From e89b7e852717b8aad4685c9e686403955d11acf6 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 17:30:19 -0700 Subject: [PATCH 01/13] Update gstreamer --- support/linux/gstreamer/gstreamer.sh | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/support/linux/gstreamer/gstreamer.sh b/support/linux/gstreamer/gstreamer.sh index 07f3acd3f3e3..b3430c9f5099 100644 --- a/support/linux/gstreamer/gstreamer.sh +++ b/support/linux/gstreamer/gstreamer.sh @@ -6,5 +6,5 @@ set -o errexit -curl -L https://servo-deps.s3.amazonaws.com/gstreamer/gstreamer-1.14-x86_64-linux-gnu.20190213.tar.gz | tar xz +curl -L https://servo-deps.s3.amazonaws.com/gstreamer/gstreamer-1.16-x86_64-linux-gnu.20190515.tar.gz | tar xz sed -i "s;prefix=/opt/gst;prefix=$PWD/gst;g" $PWD/gst/lib/pkgconfig/*.pc From ec15946f987433326a683feb4a147f01c3ada4bd Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Wed, 15 May 2019 09:12:08 -0700 Subject: [PATCH 02/13] Bump gstreamer in Travis --- .travis.yml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/.travis.yml b/.travis.yml index d550610a1e36..c33216d19fe8 100644 --- a/.travis.yml +++ b/.travis.yml @@ -23,7 +23,7 @@ matrix: - sudo apt-get update -q - sudo apt-get install clang-3.9 llvm-3.9-dev llvm-3.9-runtime libunwind8-dev -y - pip install virtualenv - - curl -L https://servo-deps.s3.amazonaws.com/gstreamer/gstreamer-1.14-x86_64-linux-gnu.20190213.tar.gz | tar xz + - curl -L https://servo-deps.s3.amazonaws.com/gstreamer/gstreamer-1.16-x86_64-linux-gnu.20190515.tar.gz | tar xz - sed -i "s;prefix=/opt/gst;prefix=$PWD/gst;g" $PWD/gst/lib/pkgconfig/*.pc - export PKG_CONFIG_PATH=$PWD/gst/lib/pkgconfig - export GST_PLUGIN_SYSTEM_PATH=$PWD/gst/lib/gstreamer-1.0 From 96b1977f242820b26592b90d99ddf0176fa49981 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 11:54:23 -0700 Subject: [PATCH 03/13] Enable webrtc tests --- tests/wpt/include.ini | 2 ++ 1 file changed, 2 insertions(+) diff --git a/tests/wpt/include.ini b/tests/wpt/include.ini index 09299d7927f4..0ed4fef25fa3 100644 --- a/tests/wpt/include.ini +++ b/tests/wpt/include.ini @@ -127,6 +127,8 @@ skip: true skip: false [webgl] skip: false +[webrtc] + skip: false [webvr] skip: false [WebIDL] From 30de72d4f4b2980609046f63ddf5566733fcf7e0 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 11:56:22 -0700 Subject: [PATCH 04/13] Enable webrtc wpt tests --- .../RTCCertificate-postMessage.html.ini | 10 + .../metadata/webrtc/RTCCertificate.html.ini | 16 + .../RTCConfiguration-bundlePolicy.html.ini | 46 + ...onfiguration-iceCandidatePoolSize.html.ini | 28 + .../RTCConfiguration-iceServers.html.ini | 226 +++ ...CConfiguration-iceTransportPolicy.html.ini | 49 + .../RTCConfiguration-rtcpMuxPolicy.html.ini | 40 + .../RTCDTMFSender-insertDTMF.https.html.ini | 22 + ...TMFSender-ontonechange-long.https.html.ini | 4 + .../RTCDTMFSender-ontonechange.https.html.ini | 40 + .../RTCDataChannel-bufferedAmount.html.ini | 37 + .../webrtc/RTCDataChannel-id.html.ini | 13 + .../webrtc/RTCDataChannel-send.html.ini | 34 + .../RTCDataChannelEvent-constructor.html.ini | 13 + ...lsTransport-getRemoteCertificates.html.ini | 4 + .../webrtc/RTCDtlsTransport-state.html.ini | 10 + tests/wpt/metadata/webrtc/RTCError.html.ini | 70 + .../RTCIceCandidate-constructor.html.ini | 55 + ...nectionState-candidate-pair.https.html.ini | 4 + .../RTCIceTransport-extension.https.html.ini | 56 + .../metadata/webrtc/RTCIceTransport.html.ini | 7 + ...ction-add-track-no-deadlock.https.html.ini | 4 + ...RTCPeerConnection-addIceCandidate.html.ini | 82 + .../RTCPeerConnection-addTrack.https.html.ini | 28 + ...erConnection-addTransceiver.https.html.ini | 37 + ...onnection-canTrickleIceCandidates.html.ini | 10 + ...rConnection-connectionState.https.html.ini | 19 + .../RTCPeerConnection-constructor.html.ini | 67 + .../RTCPeerConnection-createAnswer.html.ini | 10 + ...CPeerConnection-createDataChannel.html.ini | 124 ++ .../RTCPeerConnection-createOffer.html.ini | 13 + ...eerConnection-generateCertificate.html.ini | 25 + ...erConnection-getDefaultIceServers.html.ini | 4 + .../RTCPeerConnection-getStats.https.html.ini | 40 + ...RTCPeerConnection-getTransceivers.html.ini | 4 + ...onnectionState-disconnected.https.html.ini | 4 + ...nnection-iceConnectionState.https.html.ini | 16 + ...CPeerConnection-iceGatheringState.html.ini | 10 + .../RTCPeerConnection-ondatachannel.html.ini | 25 + ...eerConnection-onnegotiationneeded.html.ini | 38 + ...ion-onsignalingstatechanged.https.html.ini | 4 + .../RTCPeerConnection-ontrack.https.html.ini | 16 + ...onnection-remote-track-mute.https.html.ini | 13 + ...CPeerConnection-removeTrack.https.html.ini | 40 + ...ection-setDescription-transceiver.html.ini | 19 + ...ection-setLocalDescription-answer.html.ini | 19 + ...nection-setLocalDescription-offer.html.ini | 19 + ...tion-setLocalDescription-pranswer.html.ini | 13 + ...tion-setLocalDescription-rollback.html.ini | 13 + ...eerConnection-setLocalDescription.html.ini | 10 + ...ction-setRemoteDescription-answer.html.ini | 10 + ...ction-setRemoteDescription-nomsid.html.ini | 4 + ...ection-setRemoteDescription-offer.html.ini | 16 + ...ion-setRemoteDescription-pranswer.html.ini | 13 + ...oteDescription-replaceTrack.https.html.ini | 19 + ...ion-setRemoteDescription-rollback.html.ini | 16 + ...setRemoteDescription-tracks.https.html.ini | 43 + ...erConnection-setRemoteDescription.html.ini | 16 + ...CPeerConnection-track-stats.https.html.ini | 55 + ...PeerConnection-transceivers.https.html.ini | 133 ++ ...eerConnectionIceEvent-constructor.html.ini | 25 + .../webrtc/RTCRtpParameters-codecs.html.ini | 19 + ...pParameters-degradationPreference.html.ini | 7 + .../RTCRtpParameters-encodings.html.ini | 73 + ...RTCRtpParameters-headerExtensions.html.ini | 4 + .../webrtc/RTCRtpParameters-rtcp.html.ini | 7 + .../RTCRtpParameters-transactionId.html.ini | 16 + .../RTCRtpReceiver-getCapabilities.html.ini | 10 + ...iver-getContributingSources.https.html.ini | 7 + .../RTCRtpReceiver-getParameters.html.ini | 10 + .../RTCRtpReceiver-getStats.https.html.ini | 7 + ...r-getSynchronizationSources.https.html.ini | 37 + .../RTCRtpSender-getCapabilities.html.ini | 10 + .../RTCRtpSender-getStats.https.html.ini | 7 + .../RTCRtpSender-replaceTrack.https.html.ini | 28 + .../RTCRtpSender-setParameters.html.ini | 4 + .../RTCRtpSender-transport.https.html.ini | 16 + .../RTCRtpTransceiver-direction.html.ini | 10 + ...tpTransceiver-setCodecPreferences.html.ini | 28 + .../webrtc/RTCRtpTransceiver-stop.html.ini | 13 + .../webrtc/RTCRtpTransceiver.https.html.ini | 109 ++ .../RTCSctpTransport-constructor.html.ini | 13 + .../webrtc/RTCSctpTransport-events.html.ini | 7 + .../RTCSctpTransport-maxMessageSize.html.ini | 16 + .../webrtc/RTCTrackEvent-constructor.html.ini | 22 + .../webrtc/RTCTrackEvent-fire.html.ini | 7 + .../webrtc/datachannel-emptystring.html.ini | 4 + tests/wpt/metadata/webrtc/getstats.html.ini | 4 + tests/wpt/metadata/webrtc/historical.html.ini | 37 + .../webrtc/idlharness.https.window.js.ini | 1495 +++++++++++++++++ ...RTCPeerConnection-addStream.https.html.ini | 4 + ...ection-createOffer-offerToReceive.html.ini | 55 + ...with-OfferToReceive-options.https.html.ini | 13 + .../webrtc/legacy/onaddstream.https.html.ini | 4 + .../metadata/webrtc/no-media-call.html.ini | 4 + .../metadata/webrtc/promises-call.html.ini | 4 + .../webrtc/protocol/ice-state.https.html.ini | 10 + .../jsep-initial-offer.https.html.ini | 4 + .../webrtc/protocol/missing-fields.html.ini | 7 + .../webrtc/protocol/msid-parse.html.ini | 13 + .../webrtc/protocol/simulcast-answer.html.ini | 4 + .../webrtc/protocol/simulcast-offer.html.ini | 4 + .../protocol/video-codecs.https.html.ini | 10 + .../webrtc/simplecall-no-ssrcs.https.html.ini | 4 + .../metadata/webrtc/simplecall.https.html.ini | 4 + 105 files changed, 4031 insertions(+) create mode 100644 tests/wpt/metadata/webrtc/RTCCertificate-postMessage.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCCertificate.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDTMFSender-insertDTMF.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDataChannel-bufferedAmount.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDataChannel-id.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDataChannel-send.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDataChannelEvent-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCDtlsTransport-state.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCError.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCIceTransport.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-generateCertificate.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-getDefaultIceServers.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-getTransceivers.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-ondatachannel.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-codecs.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-degradationPreference.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-encodings.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-headerExtensions.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-rtcp.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpParameters-transactionId.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpReceiver-getCapabilities.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpReceiver-getParameters.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpSender-getCapabilities.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpSender-setParameters.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpTransceiver-direction.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpTransceiver-stop.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCSctpTransport-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCSctpTransport-events.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCSctpTransport-maxMessageSize.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCTrackEvent-constructor.html.ini create mode 100644 tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini create mode 100644 tests/wpt/metadata/webrtc/datachannel-emptystring.html.ini create mode 100644 tests/wpt/metadata/webrtc/getstats.html.ini create mode 100644 tests/wpt/metadata/webrtc/historical.html.ini create mode 100644 tests/wpt/metadata/webrtc/idlharness.https.window.js.ini create mode 100644 tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-addStream.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini create mode 100644 tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/no-media-call.html.ini create mode 100644 tests/wpt/metadata/webrtc/promises-call.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/missing-fields.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/simulcast-answer.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/simulcast-offer.html.ini create mode 100644 tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini create mode 100644 tests/wpt/metadata/webrtc/simplecall.https.html.ini diff --git a/tests/wpt/metadata/webrtc/RTCCertificate-postMessage.html.ini b/tests/wpt/metadata/webrtc/RTCCertificate-postMessage.html.ini new file mode 100644 index 000000000000..1de3d768d448 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCCertificate-postMessage.html.ini @@ -0,0 +1,10 @@ +[RTCCertificate-postMessage.html] + [Check cross-origin created RTCCertificate] + expected: FAIL + + [Check cross-origin RTCCertificate serialization] + expected: FAIL + + [Check same-origin RTCCertificate serialization] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCCertificate.html.ini b/tests/wpt/metadata/webrtc/RTCCertificate.html.ini new file mode 100644 index 000000000000..9f13929c0cbb --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCCertificate.html.ini @@ -0,0 +1,16 @@ +[RTCCertificate.html] + [Constructing RTCPeerConnection with expired certificate should reject with InvalidAccessError] + expected: FAIL + + [RTCCertificate should have at least one fingerprint] + expected: FAIL + + [RTCPeerConnection({ certificates }) should generate offer SDP with fingerprint of provided certificate] + expected: FAIL + + [RTCPeerConnection({ certificates }) should generate offer SDP with fingerprint of all provided certificates] + expected: FAIL + + [Calling setConfiguration with different set of certs should reject with InvalidModificationError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini new file mode 100644 index 000000000000..6e84784af60a --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini @@ -0,0 +1,46 @@ +[RTCConfiguration-bundlePolicy.html] + [setConfiguration({ bundlePolicy: 'max-compat' }) with initial bundlePolicy max-bundle should throw InvalidModificationError] + expected: FAIL + + [setConfiguration({ bundlePolicy: balanced }) with initial default bundlePolicy balanced should succeed] + expected: FAIL + + [setConfiguration({ bundlePolicy: 'balanced' }) with initial bundlePolicy balanced should succeed] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: 'max-compat' }) should succeed] + expected: FAIL + + [setConfiguration({}) with initial default bundlePolicy balanced should succeed] + expected: FAIL + + [setConfiguration({ bundlePolicy: 'max-compat' }) with initial bundlePolicy max-compat should succeed] + expected: FAIL + + [setConfiguration({}) with initial bundlePolicy max-bundle should throw InvalidModificationError] + expected: FAIL + + [setConfiguration({ bundlePolicy: 'max-bundle' }) with initial bundlePolicy max-bundle should succeed] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: undefined }) should have bundlePolicy balanced] + expected: FAIL + + [setConfiguration({}) with initial bundlePolicy balanced should succeed] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: 'max-bundle' }) should succeed] + expected: FAIL + + [Default bundlePolicy should be balanced] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: 'invalid' }) should throw TypeError] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: null }) should throw TypeError] + expected: FAIL + + [new RTCPeerConnection({ bundlePolicy: 'balanced' }) should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini new file mode 100644 index 000000000000..3ed2265cbd75 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini @@ -0,0 +1,28 @@ +[RTCConfiguration-iceCandidatePoolSize.html] + [Initialize a new RTCPeerConnection with iceCandidatePoolSize: 255] + expected: FAIL + + [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to 0] + expected: FAIL + + [Initialize a new RTCPeerConnection with