tree a2e70be1628fa99b90cc1614be2cc194096b4648
parent a4e1b8023155feafd874d5298163418965da3680
author commit-queue@webkit.org <commit-queue@webkit.org@268f45cc-cd09-0410-ab3c-d52691b4dbfc> 1491930413 +0000
committer commit-queue@webkit.org <commit-queue@webkit.org@268f45cc-cd09-0410-ab3c-d52691b4dbfc> 1491930413 +0000

MediaStream id should be equal to msid
https://bugs.webkit.org/show_bug.cgi?id=170712

Patch by Youenn Fablet <youenn@apple.com> on 2017-04-11
Reviewed by Eric Carlson.

LayoutTests/imported/w3c:

* web-platform-tests/webrtc/datachannel-emptystring-expected.txt:
* web-platform-tests/webrtc/no-media-call-expected.txt:
* web-platform-tests/webrtc/promises-call-expected.txt:
* web-platform-tests/webrtc/rtcpeerconnection/setRemoteDescription-expected.txt:

Source/WebCore:

Covered by rebased tests.

Setting MediaStream id to libwebrtc mediastream label.
Refactoring to use more Ref<> in MediaStream code.

Making PeerConnection use the libwebrtc backend by default for layout tests instead of the mock.

* Modules/mediastream/MediaStream.cpp:
(WebCore::MediaStream::create):
(WebCore::createTrackPrivateVector):
(WebCore::MediaStream::MediaStream):
* Modules/mediastream/MediaStream.h:
* Modules/mediastream/MediaStreamRegistry.cpp:
(WebCore::MediaStreamRegistry::lookUp):
* Modules/mediastream/UserMediaRequest.cpp:
(WebCore::UserMediaRequest::allow):
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
(WebCore::LibWebRTCMediaEndpoint::mediaStreamFromRTCStream):
(WebCore::LibWebRTCMediaEndpoint::addRemoteStream):
(WebCore::LibWebRTCMediaEndpoint::addRemoteTrack):
* Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
* platform/graphics/MediaPlayer.cpp:
(WebCore::MediaPlayer::load):
* platform/graphics/MediaPlayer.h:
* platform/mediastream/MediaStreamPrivate.cpp:
(WebCore::MediaStreamPrivate::MediaStreamPrivate):
* platform/mediastream/MediaStreamPrivate.h:
(WebCore::MediaStreamPrivate::create):
* testing/Internals.cpp:
(WebCore::Internals::Internals):

LayoutTests:

Removing no longer needed/Adding needed calls to useMockRTCPeerConnectionFactory.

* fast/mediastream/RTCPeerConnection-inspect-answer.html:
* fast/mediastream/RTCPeerConnection-inspect-offer.html:
* fast/mediastream/RTCPeerConnection-legacy-stream-based-api.html:
* fast/mediastream/RTCPeerConnection-media-setup-single-dialog.html:
* fast/mediastream/RTCPeerConnection-remotely-assigned-transceiver-mid.html:
* fast/mediastream/RTCPeerConnection-setLocalDescription-offer.html:
* fast/mediastream/RTCPeerConnection-setRemoteDescription-offer.html:
* webrtc/audio-peer-connection-webaudio.html:
* webrtc/audio-replace-track.html:
* webrtc/connection-state.html:
* webrtc/datachannel/basic.html:
* webrtc/datachannel/bufferedAmountLowThreshold.html:
* webrtc/datachannel/datachannel-gc.html:
* webrtc/datachannel/datachannel-stats.html:
* webrtc/datachannel/filter-ice-candidate.html:
* webrtc/ephemeral-certificates-and-cnames.html:
* webrtc/libwebrtc/descriptionGetters.html:
* webrtc/negotiatedneeded-event-addStream.html:
* webrtc/no-port-zero-in-upd-candidates.html:
* webrtc/peer-connection-audio-mute.html:
* webrtc/peer-connection-audio-mute2.html:
* webrtc/peer-connection-remote-audio-mute.html:
* webrtc/peer-connection-remote-audio-mute2.html:
* webrtc/release-after-getting-track.html:
* webrtc/video-disabled-black.html:
* webrtc/video-getParameters.html:
* webrtc/video-mediastreamtrack-stats.html:
* webrtc/video-mute.html:
* webrtc/video-remote-mute.html:
* webrtc/video-replace-track-to-null.html:
* webrtc/video-replace-track.html:
* webrtc/video-stats.html:
* webrtc/video-with-receiver.html:
* webrtc/video.html:

git-svn-id: http://svn.webkit.org/repository/webkit/trunk@215238 268f45cc-cd09-0410-ab3c-d52691b4dbfc