by commit-queue@webkit.org
· 7 years ago
b5326af
WebRTC video does not resume receiving when switching back to Safari 11 on iOS
by commit-queue@webkit.org
· 7 years ago
295249e
webrtc/peer-connection-audio-mute.html is sometimes flaky
by commit-queue@webkit.org
· 7 years ago
abfdb99
Make captureCanvas-webrtc.html more robust
by commit-queue@webkit.org
· 7 years ago
1b12247
LayoutTest webrtc/video-mute.html is very often failing
by commit-queue@webkit.org
· 7 years ago
e09dad4
Make webrtc replace track tests less flaky
by commit-queue@webkit.org
· 7 years ago
a51b27d
webrtc/video-rotation.html is failing and now occasionally times out.
by commit-queue@webkit.org
· 7 years ago
cb3f877
LayoutTest webrtc/video-getParameters.html is failing
by commit-queue@webkit.org
· 7 years ago
f749580
RTCDataChannel connectivity issues in Safari 11
by commit-queue@webkit.org
· 7 years ago
47bc29a
WebRTC: silence data not sent for disabled audio track
by commit-queue@webkit.org
· 8 years ago
5bcf6b1
Accessing localDescription, remoteDescription, etc. after setTimeout raises EXC_BAD_ACCESS
by commit-queue@webkit.org
· 8 years ago
1a6e8b7
WebRTC: Incorrect sdpMLineIndex for video breaks Firefox interop
by commit-queue@webkit.org
· 8 years ago
f7583b4
We should do ICE candidate filtering at the Document level
by commit-queue@webkit.org
· 8 years ago
e592176
Receiving tracks should be ended when peer connection is being closed
by commit-queue@webkit.org
· 8 years ago
681da36
WebAudioSourceProviderAVFObjC should not reconfigure for each data call
by commit-queue@webkit.org
· 8 years ago
fee8bd6
LayoutTest webrtc/datachannel/multiple-connections.html is a flaky timeout
by commit-queue@webkit.org
· 8 years ago
421d2bf
Make webrtc/video-replace-track-to-null.html more robust
by commit-queue@webkit.org
· 8 years ago
4f9221b
Remove use of mock webrtc backend factory at injected bundle reset time
by commit-queue@webkit.org
· 8 years ago
c3824f4
Add a binary data channel webrtc test
by commit-queue@webkit.org
· 8 years ago
db187bb
webrtc/routines.js should call createAnswer once setRemoteDescription promise is resolved
by commit-queue@webkit.org
· 8 years ago
fa32b3e
webrtc::WebRtcSession is not handling correctly its state when setLocalDescription fails and is called again
by commit-queue@webkit.org
· 8 years ago
04289e4aa
Add a test for multi data channel peer connection
by commit-queue@webkit.org
· 8 years ago
2328aba
[WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac
by commit-queue@webkit.org
· 8 years ago
d2f07b2
Improve debugging ability of some webrtc tests
by commit-queue@webkit.org
· 8 years ago
50bd1d9
Unreviewed, rolling out r218505. https://bugs.webkit.org/show_bug.cgi?id=173563
by commit-queue@webkit.org
· 8 years ago
761962d
[WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac
by commit-queue@webkit.org
· 8 years ago
3d93dc4
A cloned MediaStreamTrack should mute independently other tracks using the same source
by commit-queue@webkit.org
· 8 years ago
Next »