iceCandidatePoolSize: 0] + expected: FAIL + + [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to 255] + expected: FAIL + + [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to 256 (Out Of Range)] + expected: FAIL + + [Initialize a new RTCPeerConnection with iceCandidatePoolSize: 256 (Out Of Range)] + expected: FAIL + + [Initialize a new RTCPeerConnection with no iceCandidatePoolSize] + expected: FAIL + + [Initialize a new RTCPeerConnection with iceCandidatePoolSize: -1 (Out Of Range)] + expected: FAIL + + [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to -1 (Out Of Range)] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini new file mode 100644 index 000000000000..9d995562d6fd --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini @@ -0,0 +1,226 @@ +[RTCConfiguration-iceServers.html] + [setConfiguration(config) - { iceServers: undefined } should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with turn server, username, credential should succeed] + expected: FAIL + + [setConfiguration(config) - with turns server and no credentials should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with 2 stun servers should succeed] + expected: FAIL + + [setConfiguration(config) - with url field should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with invalid credentialType should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - with ["stun:stun1.example.net", ""\] url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server and only credential should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with stun server, credentialType oauth, and string credential should succeed] + expected: FAIL + + [setConfiguration(config) - with one turns server, one turn server, username, credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with "" url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with stun server, credentialType password, and RTCOAuthCredential credential should succeed] + expected: FAIL + + [new RTCPeerConnection() should have default configuration.iceServers of undefined] + expected: FAIL + + [new RTCPeerConnection(config) - with credentialType token should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with turns server, credentialType oauth and RTCOAuthCredential credential should succeed] + expected: FAIL + + [setConfiguration(config) - { iceServers: null } should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: undefined } should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server and only username should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with stun server should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - {} should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server, credentialType oauth and RTCOAuthCredential credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with stun server array should succeed] + expected: FAIL + + [setConfiguration(config) - with stun server, credentialType oauth, and string credential should succeed] + expected: FAIL + + [setConfiguration(config) - with invalid stun url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with turn server, username, credential should succeed] + expected: FAIL + + [setConfiguration(config) - with turns server and only credential should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with turns server, credentialType oauth, and string credential should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with turn server and only credential should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with http url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with turns server and only username should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with turn server and only username should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with stun server should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server, credentialType oauth, and string credential should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with turn server and only credential should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with turns server and empty string username, credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with both turns and stun server, credentialType oauth and RTCOAuthCredential credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with url field should throw TypeError] + expected: FAIL + + [setConfiguration(config) - {} should succeed] + expected: FAIL + + [setConfiguration(config) - with stun server and credentialType password should succeed] + expected: FAIL + + [setConfiguration(config) - { iceServers: [{}\] } should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - with turn server and no credentials should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with stun server and credentialType password should succeed] + expected: FAIL + + [setConfiguration(config) - with empty urls should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with turn server and empty string username, credential should succeed] + expected: FAIL + + [setConfiguration(config) - with 2 stun servers should succeed] + expected: FAIL + + [setConfiguration(config) - with stun server array should succeed] + expected: FAIL + + [setConfiguration(config) - with relative url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server, credentialType password, and RTCOauthCredential credential should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with invalid turn url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - { iceServers: [undefined\] } should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server and empty string username, credential should succeed] + expected: FAIL + + [setConfiguration(config) - with "" url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with credentialType token should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: [{}\] } should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: [null\] } should throw TypeError] + expected: FAIL + + [setConfiguration(config) - { iceServers: [\] } should succeed] + expected: FAIL + + [setConfiguration(config) - { iceServers: [null\] } should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with turn server and no credentials should throw InvalidAccessError] + expected: FAIL + + [setConfiguration(config) - with stun server, credentialType password, and RTCOAuthCredential credential should succeed] + expected: FAIL + + [setConfiguration(config) - with ["stun:stun1.example.net", ""\] url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with turn server and only username should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: [\] } should succeed] + expected: FAIL + + [setConfiguration(config) - with both turns and stun server, credentialType oauth and RTCOAuthCredential credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with relative url should throw SyntaxError] + expected: FAIL + + [setConfiguration(config) - with invalid turn url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with turn server and empty string username, credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: null } should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with http url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with invalid stun url should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - with empty urls should throw SyntaxError] + expected: FAIL + + [new RTCPeerConnection(config) - { iceServers: [undefined\] } should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - with invalid credentialType should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with turns server, credentialType password, and RTCOauthCredential credential should throw InvalidAccessError] + expected: FAIL + + [new RTCPeerConnection(config) - with one turns server, one turn server, username, credential should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with turns server and no credentials should throw InvalidAccessError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini new file mode 100644 index 000000000000..b2f3e0a5c479 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini @@ -0,0 +1,49 @@ +[RTCConfiguration-iceTransportPolicy.html] + [setConfiguration(config) - with null iceTransportPolicy should throw TypeError] + expected: FAIL + + [new RTCPeerConnection(config) - with none iceTransportPolicy should throw TypeError] + expected: FAIL + + [new RTCPeerConnection({ iceTransportPolicy: undefined }) should have default iceTransportPolicy all] + expected: FAIL + + [setConfiguration({}) with initial iceTransportPolicy relay should set new value to all] + expected: FAIL + + [new RTCPeerConnection(config) - with invalid iceTransportPolicy should throw TypeError] + expected: FAIL + + [new RTCPeerConnection({ iceTransportPolicy: 'relay' }) should succeed] + expected: FAIL + + [new RTCPeerConnection(config) - with null iceTransportPolicy should throw TypeError] + expected: FAIL + + [setConfiguration({ iceTransportPolicy: 'relay' }) with initial iceTransportPolicy all should succeed] + expected: FAIL + + [new RTCPeerConnection({ iceTransports: 'invalid' }) should have no effect] + expected: FAIL + + [new RTCPeerConnection() should have default iceTransportPolicy all] + expected: FAIL + + [new RTCPeerConnection({ iceTransports: 'relay' }) should have no effect] + expected: FAIL + + [setConfiguration({ iceTransportPolicy: 'all' }) with initial iceTransportPolicy relay should succeed] + expected: FAIL + + [new RTCPeerConnection({ iceTransports: null }) should have no effect] + expected: FAIL + + [setConfiguration(config) - with none iceTransportPolicy should throw TypeError] + expected: FAIL + + [setConfiguration(config) - with invalid iceTransportPolicy should throw TypeError] + expected: FAIL + + [new RTCPeerConnection({ iceTransportPolicy: 'all' }) should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini new file mode 100644 index 000000000000..2d233dfc3703 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini @@ -0,0 +1,40 @@ +[RTCConfiguration-rtcpMuxPolicy.html] + [setConfiguration(config) - with { rtcpMuxPolicy: 'invalid' } should throw TypeError] + expected: FAIL + + [setRemoteDescription throws InvalidAccessError when called with an answer without rtcp-mux and rtcpMuxPolicy is set to require] + expected: FAIL + + [new RTCPeerConnection({ rtcpMuxPolicy: undefined }) should have default rtcpMuxPolicy require] + expected: FAIL + + [new RTCPeerConnection({ rtcpMuxPolicy: 'require' }) should succeed] + expected: FAIL + + [new RTCPeerConnection({ rtcpMuxPolicy: 'negotiate' }) may succeed or throw NotSupportedError] + expected: FAIL + + [new RTCPeerConnection(config) - with { rtcpMuxPolicy: 'invalid' } should throw TypeError] + expected: FAIL + + [setConfiguration({ rtcpMuxPolicy: 'require' }) with initial rtcpMuxPolicy negotiate should throw InvalidModificationError] + expected: FAIL + + [setConfiguration(config) - with { rtcpMuxPolicy: null } should throw TypeError] + expected: FAIL + + [setRemoteDescription throws InvalidAccessError when called with an offer without rtcp-mux and rtcpMuxPolicy is set to require] + expected: FAIL + + [setConfiguration({}) with initial rtcpMuxPolicy negotiate should throw InvalidModificationError] + expected: FAIL + + [setConfiguration({ rtcpMuxPolicy: 'negotiate' }) with initial rtcpMuxPolicy require should throw InvalidModificationError] + expected: FAIL + + [new RTCPeerConnection() should have default rtcpMuxPolicy require] + expected: FAIL + + [new RTCPeerConnection(config) - with { rtcpMuxPolicy: null } should throw TypeError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDTMFSender-insertDTMF.https.html.ini b/tests/wpt/metadata/webrtc/RTCDTMFSender-insertDTMF.https.html.ini new file mode 100644 index 000000000000..4f60e5ef211c --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDTMFSender-insertDTMF.https.html.ini @@ -0,0 +1,22 @@ +[RTCDTMFSender-insertDTMF.https.html] + [insertDTMF() should throw InvalidCharacterError if tones contains invalid DTMF characters] + expected: FAIL + + [insertDTMF() after remove and close should reject] + expected: FAIL + + [insertDTMF() should throw InvalidStateError if transceiver is stopped] + expected: FAIL + + [insertDTMF() should set toneBuffer to provided tones normalized, with old tones overridden] + expected: FAIL + + [insertDTMF() should throw InvalidStateError if transceiver.currentDirection is recvonly] + expected: FAIL + + [insertDTMF() should succeed if tones contains valid DTMF characters] + expected: FAIL + + [insertDTMF() should throw InvalidStateError if transceiver.currentDirection is inactive] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini b/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini new file mode 100644 index 000000000000..2bd47cbeb7bc --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange-long.https.html.ini @@ -0,0 +1,4 @@ +[RTCDTMFSender-ontonechange-long.https.html] + [insertDTMF with duration greater than 6000 should be clamped to 6000] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange.https.html.ini b/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange.https.html.ini new file mode 100644 index 000000000000..d998f28fcf10 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDTMFSender-ontonechange.https.html.ini @@ -0,0 +1,40 @@ +[RTCDTMFSender-ontonechange.https.html] + [Tone change event with unexpected name should not crash] + expected: FAIL + + [insertDTMF('') should not fire any tonechange event, including for '' tone] + expected: FAIL + + [Calling insertDTMF('') in the middle of tonechange events should stop future tonechange events from firing] + expected: FAIL + + [Setting transceiver.currentDirection to recvonly in the middle of tonechange events should stop future tonechange events from firing] + expected: FAIL + + [Tone change event constructor works] + expected: FAIL + + [insertDTMF() with duration less than 40 should be clamped to 40] + expected: FAIL + + [insertDTMF() with explicit duration and intertoneGap should fire tonechange events at the expected time] + expected: FAIL + + [Calling insertDTMF() in the middle of tonechange events should cause future tonechanges to be updated to new tones] + expected: FAIL + + [insertDTMF with comma should delay next tonechange event for a constant 2000ms] + expected: FAIL + + [insertDTMF() with interToneGap less than 30 should be clamped to 30] + expected: FAIL + + [insertDTMF() with default duration and intertoneGap should fire tonechange events at the expected time] + expected: FAIL + + [insertDTMF() with transceiver stopped in the middle should stop future tonechange events from firing] + expected: FAIL + + [Calling insertDTMF() multiple times in the middle of tonechange events should cause future tonechanges to be updated the last provided tones] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDataChannel-bufferedAmount.html.ini b/tests/wpt/metadata/webrtc/RTCDataChannel-bufferedAmount.html.ini new file mode 100644 index 000000000000..149a1d39488b --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDataChannel-bufferedAmount.html.ini @@ -0,0 +1,37 @@ +[RTCDataChannel-bufferedAmount.html] + [bufferedAmount should not decrease immediately after initiating closure] + expected: FAIL + + [Data channel bufferedamount is data.length on send(data)] + expected: FAIL + + [Data channel bufferedamountlow event fires only once after multiple consecutive send() calls] + expected: FAIL + + [bufferedAmount should increase to byte length of buffer sent] + expected: FAIL + + [Data channel bufferedamountlow event fires after each sent message] + expected: FAIL + + [bufferedAmount should increase by byte length for each message sent] + expected: FAIL + + [Data channel bufferedamount returns the same amount if no more data is sent on the channel] + expected: FAIL + + [bufferedAmount initial value should be 0 for both peers] + expected: FAIL + + [Data channel bufferedamountlow event fires after send() is complete] + expected: FAIL + + [bufferedAmount should not decrease after closing the peer connection] + expected: FAIL + + [bufferedAmount should increase to size of blob sent] + expected: FAIL + + [bufferedAmount should increase to byte length of encoded unicode string sent] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDataChannel-id.html.ini b/tests/wpt/metadata/webrtc/RTCDataChannel-id.html.ini new file mode 100644 index 000000000000..045d6b7385c0 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDataChannel-id.html.ini @@ -0,0 +1,13 @@ +[RTCDataChannel-id.html] + [DTLS server uses even data channel IDs] + expected: FAIL + + [Odd/even role should not be violated when mixing with negotiated channels] + expected: FAIL + + [In-band negotiation with a specific ID should not work] + expected: FAIL + + [DTLS client uses odd data channel IDs] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDataChannel-send.html.ini b/tests/wpt/metadata/webrtc/RTCDataChannel-send.html.ini new file mode 100644 index 000000000000..4c01d24cf90b --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDataChannel-send.html.ini @@ -0,0 +1,34 @@ +[RTCDataChannel-send.html] + [Data channel should be able to send unicode string and receive as unicode string] + expected: FAIL + + [Sending multiple messages with different types should succeed and be received] + expected: FAIL + + [Data channel should be able to send Uint8Array message and receive as ArrayBuffer] + expected: FAIL + + [Calling send() up to max size should succeed, above max size should fail] + expected: FAIL + + [Data channel should be able to send simple string and receive as string] + expected: FAIL + + [Calling send() when data channel is in connecting state should throw InvalidStateError] + expected: FAIL + + [Data channel binaryType should receive message as Blob by default] + expected: FAIL + + [Data channel should ignore binaryType and always receive string message as string] + expected: FAIL + + [Data channel should be able to send Blob message and receive as ArrayBuffer] + expected: FAIL + + [Data channel should be able to send ArrayBuffer message and receive as Blob] + expected: FAIL + + [Data channel should be able to send ArrayBuffer message and receive as ArrayBuffer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDataChannelEvent-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCDataChannelEvent-constructor.html.ini new file mode 100644 index 000000000000..a0a7c8e93875 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDataChannelEvent-constructor.html.ini @@ -0,0 +1,13 @@ +[RTCDataChannelEvent-constructor.html] + [RTCDataChannelEvent constructor with a channel passed as undefined.] + expected: FAIL + + [RTCDataChannelEvent constructor without a required argument.] + expected: FAIL + + [RTCDataChannelEvent constructor with channel passed as null.] + expected: FAIL + + [RTCDataChannelEvent constructor with full arguments.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini b/tests/wpt/metadata/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini new file mode 100644 index 000000000000..f8a830f38553 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDtlsTransport-getRemoteCertificates.html.ini @@ -0,0 +1,4 @@ +[RTCDtlsTransport-getRemoteCertificates.html] + [RTCDtlsTransport.prototype.getRemoteCertificates] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCDtlsTransport-state.html.ini b/tests/wpt/metadata/webrtc/RTCDtlsTransport-state.html.ini new file mode 100644 index 000000000000..0257405637b2 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCDtlsTransport-state.html.ini @@ -0,0 +1,10 @@ +[RTCDtlsTransport-state.html] + [close() causes the other end's DTLS transport to close] + expected: FAIL + + [DTLS transport goes to connected state] + expected: FAIL + + [close() causes the local transport to close immediately] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCError.html.ini b/tests/wpt/metadata/webrtc/RTCError.html.ini new file mode 100644 index 000000000000..1020c07a75f9 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCError.html.ini @@ -0,0 +1,70 @@ +[RTCError.html] + [RTCError.sentAlert is readonly] + expected: FAIL + + [RTCError.errorDetail is readonly.] + expected: FAIL + + [RTCError constructor throws TypeError if the errorDetail is invalid] + expected: FAIL + + [RTCError constructor with errorDetail and message] + expected: FAIL + + [RTCError.receivedAlert is settable by constructor] + expected: FAIL + + [RTCError.sctpCauseCode is readonly] + expected: FAIL + + [RTCError.code is 0] + expected: FAIL + + [RTCError.sdpLineNumber is null by default] + expected: FAIL + + [RTCError constructor's message argument is optional] + expected: FAIL + + [RTCError.httpRequestStatusCode is settable by constructor] + expected: FAIL + + [RTCError.sentAlert is null by default] + expected: FAIL + + [RTCError.sentAlert is settable by constructor] + expected: FAIL + + [RTCError.receivedAlert is null by default] + expected: FAIL + + [RTCError.httpRequestStatusCode is null by default] + expected: FAIL + + [RTCError.receivedAlert is readonly] + expected: FAIL + + [RTCErrorInit.errorDetail is the only required attribute] + expected: FAIL + + [RTCError.sdpLineNumber is settable by constructor] + expected: FAIL + + [RTCError constructor throws TypeError if arguments are missing] + expected: FAIL + + [RTCError.sdpLineNumber is readonly] + expected: FAIL + + [RTCError.sctpCauseCode is settable by constructor] + expected: FAIL + + [RTCError.sctpCauseCode is null by default] + expected: FAIL + + [RTCError.httpRequestStatusCode is readonly] + expected: FAIL + + [RTCError.name is 'RTCError'] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini new file mode 100644 index 000000000000..e2ba0bd7c582 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini @@ -0,0 +1,55 @@ +[RTCIceCandidate-constructor.html] + [new RTCIceCandidate({ sdpMLineIndex: 0 })] + expected: FAIL + + [new RTCIceCandidate({ ... }) with invalid sdpMid] + expected: FAIL + + [new RTCIceCandidate({ ... }) with invalid sdpMLineIndex] + expected: FAIL + + [new RTCIceCandidate()] + expected: FAIL + + [new RTCIceCandidate({ sdpMid: null, sdpMLineIndex: null })] + expected: FAIL + + [new RTCIceCandidate({ ... }) with valid candidate string and sdpMid] + expected: FAIL + + [new RTCIceCandidate({ ... }) with nondefault values for all fields, tcp candidate] + expected: FAIL + + [new RTCIceCandidate({ candidate: '', sdpMid: 'audio' }] + expected: FAIL + + [new RTCIceCandidate({ candidate: null })] + expected: FAIL + + [new RTCIceCandidate({})] + expected: FAIL + + [new RTCIceCandidate({ sdpMid: 'audio' })] + expected: FAIL + + [new RTCIceCandidate({ ... }) with manually filled default values] + expected: FAIL + + [new RTCIceCandidate({ candidate: '', sdpMLineIndex: 0 }] + expected: FAIL + + [new RTCIceCandidate({ ... }) with nondefault values for all fields] + expected: FAIL + + [new RTCIceCandidate({ candidate: '' })] + expected: FAIL + + [new RTCIceCandidate({ ... }) with invalid candidate string and sdpMid] + expected: FAIL + + [new RTCIceCandidate({ ... }) with valid candidate string only] + expected: FAIL + + [new RTCIceCandidate({ sdpMid: 'audio', sdpMLineIndex: 0 })] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini b/tests/wpt/metadata/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini new file mode 100644 index 000000000000..0d56318781f7 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCIceConnectionState-candidate-pair.https.html.ini @@ -0,0 +1,4 @@ +[RTCIceConnectionState-candidate-pair.https.html] + [On ICE connected, getStats() contains a connected candidate-pair] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini new file mode 100644 index 000000000000..7c58f72543c8 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini @@ -0,0 +1,56 @@ +[RTCIceTransport-extension.https.html] + expected: ERROR + [RTCIceTransport initial properties are set] + expected: FAIL + + [gather() with { iceServers: null } should throw TypeError] + expected: FAIL + + [gather() returns no candidates with { gatherPolicy: 'relay'} and no turn servers] + expected: FAIL + + [gather() throws if called twice] + expected: FAIL + + [gather() returns at least one host candidate] + expected: FAIL + + [gather() with one turns server, one turn server, username, credential should succeed] + expected: FAIL + + [onicecandidate fires with null candidate before gatheringState transitions to 'complete'] + expected: FAIL + + [start() does not transition state to 'checking' if no remote candidates added] + expected: FAIL + + [gather() transitions gatheringState to 'gathering'] + expected: FAIL + + [start() with default role sets role attribute to 'controlled'] + expected: FAIL + + [eventually transition gatheringState to 'complete'] + expected: FAIL + + [gather() with { iceServers: undefined } should succeed] + expected: FAIL + + [RTCIceTransport constructor does not throw] + expected: FAIL + + [gather() with 2 stun servers should succeed] + expected: FAIL + + [start() throws if usernameFragment or password not set] + expected: FAIL + + [start() sets role attribute to 'controlling'] + expected: FAIL + + [start() throws if closed] + expected: FAIL + + [gather() throws if closed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCIceTransport.html.ini b/tests/wpt/metadata/webrtc/RTCIceTransport.html.ini new file mode 100644 index 000000000000..bccbf4558ea8 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCIceTransport.html.ini @@ -0,0 +1,7 @@ +[RTCIceTransport.html] + [Two connected iceTransports should has matching local/remote candidates returned] + expected: FAIL + + [Unconnected iceTransport should have empty remote candidates and selected pair] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini new file mode 100644 index 000000000000..3e9d2b4aaca3 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-add-track-no-deadlock.https.html] + [RTCPeerConnection addTrack does not deadlock.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini new file mode 100644 index 000000000000..e82723288688 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini @@ -0,0 +1,82 @@ +[RTCPeerConnection-addIceCandidate.html] + [Add candidate with only valid sdpMLineIndex should succeed] + expected: FAIL + + [addIceCandidate({usernameFragment: usernameFragment1, sdpMid: sdpMid1}) should work, and add a=end-of-candidates to the first m-section] + expected: FAIL + + [addIceCandidate(null) should work, and add a=end-of-candidates to both m-sections] + expected: FAIL + + [Add candidate with only valid sdpMid should succeed] + expected: FAIL + + [Add ICE candidate after setting remote description should succeed] + expected: FAIL + + [addIceCandidate({usernameFragment: "no such ufrag"}) should not work] + expected: FAIL + + [Add candidate with invalid candidate string and both sdpMid and sdpMLineIndex null should reject with TypeError] + expected: FAIL + + [Add candidate with both sdpMid and sdpMLineIndex manually set to null should reject with TypeError] + expected: FAIL + + [addIceCandidate with second sdpMid and sdpMLineIndex should add candidate to second media stream] + expected: FAIL + + [Add candidate with invalid sdpMid should reject with OperationError] + expected: FAIL + + [Add candidate with sdpMid belonging to different usernameFragment should reject with OperationError] + expected: FAIL + + [Add candidate with only valid candidate string should reject with TypeError] + expected: FAIL + + [Add candidate with invalid sdpMLineIndex should reject with OperationError] + expected: FAIL + + [Add ICE candidate with RTCIceCandidate should succeed] + expected: FAIL + + [Add candidate for first media stream with null usernameFragment should add candidate to first media stream] + expected: FAIL + + [Add candidate for media stream 2 with null usernameFragment should succeed] + expected: FAIL + + [Adding multiple candidates should add candidates to their corresponding media stream] + expected: FAIL + + [addIceCandidate({usernameFragment: usernameFragment2, sdpMLineIndex: 1}) should work, and add a=end-of-candidates to the first m-section] + expected: FAIL + + [Add ICE candidate before setting remote description should reject with InvalidStateError] + expected: FAIL + + [Add candidate with invalid candidate string should reject with OperationError] + expected: FAIL + + [addIceCandidate(undefined) should work, and add a=end-of-candidates to both m-sections] + expected: FAIL + + [Add candidate with invalid usernameFragment should reject with OperationError] + expected: FAIL + + [Add with empty candidate string (end of candidate) should succeed] + expected: FAIL + + [addIceCandidate({}) should work, and add a=end-of-candidates to both m-sections] + expected: FAIL + + [addIceCandidate with first sdpMid and sdpMLineIndex add candidate to first media stream] + expected: FAIL + + [addIceCandidate({"candidate":"","sdpMid":null,"sdpMLineIndex":null}) should work, and add a=end-of-candidates to both m-sections] + expected: FAIL + + [Invalid sdpMLineIndex should be ignored if valid sdpMid is provided] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini new file mode 100644 index 000000000000..e3ad91184bb7 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini @@ -0,0 +1,28 @@ +[RTCPeerConnection-addTrack.https.html] + [addTrack with existing sender that has been used to send should create new sender] + expected: FAIL + + [addTrack with single track argument and multiple streams should succeed] + expected: FAIL + + [addTrack with existing sender with null track, different kind, and recvonly direction should create new sender] + expected: FAIL + + [addTrack with single track argument and no stream should succeed] + expected: FAIL + + [addTrack with existing sender that has not been used to send should reuse the sender] + expected: FAIL + + [addTrack with single track argument and single stream should succeed] + expected: FAIL + + [addTrack when pc is closed should throw InvalidStateError] + expected: FAIL + + [addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender] + expected: FAIL + + [Adding the same track multiple times should throw InvalidAccessError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini new file mode 100644 index 000000000000..1b5424929ef5 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini @@ -0,0 +1,37 @@ +[RTCPeerConnection-addTransceiver.https.html] + [addTransceiver() with direction inactive should have result transceiver.direction be the same] + expected: FAIL + + [addTransceiver() with direction sendonly should have result transceiver.direction be the same] + expected: FAIL + + [addTransceiver() with valid sendEncodings should succeed] + expected: FAIL + + [addTransceiver('audio') should return an audio transceiver] + expected: FAIL + + [addTransceiver() with valid rid value should succeed] + expected: FAIL + + [addTransceiver('video') should return a video transceiver] + expected: FAIL + + [addTransceiver() with rid longer than 16 characters should throw TypeError] + expected: FAIL + + [addTransceiver() with rid containing invalid non-alphanumeric characters should throw TypeError] + expected: FAIL + + [addTransceiver(track) should have result with sender.track be given track] + expected: FAIL + + [addTransceiver() with string argument as invalid kind should throw TypeError] + expected: FAIL + + [addTransceiver() with invalid direction should throw TypeError] + expected: FAIL + + [addTransceiver(track) multiple times should create multiple transceivers] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini new file mode 100644 index 000000000000..03e5bc90ed64 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-canTrickleIceCandidates.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-canTrickleIceCandidates.html] + [canTrickleIceCandidates property is true after setRemoteDescription with a=ice-options:trickle] + expected: FAIL + + [canTrickleIceCandidates property is false after setRemoteDescription without a=ice-options:trickle] + expected: FAIL + + [canTrickleIceCandidates property is null prior to setRemoteDescription] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini new file mode 100644 index 000000000000..1b387aeb1f73 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -0,0 +1,19 @@ +[RTCPeerConnection-connectionState.https.html] + [Initial connectionState should be new] + expected: FAIL + + [connectionState remains new when not adding remote ice candidates] + expected: FAIL + + [connectionState transitions to connected via connecting] + expected: FAIL + + [connection with one data channel should eventually have transports in connected state] + expected: FAIL + + [connection with one data channel should eventually have connected connection state] + expected: FAIL + + [Closing the connection should set connectionState to closed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini new file mode 100644 index 000000000000..0e8361a8bb5b --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini @@ -0,0 +1,67 @@ +[RTCPeerConnection-constructor.html] + [signalingState initial value] + expected: FAIL + + [iceConnectionState initial value] + expected: FAIL + + [new RTCPeerConnection()] + expected: FAIL + + [connectionState initial value] + expected: FAIL + + [pendingRemoteDescription initial value] + expected: FAIL + + [pendingLocalDescription initial value] + expected: FAIL + + [new RTCPeerConnection({ iceCandidatePoolSize: toNumberThrows })] + expected: FAIL + + [iceGatheringState initial value] + expected: FAIL + + [RTCPeerConnection.length] + expected: FAIL + + [new RTCPeerConnection({})] + expected: FAIL + + [new RTCPeerConnection({ certificates: null })] + expected: FAIL + + [canTrickleIceCandidates initial value] + expected: FAIL + + [localDescription initial value] + expected: FAIL + + [currentLocalDescription initial value] + expected: FAIL + + [new RTCPeerConnection({ certificates: [null\] })] + expected: FAIL + + [new RTCPeerConnection({ certificates: undefined })] + expected: FAIL + + [currentRemoteDescription initial value] + expected: FAIL + + [new RTCPeerConnection(null)] + expected: FAIL + + [new RTCPeerConnection({ certificates: [\] })] + expected: FAIL + + [new RTCPeerConnection(undefined)] + expected: FAIL + + [remoteDescription initial value] + expected: FAIL + + [new RTCPeerConnection({ certificates: [undefined\] })] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini new file mode 100644 index 000000000000..3fc9569af419 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-createAnswer.html] + [createAnswer() when connection is closed reject with InvalidStateError] + expected: FAIL + + [createAnswer() after setting remote description should succeed] + expected: FAIL + + [createAnswer() with null remoteDescription should reject with InvalidStateError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini new file mode 100644 index 000000000000..deea9840f2a7 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini @@ -0,0 +1,124 @@ +[RTCPeerConnection-createDataChannel.html] + [createDataChannel with maxPacketLifeTime 0 should succeed] + expected: FAIL + + [createDataChannel with protocol null should succeed] + expected: FAIL + + [createDataChannel with closed connection should throw InvalidStateError] + expected: FAIL + + [createDataChannel with protocol undefined should succeed] + expected: FAIL + + [createDataChannel with negotiated false should succeed] + expected: FAIL + + [createDataChannel with negotiated true and id not defined should throw TypeError] + expected: FAIL + + [createDataChannel with negotiated false and id 42 should ignore the id] + expected: FAIL + + [createDataChannel with both maxPacketLifeTime and maxRetransmits should throw TypeError] + expected: FAIL + + [createDataChannel with label undefined should succeed] + expected: FAIL + + [Reusing a data channel id that is in use (after setRemoteDescription, negotiated via DCEP) should throw OperationError] + expected: FAIL + + [Reusing a data channel id that is in use (after setRemoteDescription) should throw OperationError] + expected: FAIL + + [New data channel should be in the connecting state after creation (after connection establishment)] + expected: FAIL + + [createDataChannel with maxRetransmits 0 should succeed] + expected: FAIL + + [createDataChannel with ordered null/undefined should succeed] + expected: FAIL + + [createDataChannel with too long label should throw TypeError] + expected: FAIL + + [createDataChannel with label "foo" should succeed] + expected: FAIL + + [Reusing a data channel id that is in use should throw OperationError] + expected: FAIL + + [createDataChannel with no argument should throw TypeError] + expected: FAIL + + [createDataChannel with too long label (2 byte unicode) should throw TypeError] + expected: FAIL + + [createDataChannel with id -1 should throw TypeError] + expected: FAIL + + [createDataChannel with invalid priority should throw TypeError] + expected: FAIL + + [createDataChannel with id 0 should succeed] + expected: FAIL + + [createDataChannel attribute default values] + expected: FAIL + + [createDataChannel with too long protocol (2 byte unicode) should throw TypeError] + expected: FAIL + + [Channels created (after setRemoteDescription) should have id assigned] + expected: FAIL + + [createDataChannel with ordered false should succeed] + expected: FAIL + + [createDataChannel with too long protocol should throw TypeError] + expected: FAIL + + [createDataChannel with label lone surrogate should succeed] + expected: FAIL + + [createDataChannel with label null should succeed] + expected: FAIL + + [createDataChannel with id 1 should succeed] + expected: FAIL + + [createDataChannel with both maxPacketLifeTime and maxRetransmits undefined should succeed] + expected: FAIL + + [createDataChannel with negotiated true and id should succeed] + expected: FAIL + + [createDataChannel with maximum length label and protocol should succeed] + expected: FAIL + + [createDataChannel with provided parameters should initialize attributes to provided values] + expected: FAIL + + [createDataChannel with protocol "foo" should succeed] + expected: FAIL + + [createDataChannel with priority "high" should succeed] + expected: FAIL + + [createDataChannel with protocol lone surrogate should succeed] + expected: FAIL + + [createDataChannel with id 65534 should succeed] + expected: FAIL + + [createDataChannel with same label used twice should not throw] + expected: FAIL + + [createDataChannel with id 65535 should throw TypeError] + expected: FAIL + + [createDataChannel with id 65536 should throw TypeError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini new file mode 100644 index 000000000000..794e6de939a9 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini @@ -0,0 +1,13 @@ +[RTCPeerConnection-createOffer.html] + [createOffer() after connection is closed should reject with InvalidStateError] + expected: FAIL + + [When media stream is added when createOffer() is running in parallel, the result offer should contain the new media stream] + expected: FAIL + + [createOffer() and then setLocalDescription() should succeed] + expected: FAIL + + [createOffer() with no argument from newly created RTCPeerConnection should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-generateCertificate.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-generateCertificate.html.ini new file mode 100644 index 000000000000..6bd76a79b4a2 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-generateCertificate.html.ini @@ -0,0 +1,25 @@ +[RTCPeerConnection-generateCertificate.html] + [generateCertificate() with invalid string algorithm should reject with NotSupportedError] + expected: FAIL + + [generateCertificate() with invalid range for expires should reject with TypeError] + expected: FAIL + + [generateCertificate() with 0 expires parameter should generate expired cert] + expected: FAIL + + [generateCertificate() with compulsary ECDSA parameters should succeed] + expected: FAIL + + [generateCertificate() with invalid algorithm dict should reject with NotSupportedError] + expected: FAIL + + [generateCertificate() with valid expires parameter should succeed] + expected: FAIL + + [generateCertificate() with invalid type for expires should reject with TypeError] + expected: FAIL + + [generateCertificate() with compulsary RSASSA-PKCS1-v1_5 parameters should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getDefaultIceServers.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getDefaultIceServers.html.ini new file mode 100644 index 000000000000..092d1f657d3e --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getDefaultIceServers.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-getDefaultIceServers.html] + [RTCPeerConnection.getDefaultIceServers() should return array of RTCIceServer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini new file mode 100644 index 000000000000..0b0f45d1843e --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -0,0 +1,40 @@ +[RTCPeerConnection-getStats.https.html] + [getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats] + expected: FAIL + + [getStats() with no argument should return stats report containing peer-connection stats and outbound-track-stats] + expected: FAIL + + [getStats() with track associated with both sender and receiver should reject with InvalidAccessError] + expected: FAIL + + [getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats] + expected: FAIL + + [getStats() with no argument should return stats for no-stream tracks] + expected: FAIL + + [getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats] + expected: FAIL + + [getStats() with no argument should succeed] + expected: FAIL + + [getStats(null) should succeed] + expected: FAIL + + [getStats() with track not added to connection should reject with InvalidAccessError] + expected: FAIL + + [getStats() with no argument should return stats report containing peer-connection stats on an empty PC] + expected: FAIL + + [getStats() with track added via addTransceiver should succeed] + expected: FAIL + + [getStats() with track associated with more than one sender should reject with InvalidAccessError] + expected: FAIL + + [getStats() with track added via addTrack should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getTransceivers.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getTransceivers.html.ini new file mode 100644 index 000000000000..163d418cffa6 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getTransceivers.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-getTransceivers.html] + [Initial peer connection should have list of zero senders, receivers and transceivers] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini new file mode 100644 index 000000000000..5914c3ea93e2 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-iceConnectionState-disconnected.https.html] + [ICE goes to disconnected if the other side goes away] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini new file mode 100644 index 000000000000..a6e68735fbfd --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-iceConnectionState.https.html] + [Initial iceConnectionState should be new] + expected: FAIL + + [ICE can connect in a recvonly usecase] + expected: FAIL + + [Closing the connection should set iceConnectionState to closed] + expected: FAIL + + [connection with one data channel should eventually have connected connection state] + expected: FAIL + + [connection with one data channel should eventually have connected or completed connection state] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini new file mode 100644 index 000000000000..3423f87aabb0 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-iceGatheringState.html] + [iceGatheringState should eventually become complete after setLocalDescription] + expected: FAIL + + [Initial iceGatheringState should be new] + expected: FAIL + + [connection with one data channel should eventually have connected connection state] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-ondatachannel.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-ondatachannel.html.ini new file mode 100644 index 000000000000..f813f6444e18 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-ondatachannel.html.ini @@ -0,0 +1,25 @@ +[RTCPeerConnection-ondatachannel.html] + [Open event should be raised when closing the channel in the datachannel event after enqueuing a task] + expected: FAIL + + [Data channel event should fire when new data channel is announced to the remote peer] + expected: FAIL + + [Negotiated channel should not fire datachannel event on remote peer] + expected: FAIL + + [In-band negotiated channel created on remote peer should match the same (default) configuration as local peer] + expected: FAIL + + [Open event should not be raised when sending and immediately closing the channel in the datachannel event] + expected: FAIL + + [Open event should not be raised when closing the channel in the datachannel event] + expected: FAIL + + [Should be able to send data in a datachannel event handler] + expected: FAIL + + [In-band negotiated channel created on remote peer should match the same configuration as local peer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini new file mode 100644 index 000000000000..124c16268d9c --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -0,0 +1,38 @@ +[RTCPeerConnection-onnegotiationneeded.html] + expected: ERROR + [negotiationneeded event should not fire if signaling state is not stable] + expected: FAIL + + [addTransceiver() should fire negotiationneeded event] + expected: FAIL + + [removeTrack should cause negotiationneeded to fire on the caller] + expected: TIMEOUT + + [Calling both addTransceiver() and createDataChannel() should fire negotiationneeded event once] + expected: FAIL + + [negotiationneeded event should fire only after signaling state go back to stable after setRemoteDescription] + expected: FAIL + + [Calling addTransceiver() twice should fire negotiationneeded event once] + expected: FAIL + + [Updating the direction of the transceiver should cause negotiationneeded to fire] + expected: FAIL + + [negotiationneeded event should fire only after signaling state go back to stable after setLocalDescription] + expected: FAIL + + [removeTrack should cause negotiationneeded to fire on the callee] + expected: TIMEOUT + + [Creating first data channel should fire negotiationneeded event] + expected: FAIL + + [addTrack should cause negotiationneeded to fire] + expected: FAIL + + [calling createDataChannel twice should fire negotiationneeded event once] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini new file mode 100644 index 000000000000..d564904adfb2 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-onsignalingstatechanged.https.html] + [RTCPeerConnection onsignalingstatechanged] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini new file mode 100644 index 000000000000..f911ff8026ec --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-ontrack.https.html] + [addTrack() should cause remote connection to fire ontrack when setRemoteDescription()] + expected: FAIL + + [setRemoteDescription() with m= line of recvonly direction should not trigger track event] + expected: FAIL + + [setRemoteDescription should trigger ontrack event when the MSID of the stream is is parsed.] + expected: FAIL + + [addTransceiver('video') should cause remote connection to fire ontrack when setRemoteDescription()] + expected: FAIL + + [addTransceiver() with inactive direction should not cause remote connection to fire ontrack when setRemoteDescription()] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini new file mode 100644 index 000000000000..b8cc12e7e538 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -0,0 +1,13 @@ +[RTCPeerConnection-remote-track-mute.https.html] + [Changing transceiver direction to 'sendrecv' unmutes the remote track] + expected: FAIL + + [pc.close() mutes remote tracks] + expected: FAIL + + [Changing transceiver direction to 'inactive' mutes the remote track] + expected: FAIL + + [ontrack: track goes from muted to unmuted] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini new file mode 100644 index 000000000000..03f54a272326 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini @@ -0,0 +1,40 @@ +[RTCPeerConnection-removeTrack.https.html] + [Calling removeTrack with currentDirection sendonly should set direction to inactive] + expected: FAIL + + [addTrack - Calling removeTrack when connection is closed should throw InvalidStateError] + expected: FAIL + + [addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError] + expected: FAIL + + [addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError] + expected: FAIL + + [addTransceiver - Calling removeTrack on different connection that is closed should throw InvalidStateError] + expected: FAIL + + [addTransceiver - Calling removeTrack with valid sender should set sender.track to null] + expected: FAIL + + [Calling removeTrack on a stopped transceiver should be a no-op] + expected: FAIL + + [addTransceiver - Calling removeTrack when connection is closed should throw InvalidStateError] + expected: FAIL + + [addTrack - Calling removeTrack on different connection should throw InvalidAccessError] + expected: FAIL + + [Calling removeTrack with currentDirection inactive should not change direction] + expected: FAIL + + [Calling removeTrack with currentDirection sendrecv should set direction to recvonly] + expected: FAIL + + [Calling removeTrack with currentDirection recvonly should not change direction] + expected: FAIL + + [addTrack - Calling removeTrack with valid sender should set sender.track to null] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini new file mode 100644 index 000000000000..2eccaf1bedf5 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setDescription-transceiver.html.ini @@ -0,0 +1,19 @@ +[RTCPeerConnection-setDescription-transceiver.html] + [setRemoteDescription(rollback) should remove newly created transceiver from transceiver list] + expected: FAIL + + [setRemoteDescription(offer) with m= section and no existing transceiver should create corresponding transceiver] + expected: FAIL + + [setLocalDescription(offer) with m= section should assign mid to corresponding transceiver] + expected: FAIL + + [setLocalDescription(rollback) should unset transceiver.mid] + expected: FAIL + + [setRemoteDescription should stop the transceiver if its corresponding m section is rejected] + expected: FAIL + + [setLocalDescription(rollback) should only unset transceiver mids associated with current round] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini new file mode 100644 index 000000000000..7d6d3f228390 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini @@ -0,0 +1,19 @@ +[RTCPeerConnection-setLocalDescription-answer.html] + [Setting previously generated answer after a call to createOffer should work] + expected: FAIL + + [Calling setLocalDescription(answer) from have-local-offer state should reject with InvalidModificationError] + expected: FAIL + + [setLocalDescription() with type answer and null sdp should use lastAnswer generated from createAnswer] + expected: FAIL + + [Calling setLocalDescription(answer) from stable state should reject with InvalidModificationError] + expected: FAIL + + [setLocalDescription() with answer not created by own createAnswer() should reject with InvalidModificationError] + expected: FAIL + + [setLocalDescription() with valid answer should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini new file mode 100644 index 000000000000..415598909b2a --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini @@ -0,0 +1,19 @@ +[RTCPeerConnection-setLocalDescription-offer.html] + [Set created offer other than last offer should reject with InvalidModificationError] + expected: FAIL + + [setLocalDescription() with offer not created by own createOffer() should reject with InvalidModificationError] + expected: FAIL + + [setLocalDescription with valid offer should succeed] + expected: FAIL + + [Setting previously generated offer after a call to createAnswer should work] + expected: FAIL + + [setLocalDescription with type offer and null sdp should use lastOffer generated from createOffer] + expected: FAIL + + [Creating and setting offer multiple times should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini new file mode 100644 index 000000000000..f3c47f23515a --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini @@ -0,0 +1,13 @@ +[RTCPeerConnection-setLocalDescription-pranswer.html] + [setLocalDescription(pranswer) should succeed] + expected: FAIL + + [setLocalDescription(pranswer) from stable state should reject with InvalidStateError] + expected: FAIL + + [setLocalDescription(answer) from have-local-pranswer state should succeed] + expected: FAIL + + [setLocalDescription(pranswer) can be applied multiple times while still in have-local-pranswer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini new file mode 100644 index 000000000000..0f1c2de4941b --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini @@ -0,0 +1,13 @@ +[RTCPeerConnection-setLocalDescription-rollback.html] + [setLocalDescription(rollback) after setting answer description should reject with InvalidStateError] + expected: FAIL + + [setLocalDescription(rollback) should ignore invalid sdp content and succeed] + expected: FAIL + + [setLocalDescription(rollback) from stable state should reject with InvalidStateError] + expected: FAIL + + [setLocalDescription(rollback) from have-local-offer state should reset back to stable state] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini new file mode 100644 index 000000000000..80d16956b1ac --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-setLocalDescription.html] + [Switching role from answerer to offerer after going back to stable state should succeed] + expected: FAIL + + [onsignalingstatechange fires before setLocalDescription resolves] + expected: FAIL + + [Calling createOffer() and setLocalDescription() again after one round of local-offer/remote-answer should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini new file mode 100644 index 000000000000..90f8e96b5135 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-answer.html.ini @@ -0,0 +1,10 @@ +[RTCPeerConnection-setRemoteDescription-answer.html] + [Calling setRemoteDescription(answer) from stable state should reject with InvalidStateError] + expected: FAIL + + [Calling setRemoteDescription(answer) from have-remote-offer state should reject with InvalidStateError] + expected: FAIL + + [setRemoteDescription() with valid state and answer should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini new file mode 100644 index 000000000000..a26b4f69f8ba --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-setRemoteDescription-nomsid.html] + [setRemoteDescription with an SDP without a=msid lines triggers ontrack with a default stream.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini new file mode 100644 index 000000000000..46d9fc38844b --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-offer.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-setRemoteDescription-offer.html] + [setRemoteDescription multiple times should succeed] + expected: FAIL + + [setRemoteDescription multiple times with different offer should succeed] + expected: FAIL + + [setRemoteDescription(offer) with invalid SDP should reject with RTCError] + expected: FAIL + + [setRemoteDescription(offer) from have-local-offer state should reject with InvalidStateError] + expected: FAIL + + [setRemoteDescription with valid offer should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini new file mode 100644 index 000000000000..830a3ac69d30 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini @@ -0,0 +1,13 @@ +[RTCPeerConnection-setRemoteDescription-pranswer.html] + [setRemoteDescription(answer) from have-remote-pranswer state should succeed] + expected: FAIL + + [setRemoteDescription(pranswer) multiple times should succeed] + expected: FAIL + + [setRemoteDescription(pranswer) from stable state should reject with InvalidStateError] + expected: FAIL + + [setRemoteDescription(pranswer) from have-local-offer state should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini new file mode 100644 index 000000000000..ed1103294b00 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-replaceTrack.https.html.ini @@ -0,0 +1,19 @@ +[RTCPeerConnection-setRemoteDescription-replaceTrack.https.html] + [replaceTrack() sets the track attribute to null.] + expected: FAIL + + [replaceTrack() rejects when the peer connection is closed.] + expected: FAIL + + [replaceTrack() sets the track attribute to a new track.] + expected: FAIL + + [replaceTrack() does not reject after a subsequent removeTrack().] + expected: FAIL + + [replaceTrack() does not reject when invoked after removeTrack().] + expected: FAIL + + [replaceTrack() does not set the track synchronously.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini new file mode 100644 index 000000000000..7d8eb7285488 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-setRemoteDescription-rollback.html] + [setRemoteDescription(rollback) from stable state should reject with InvalidStateError] + expected: FAIL + + [setRemoteDescription(rollback) should ignore invalid sdp content and succeed] + expected: FAIL + + [local offer created before setRemoteDescription(remote offer) then rollback should still be usable] + expected: FAIL + + [setRemoteDescription(rollback) in have-remote-offer state should revert to stable state] + expected: FAIL + + [local offer created before setRemoteDescription(remote offer) with different transceiver level assignments then rollback should still be usable] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini new file mode 100644 index 000000000000..a8cb104dff57 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -0,0 +1,43 @@ +[RTCPeerConnection-setRemoteDescription-tracks.https.html] + [ontrack fires before setRemoteDescription resolves.] + expected: FAIL + + [addTrack() for an existing stream makes stream.onaddtrack fire.] + expected: FAIL + + [ontrack's receiver matches getReceivers().] + expected: FAIL + + [track.onmute fires before setRemoteDescription resolves.] + expected: FAIL + + [addTrack() with two tracks and one stream makes ontrack fire twice with the tracks and shared stream.] + expected: FAIL + + [stream.onaddtrack fires before setRemoteDescription resolves.] + expected: FAIL + + [addTrack() with a track and no stream makes ontrack fire with a track and no stream.] + expected: FAIL + + [addTrack() with a track and a stream makes ontrack fire with a track and a stream.] + expected: FAIL + + [removeTrack() makes track.onmute fire and the track to be muted.] + expected: FAIL + + [addTrack() with a track and two streams makes ontrack fire with a track and two streams.] + expected: FAIL + + [stream.onremovetrack fires before setRemoteDescription resolves.] + expected: FAIL + + [removeTrack() makes stream.onremovetrack fire and the track to be removed from the stream.] + expected: FAIL + + [removeTrack() does not remove the receiver.] + expected: FAIL + + [removeTrack() twice is safe.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini new file mode 100644 index 000000000000..5ec70d9a9320 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini @@ -0,0 +1,16 @@ +[RTCPeerConnection-setRemoteDescription.html] + [Calling setRemoteDescription() again after one round of remote-offer/local-answer should succeed] + expected: FAIL + + [Switching role from offerer to answerer after going back to stable state should succeed] + expected: FAIL + + [Negotiation should fire signalingsstate events] + expected: FAIL + + [setRemoteDescription() with invalid SDP and stable state should reject with InvalidStateError] + expected: FAIL + + [setRemoteDescription with invalid type and invalid SDP should reject with TypeError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini new file mode 100644 index 000000000000..cfca4a6ac221 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini @@ -0,0 +1,55 @@ +[RTCPeerConnection-track-stats.https.html] + [replaceTrack() before offer: new track attachment stats present] + expected: FAIL + + [replaceTrack() after offer, before answer: new track attachment stats present] + expected: FAIL + + [addTrack() with setLocalDescription() yields media stream stats] + expected: FAIL + + [RTCRtpSender.getStats() contains only outbound-rtp and related stats] + expected: FAIL + + [RTCRtpReceiver.getStats() contains only inbound-rtp and related stats] + expected: FAIL + + [O/A exchange yields inbound RTP stream stats for receiving track] + expected: FAIL + + [replaceTrack(): original track attachment stats present after replacing] + expected: FAIL + + [Media stream stats references track stats] + expected: FAIL + + [O/A exchange yields outbound RTP stream stats for sending track] + expected: FAIL + + [addTrack(): Media stream stats references track stats] + expected: FAIL + + [addTrack() without setLocalDescription() yields track stats] + expected: FAIL + + [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are zero senders or receivers for the track] + expected: FAIL + + [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are multiple senders for the track] + expected: FAIL + + [replaceTrack() after answer: new track attachment stats present] + expected: FAIL + + [RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats()] + expected: FAIL + + [RTCPeerConnection.getStats(sendingTrack) is the same as RTCRtpSender.getStats()] + expected: FAIL + + [addTrack() with setLocalDescription() yields track stats] + expected: FAIL + + [addTrack() without setLocalDescription() yields media stream stats] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini new file mode 100644 index 000000000000..f630372bc8f0 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -0,0 +1,133 @@ +[RTCPeerConnection-transceivers.https.html] + [setLocalDescription(answer): transceiver.currentDirection is recvonly] + expected: FAIL + + [setRemoteDescription(offer): ontrack fires with a track] + expected: FAIL + + [Can setup two-way call using a single transceiver] + expected: FAIL + + [setRemoteDescription(offer): transceiver.mid is the same on both ends] + expected: FAIL + + [addTransceiver(track, init): initialize sendEncodings[0\].active to false] + expected: FAIL + + [addTransceiver('video'): transceiver.receiver.track.kind == 'video'] + expected: FAIL + + [addTransceiver('audio'): transceiver.stopped is false] + expected: FAIL + + [setLocalDescription(answer): transceiver.currentDirection is sendonly] + expected: FAIL + + [addTrack: transceiver is not associated with an m-section] + expected: FAIL + + [transceiver.sender.track does not revert to an old state] + expected: FAIL + + [addTrack: transceiver.receiver has its own track] + expected: FAIL + + [addTrack: "transceiver == {sender,receiver}"] + expected: FAIL + + [setLocalDescription(offer): transceiver.mid matches the offer SDP] + expected: FAIL + + [setRemoteDescription(offer): transceiver.direction is recvonly] + expected: FAIL + + [Changing transceiver direction to 'sendrecv' makes ontrack fire] + expected: FAIL + + [addTrack: transceiver is not stopped] + expected: FAIL + + [addTrack(1 stream): ontrack fires with corresponding stream] + expected: FAIL + + [addTransceiver(track, init): initialize direction to inactive] + expected: FAIL + + [addTrack(0 streams): ontrack fires with no stream] + expected: FAIL + + [addTrack: transceiver's direction is sendrecv] + expected: FAIL + + [addTrack(2 streams): ontrack fires with corresponding two streams] + expected: FAIL + + [addTransceiver(0 streams): ontrack fires with no stream] + expected: FAIL + + [addTransceiver(1 stream): ontrack fires with corresponding stream] + expected: FAIL + + [setRemoteDescription(offer): transceiver.currentDirection is null] + expected: FAIL + + [addTrack: transceiver.sender is associated with the track] + expected: FAIL + + [addTransceiver does not reuse reusable transceivers] + expected: FAIL + + [addTransceiver(track): "transceiver == {sender,receiver}"] + expected: FAIL + + [transceiver.direction does not revert to an old state] + expected: FAIL + + [addTransceiver('audio'): transceiver.sender.track == null] + expected: FAIL + + [addTransceiver('audio'): creates a transceiver with direction sendrecv] + expected: FAIL + + [setLocalDescription(offer): transceiver gets associated with an m-section] + expected: FAIL + + [addTransceiver('audio'): transceiver.currentDirection is null] + expected: FAIL + + [setRemoteDescription(offer): ontrack's stream.id is the same as stream.id] + expected: FAIL + + [setRemoteDescription(offer): transceiver.stopped is false] + expected: FAIL + + [addTransceiver('audio'): transceiver.receiver.track.kind == 'audio'] + expected: FAIL + + [addTransceiver(track): creates a transceiver for the track] + expected: FAIL + + [addTransceiver(2 streams): ontrack fires with corresponding two streams] + expected: FAIL + + [setRemoteDescription(offer): "transceiver == {sender,receiver}"] + expected: FAIL + + [addTrack: transceiver's currentDirection is null] + expected: FAIL + + [addTrack reuses reusable transceivers] + expected: FAIL + + [addTrack: creates a transceiver for the sender] + expected: FAIL + + [Closing the PC stops the transceivers] + expected: FAIL + + [setRemoteDescription(offer): ontrack fires with a transceiver.] + expected: FAIL + + [addTrack: transceiver.receiver's track is muted] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini new file mode 100644 index 000000000000..dfb88485f887 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini @@ -0,0 +1,25 @@ +[RTCPeerConnectionIceEvent-constructor.html] + [RTCPeerConnectionIceEvent.candidate is null when constructed with { candidate: null }] + expected: FAIL + + [RTCPeerConnectionIceEvent with empty object as eventInitDict (default)] + expected: FAIL + + [RTCPeerConnectionIceEvent.candidate is null when constructed with { candidate: undefined }] + expected: FAIL + + [RTCPeerConnectionIceEvent with RTCIceCandidate] + expected: FAIL + + [RTCPeerConnectionIceEvent with non RTCIceCandidate object throws] + expected: FAIL + + [RTCPeerConnectionIceEvent with no arguments throws TypeError] + expected: FAIL + + [RTCPeerConnectionIceEvent bubbles and cancelable] + expected: FAIL + + [RTCPeerConnectionIceEvent with no eventInitDict (default)] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-codecs.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-codecs.html.ini new file mode 100644 index 000000000000..b5836224c54e --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-codecs.html.ini @@ -0,0 +1,19 @@ +[RTCRtpParameters-codecs.html] + [setParameters() with codec.mimeType modified should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with new codecs inserted should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with codec.clockRate modified should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with codec.channels modified should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with codec.payloadType modified should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with codec.sdpFmtpLine modified should reject with InvalidModificationError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-degradationPreference.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-degradationPreference.html.ini new file mode 100644 index 000000000000..aa9e7a414a3f --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-degradationPreference.html.ini @@ -0,0 +1,7 @@ +[RTCRtpParameters-degradationPreference.html] + [setParameters with degradationPreference unset should succeed] + expected: FAIL + + [setParameters with degradationPreference set should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-encodings.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-encodings.html.ini new file mode 100644 index 000000000000..c0417560aeec --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-encodings.html.ini @@ -0,0 +1,73 @@ +[RTCRtpParameters-encodings.html] + [setParameters() with modified encoding.dtx should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [addTransceiver() with undefined sendEncodings should have default encoding parameter with active set to true] + expected: FAIL + + [setParameters() with unset encoding.dtx should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.priority should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [sender.getParameters() should return sendEncodings set by addTransceiver()] + expected: FAIL + + [setParameters() with modified encoding.priority should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.ptime should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.dtx should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with encoding.scaleResolutionDownBy field set to greater than 1.0 should succeed] + expected: FAIL + + [setParameters() with encoding.scaleResolutionDownBy field set to less than 1.0 should reject with RangeError] + expected: FAIL + + [setParameters() with unset encoding.dtx should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.maxFramerate should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [sender.setParameters() with mismatch number of encodings should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with modified encoding.rid field should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with modified encoding.maxFramerate should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.networkPriority should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.networkPriority should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.ptime should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.maxBitrate should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.maxBitrate should succeed without RTCRtpTransceiverInit] + expected: FAIL + + [addTransceiver() with empty list sendEncodings should have default encoding parameter with active set to true] + expected: FAIL + + [sender.setParameters() with encodings unset should reject with TypeError] + expected: FAIL + + [setParameters() with modified encoding.active should succeed with RTCRtpTransceiverInit] + expected: FAIL + + [setParameters() with modified encoding.active should succeed without RTCRtpTransceiverInit] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-headerExtensions.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-headerExtensions.html.ini new file mode 100644 index 000000000000..6ca8a5f33e62 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-headerExtensions.html.ini @@ -0,0 +1,4 @@ +[RTCRtpParameters-headerExtensions.html] + [setParameters() with modified headerExtensions should reject with InvalidModificationError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-rtcp.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-rtcp.html.ini new file mode 100644 index 000000000000..f64f1632239e --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-rtcp.html.ini @@ -0,0 +1,7 @@ +[RTCRtpParameters-rtcp.html] + [setParameters() with modified rtcp.cname should reject with InvalidModificationError] + expected: FAIL + + [setParameters() with modified rtcp.reducedSize should reject with InvalidModificationError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpParameters-transactionId.html.ini b/tests/wpt/metadata/webrtc/RTCRtpParameters-transactionId.html.ini new file mode 100644 index 000000000000..789937148d22 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpParameters-transactionId.html.ini @@ -0,0 +1,16 @@ +[RTCRtpParameters-transactionId.html] + [sender.setParameters() with transaction ID different from last getParameters() should reject with InvalidModificationError] + expected: FAIL + + [setParameters() twice with the same parameters should reject with InvalidStateError] + expected: FAIL + + [setParameters() with parameters older than last getParameters() should reject with InvalidModificationError] + expected: FAIL + + [sender.getParameters() should return different transaction IDs for each call] + expected: FAIL + + [sender.setParameters() with transaction ID unset should reject with TypeError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getCapabilities.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getCapabilities.html.ini new file mode 100644 index 000000000000..414165d0f605 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getCapabilities.html.ini @@ -0,0 +1,10 @@ +[RTCRtpReceiver-getCapabilities.html] + [RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary] + expected: FAIL + + [RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary] + expected: FAIL + + [RTCRtpSender.getCapabilities('dummy') should return null] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini new file mode 100644 index 000000000000..ff336963390c --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini @@ -0,0 +1,7 @@ +[RTCRtpReceiver-getContributingSources.https.html] + [[audio\] getContributingSources() returns an empty list in loopback call] + expected: FAIL + + [[video\] getContributingSources() returns an empty list in loopback call] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getParameters.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getParameters.html.ini new file mode 100644 index 000000000000..b282182bb8be --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getParameters.html.ini @@ -0,0 +1,10 @@ +[RTCRtpReceiver-getParameters.html] + [getParameters() with simulcast video receiver] + expected: FAIL + + [getParameters() with video receiver] + expected: FAIL + + [getParameters() with audio receiver] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini new file mode 100644 index 000000000000..a69a3eda57bd --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -0,0 +1,7 @@ +[RTCRtpReceiver-getStats.https.html] + [receiver.getStats() via addTrack should return stats report containing inbound-rtp stats] + expected: FAIL + + [receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini new file mode 100644 index 000000000000..a5a15242ac4c --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -0,0 +1,37 @@ +[RTCRtpReceiver-getSynchronizationSources.https.html] + [[audio\] RTCRtpSynchronizationSource.source is a number] + expected: FAIL + + [[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] + expected: FAIL + + [[audio\] getSynchronizationSources() eventually returns a non-empty list] + expected: FAIL + + [[video\] RTCRtpSynchronizationSource.timestamp is a number] + expected: FAIL + + [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean] + expected: FAIL + + [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] + expected: FAIL + + [[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] + expected: FAIL + + [[video\] getSynchronizationSources() eventually returns a non-empty list] + expected: FAIL + + [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] + expected: FAIL + + [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]] + expected: FAIL + + [[video\] RTCRtpSynchronizationSource.source is a number] + expected: FAIL + + [[audio\] RTCRtpSynchronizationSource.timestamp is a number] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-getCapabilities.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-getCapabilities.html.ini new file mode 100644 index 000000000000..b70af48747f3 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-getCapabilities.html.ini @@ -0,0 +1,10 @@ +[RTCRtpSender-getCapabilities.html] + [RTCRtpSender.getCapabilities('video') should return RTCRtpCapabilities dictionary] + expected: FAIL + + [RTCRtpSender.getCapabilities('audio') should return RTCRtpCapabilities dictionary] + expected: FAIL + + [RTCRtpSender.getCapabilities('dummy') should return null] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini new file mode 100644 index 000000000000..4256534ee3e2 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini @@ -0,0 +1,7 @@ +[RTCRtpSender-getStats.https.html] + [sender.getStats() via addTrack should return stats report containing outbound-rtp stats] + expected: FAIL + + [sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini new file mode 100644 index 000000000000..821925718a1d --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini @@ -0,0 +1,28 @@ +[RTCRtpSender-replaceTrack.https.html] + [Calling replaceTrack on sender with null track and not set to session description should resolve with sender.track set to given track] + expected: FAIL + + [Calling replaceTrack(null) on sender not set to session description should resolve with sender.track set to null] + expected: FAIL + + [Calling replaceTrack with track of different kind should reject with TypeError] + expected: FAIL + + [Calling replaceTrack on sender with similar track and and set to session description should resolve with sender.track set to new track] + expected: FAIL + + [Calling replaceTrack on closed connection should reject with InvalidStateError] + expected: FAIL + + [Calling replaceTrack on sender with stopped track and and set to session description should resolve with sender.track set to given track] + expected: FAIL + + [Calling replaceTrack on stopped sender should reject with InvalidStateError] + expected: FAIL + + [Calling replaceTrack on sender not set to session description should resolve with sender.track set to given track] + expected: FAIL + + [Calling replaceTrack(null) on sender set to session description should resolve with sender.track set to null] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-setParameters.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-setParameters.html.ini new file mode 100644 index 000000000000..c5eebcb0af55 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-setParameters.html.ini @@ -0,0 +1,4 @@ +[RTCRtpSender-setParameters.html] + [setParameters() when transceiver is stopped should reject with InvalidStateError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini new file mode 100644 index 000000000000..fac8652e7aa7 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini @@ -0,0 +1,16 @@ +[RTCRtpSender-transport.https.html] + [RTCRtpSender/receiver.transport at the right time, with bundle policy balanced] + expected: FAIL + + [RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle] + expected: FAIL + + [RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat] + expected: FAIL + + [RTCRtpSender/receiver.transport has a value when connected] + expected: FAIL + + [RTCRtpSender.transport is null when unconnected] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver-direction.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-direction.html.ini new file mode 100644 index 000000000000..0fe61cf3a652 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-direction.html.ini @@ -0,0 +1,10 @@ +[RTCRtpTransceiver-direction.html] + [setting direction with same direction should have no effect] + expected: FAIL + + [setting direction should change transceiver.direction] + expected: FAIL + + [setting direction should change transceiver.direction independent of transceiver.currentDirection] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini new file mode 100644 index 000000000000..e8cfff091f01 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini @@ -0,0 +1,28 @@ +[RTCRtpTransceiver-setCodecPreferences.html] + [setCodecPreferences() with modified codecs returned from getCapabilities() should throw InvalidAccessError] + expected: FAIL + + [setCodecPreferences() with user defined codec together with codecs returned from getCapabilities() should throw InvalidAccessError] + expected: FAIL + + [setCodecPreferences([\]) should succeed] + expected: FAIL + + [setCodecPreferences() on video transceiver with codecs returned from RTCRtpReceiver.getCapabilities('video') should succeed] + expected: FAIL + + [setCodecPreferences() with user defined codec should throw InvalidAccessError] + expected: FAIL + + [setCodecPreferences() on audio transceiver with codecs returned from getCapabilities('video') should throw InvalidAccessError] + expected: FAIL + + [setCodecPreferences() with reordered codecs should succeed] + expected: FAIL + + [setCodecPreferences() with both sender receiver codecs combined should succeed] + expected: FAIL + + [setCodecPreferences() on audio transceiver with codecs returned from RTCRtpSender.getCapabilities('audio') should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver-stop.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-stop.html.ini new file mode 100644 index 000000000000..9596a7c3f469 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-stop.html.ini @@ -0,0 +1,13 @@ +[RTCRtpTransceiver-stop.html] + [A stopped sendonly transceiver should generate a sendonly m-section in the offer] + expected: FAIL + + [During renegotiation, adding and stopping a transceiver should not trigger a renegotiated offer m-section generation] + expected: FAIL + + [A transceiver added and stopped before the initial offer generation should not trigger an offer m-section generation] + expected: FAIL + + [A stopped inactive transceiver should generate an inactive m-section in the offer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini new file mode 100644 index 000000000000..c65b2f802864 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -0,0 +1,109 @@ +[RTCRtpTransceiver.https.html] + [checkAddTransceiverWithTrack] + expected: FAIL + + [checkNoMidAnswer] + expected: FAIL + + [checkSetDirection] + expected: FAIL + + [checkCurrentDirection] + expected: FAIL + + [checkReplaceTrackNullDoesntPreventPairing] + expected: FAIL + + [checkAddTrackExistingTransceiverThenRemove] + expected: FAIL + + [checkMsectionReuse] + expected: FAIL + + [checkRemoteRollback] + expected: FAIL + + [checkAddTransceiverNoTrackDoesntPair] + expected: FAIL + + [checkRemoveTrackNegotiation] + expected: FAIL + + [checkStopAfterClose] + expected: FAIL + + [checkNoMidOffer] + expected: FAIL + + [checkStopAfterCreateOffer] + expected: FAIL + + [checkAddTransceiverThenAddTrackPairs] + expected: FAIL + + [checkLocalRollback] + expected: FAIL + + [checkSendrecvWithTracklessStream] + expected: FAIL + + [checkMsidNoTrackId] + expected: FAIL + + [checkAddTransceiverNoTrack] + expected: FAIL + + [checkAddTransceiverBadKind] + expected: FAIL + + [checkStopAfterCreateAnswer] + expected: FAIL + + [checkAddTransceiverWithDirection] + expected: FAIL + + [checkAddTrackPairs] + expected: FAIL + + [checkStopAfterSetLocalAnswer] + expected: FAIL + + [checkStopAfterSetRemoteOffer] + expected: FAIL + + [checkStopAfterCreateOfferWithReusedMsection] + expected: FAIL + + [checkMute] + expected: FAIL + + [checkSendrecvWithNoSendTrack] + expected: FAIL + + [checkAddTransceiverWithSetRemoteOfferSending] + expected: FAIL + + [checkAddTransceiverWithAddTrack] + expected: FAIL + + [checkAddTransceiverWithSetRemoteOfferNoSend] + expected: FAIL + + [checkRemoveAndReadd] + expected: FAIL + + [checkStopAfterSetLocalOffer] + expected: FAIL + + [checkAddTransceiverWithTrackDoesntPair] + expected: FAIL + + [checkAddTransceiverThenReplaceTrackDoesntPair] + expected: FAIL + + [checkRollbackAndSetRemoteOfferWithDifferentType] + expected: FAIL + + [checkStop] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCSctpTransport-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCSctpTransport-constructor.html.ini new file mode 100644 index 000000000000..7c6091433f47 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCSctpTransport-constructor.html.ini @@ -0,0 +1,13 @@ +[RTCSctpTransport-constructor.html] + [setRemoteDescription() with answer containing data media should initialize pc.sctp] + expected: FAIL + + [setLocalDescription() with answer containing data media should initialize pc.sctp] + expected: FAIL + + [setLocalDescription() with answer not containing data media should not initialize pc.sctp] + expected: FAIL + + [setRemoteDescription() with answer not containing data media should not initialize pc.sctp] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCSctpTransport-events.html.ini b/tests/wpt/metadata/webrtc/RTCSctpTransport-events.html.ini new file mode 100644 index 000000000000..25121409be6a --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCSctpTransport-events.html.ini @@ -0,0 +1,7 @@ +[RTCSctpTransport-events.html] + [SctpTransport reaches connected and closed state] + expected: FAIL + + [SctpTransport objects are created at appropriate times] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCSctpTransport-maxMessageSize.html.ini b/tests/wpt/metadata/webrtc/RTCSctpTransport-maxMessageSize.html.ini new file mode 100644 index 000000000000..04a99bd0fea9 --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCSctpTransport-maxMessageSize.html.ini @@ -0,0 +1,16 @@ +[RTCSctpTransport-maxMessageSize.html] + [max-message-size with a (non-zero) value provided by the remote peer] + expected: FAIL + + [Determine the local side send limitation (canSendSize) by offering a max-message-size of 0] + expected: FAIL + + [Remote offer SDP missing max-message-size attribute] + expected: FAIL + + [max-message-size with a (non-zero) value larger than canSendSize provided by the remote peer] + expected: FAIL + + [Renegotiate max-message-size with various values provided by the remote peer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCTrackEvent-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCTrackEvent-constructor.html.ini new file mode 100644 index 000000000000..17cb203923ac --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCTrackEvent-constructor.html.ini @@ -0,0 +1,22 @@ +[RTCTrackEvent-constructor.html] + [new RTCTrackEvent() with no transceiver should throw TypeError] + expected: FAIL + + [new RTCTrackEvent() with unrelated receiver, track, streams, transceiver should succeed] + expected: FAIL + + [new RTCTrackEvent() with no track should throw TypeError] + expected: FAIL + + [new RTCTrackEvent() with no receiver should throw TypeError] + expected: FAIL + + [new RTCTrackEvent() with valid receiver, track, multiple streams, transceiver should succeed] + expected: FAIL + + [new RTCTrackEvent() with valid receiver, track, transceiver should succeed] + expected: FAIL + + [new RTCTrackEvent() with valid receiver, track, streams, transceiver should succeed] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini new file mode 100644 index 000000000000..2fa5e125c03e --- /dev/null +++ b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini @@ -0,0 +1,7 @@ +[RTCTrackEvent-fire.html] + [Applying a remote description with removed msid should trigger firing a removetrack event on the corresponding stream] + expected: FAIL + + [Applying a remote description with a new msid should trigger firing an event with populated streams] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/datachannel-emptystring.html.ini b/tests/wpt/metadata/webrtc/datachannel-emptystring.html.ini new file mode 100644 index 000000000000..d3cc2f656efc --- /dev/null +++ b/tests/wpt/metadata/webrtc/datachannel-emptystring.html.ini @@ -0,0 +1,4 @@ +[datachannel-emptystring.html] + [Can send empty strings across a WebRTC data channel.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/getstats.html.ini b/tests/wpt/metadata/webrtc/getstats.html.ini new file mode 100644 index 000000000000..ba30262f2591 --- /dev/null +++ b/tests/wpt/metadata/webrtc/getstats.html.ini @@ -0,0 +1,4 @@ +[getstats.html] + [Can get stats from a basic WebRTC call.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/historical.html.ini b/tests/wpt/metadata/webrtc/historical.html.ini new file mode 100644 index 000000000000..9e0a93addfa3 --- /dev/null +++ b/tests/wpt/metadata/webrtc/historical.html.ini @@ -0,0 +1,37 @@ +[historical.html] + [RTCPeerConnection member removeStream should not exist] + expected: FAIL + + [RTCRtpTransceiver member setDirection should not exist] + expected: FAIL + + [RTCDataChannel member reliable should not exist] + expected: FAIL + + [RTCPeerConnection member getStreamById should not exist] + expected: FAIL + + [RTCPeerConnection member onremovestream should not exist] + expected: FAIL + + [RTCPeerConnection member addStream should not exist] + expected: FAIL + + [RTCPeerConnection member getLocalStreams should not exist] + expected: FAIL + + [RTCPeerConnection member getRemoteStreams should not exist] + expected: FAIL + + [RTCPeerConnection member updateIce should not exist] + expected: FAIL + + [RTCDataChannel member maxRetransmitTime should not exist] + expected: FAIL + + [RTCPeerConnection member onaddstream should not exist] + expected: FAIL + + [RTCPeerConnection member createDTMFSender should not exist] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini new file mode 100644 index 000000000000..0a145552b6e7 --- /dev/null +++ b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini @@ -0,0 +1,1495 @@ +[idlharness.https.window.html] + [RTCPeerConnection interface: attribute signalingState] + expected: FAIL + + [RTCIceCandidate interface: existence and properties of interface object] + expected: FAIL + + [RTCRtpSender interface: attribute track] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "foundation" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: attribute mid] + expected: FAIL + + [RTCPeerConnection interface: operation addTransceiver([object Object\],[object Object\], RTCRtpTransceiverInit)] + expected: FAIL + + [RTCPeerConnection interface: calling addTransceiver([object Object\],[object Object\], RTCRtpTransceiverInit) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCCertificate interface: attribute expires] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences([object Object\])" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: calling createAnswer(RTCAnswerOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getReceivers()" with the proper type] + expected: FAIL + + [RTCDataChannel interface: existence and properties of interface object] + expected: FAIL + + [RTCSessionDescription interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getDefaultIceServers()" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceTransport interface: operation getSelectedCandidatePair()] + expected: FAIL + + [RTCDataChannelEvent interface: new RTCDataChannelEvent('channel', {\n channel: new RTCPeerConnection().createDataChannel('')\n }) must inherit property "channel" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "getParameters()" with the proper type] + expected: FAIL + + [RTCDTMFSender interface: operation insertDTMF(DOMString, unsigned long, unsigned long)] + expected: FAIL + + [RTCDataChannel interface: calling send(Blob) on new RTCPeerConnection().createDataChannel('') with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: operation getTransceivers()] + expected: FAIL + + [RTCDataChannel interface: calling send(USVString) on new RTCPeerConnection().createDataChannel('') with too few arguments must throw TypeError] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(Blob)" with the proper type] + expected: FAIL + + [RTCDtlsTransport must be primary interface of idlTestObjects.dtlsTransport] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "getStats()" with the proper type] + expected: FAIL + + [RTCErrorEvent must be primary interface of new RTCErrorEvent('error')] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute pendingLocalDescription] + expected: FAIL + + [RTCRtpSender interface: attribute transport] + expected: FAIL + + [RTCTrackEvent interface: initTrackEvent() must inherit property "streams" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCError interface: attribute sentAlert] + expected: FAIL + + [RTCDataChannelEvent interface: attribute channel] + expected: FAIL + + [RTCRtpSender interface: existence and properties of interface object] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "iceTransport" with the proper type] + expected: FAIL + + [RTCTrackEvent must be primary interface of initTrackEvent()] + expected: FAIL + + [Stringification of idlTestObjects.certificate] + expected: FAIL + + [RTCPeerConnection interface: calling setLocalDescription(RTCSessionDescriptionInit) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCError interface object name] + expected: FAIL + + [RTCRtpSender interface: operation setStreams(MediaStream)] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: operation getConfiguration()] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: new RTCPeerConnectionIceEvent('ice') must inherit property "candidate" with the proper type] + expected: FAIL + + [RTCIceCandidate must be primary interface of new RTCIceCandidate({ sdpMid: 1 })] + expected: FAIL + + [RTCRtpTransceiver interface: existence and properties of interface object] + expected: FAIL + + [RTCSessionDescription must be primary interface of new RTCSessionDescription({ type: 'offer' })] + expected: FAIL + + [RTCPeerConnection interface: attribute ondatachannel] + expected: FAIL + + [RTCRtpTransceiver interface: attribute stopped] + expected: FAIL + + [RTCRtpReceiver interface: existence and properties of interface object] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "stopped" with the proper type] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "ongatheringstatechange" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "signalingState" with the proper type] + expected: FAIL + + [RTCIceCandidate interface object length] + expected: FAIL + + [RTCDataChannel interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceCandidate interface: attribute candidate] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "ordered" with the proper type] + expected: FAIL + + [RTCCertificate interface: existence and properties of interface object] + expected: FAIL + + [RTCPeerConnection interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCTrackEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "maxPacketLifeTime" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicecandidate" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute component] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "transport" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: operation getContributingSources()] + expected: FAIL + + [Test driver for asyncInitCertificate] + expected: FAIL + + [RTCPeerConnection interface: existence and properties of interface object] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "id" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "direction" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute relatedPort] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "sender" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "track" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onmessage" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback)" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface object length] + expected: FAIL + + [RTCError interface object length] + expected: FAIL + + [Stringification of new RTCPeerConnection().addTransceiver('audio')] + expected: FAIL + + [RTCPeerConnection interface: attribute onstatsended] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "currentDirection" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "maxRetransmits" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "sctp" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: calling setLocalDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpTransceiver interface object name] + expected: FAIL + + [RTCErrorEvent interface: new RTCErrorEvent('error') must inherit property "error" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute errorText] + expected: FAIL + + [RTCPeerConnection interface: operation createOffer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback, RTCOfferOptions)] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "address" with the proper type] + expected: FAIL + + [RTCTrackEvent interface: attribute transceiver] + expected: FAIL + + [RTCRtpSender interface: calling replaceTrack(MediaStreamTrack) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCDataChannel interface: attribute onbufferedamountlow] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [Stringification of new RTCSessionDescription({ type: 'offer' })] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "removeTrack(RTCRtpSender)" with the proper type] + expected: FAIL + + [RTCRtpSender interface: attribute rtcpTransport] + expected: FAIL + + [RTCDTMFSender interface object name] + expected: FAIL + + [RTCStatsReport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onstatsended" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "sdp" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: attribute sdp] + expected: FAIL + + [RTCDTMFSender interface: attribute canInsertDTMF] + expected: FAIL + + [RTCPeerConnection interface: attribute sctp] + expected: FAIL + + [RTCRtpReceiver interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').receiver with too few arguments must throw TypeError] + expected: FAIL + + [RTCSessionDescription interface object name] + expected: FAIL + + [RTCPeerConnection interface object name] + expected: FAIL + + [RTCRtpReceiver interface: operation getSynchronizationSources()] + expected: FAIL + + [RTCStatsEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: attribute onconnectionstatechange] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(ArrayBuffer)" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicegatheringstatechange" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "component" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute bufferedAmountLowThreshold] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addTrack(MediaStreamTrack, MediaStream)" with the proper type] + expected: FAIL + + [RTCIceTransport interface object length] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "transport" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCDataChannel interface: calling send(ArrayBuffer) on new RTCPeerConnection().createDataChannel('') with too few arguments must throw TypeError] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getRemoteParameters()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute iceGatheringState] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getCapabilities(DOMString)" with the proper type] + expected: FAIL + + [RTCDTMFToneChangeEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCTrackEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCRtpTransceiver interface: calling setCodecPreferences([object Object\]) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError] + expected: FAIL + + [RTCDtlsTransport interface object name] + expected: FAIL + + [RTCRtpSender interface: operation setParameters(RTCRtpSendParameters)] + expected: FAIL + + [RTCIceTransport interface: attribute gatheringState] + expected: FAIL + + [RTCSessionDescription interface: default toJSON operation on new RTCSessionDescription({ type: 'offer' })] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCPeerConnection interface: operation createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback)] + expected: FAIL + + [RTCPeerConnection interface: calling createOffer(RTCOfferOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpReceiver interface: operation getCapabilities(DOMString)] + expected: FAIL + + [RTCRtpReceiver interface: operation getStats()] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCRtpTransceiver interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onnegotiationneeded" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute pendingRemoteDescription] + expected: FAIL + + [RTCRtpSender interface: calling setStreams(MediaStream) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "onselectedcandidatepairchange" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "priority" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "protocol" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setRemoteDescription(RTCSessionDescriptionInit)" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createOffer(RTCOfferOptions)" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: attribute direction] + expected: FAIL + + [RTCIceCandidate interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getTransceivers()" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute priority] + expected: FAIL + + [RTCDTMFToneChangeEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceTransport must be primary interface of idlTestObjects.iceTransport] + expected: FAIL + + [RTCError interface: attribute errorDetail] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "remoteDescription" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createAnswer(RTCAnswerOptions)" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedPort" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "usernameFragment" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "generateCertificate(AlgorithmIdentifier)" with the proper type] + expected: FAIL + + [Stringification of new RTCIceCandidate({ sdpMid: 1 })] + expected: FAIL + + [RTCPeerConnection interface: calling addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "tcpType" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: new RTCPeerConnectionIceEvent('ice') must inherit property "url" with the proper type] + expected: FAIL + + [RTCErrorEvent interface object length] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setConfiguration(RTCConfiguration)" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute maxRetransmits] + expected: FAIL + + [RTCRtpSender interface object length] + expected: FAIL + + [RTCIceCandidate interface: toJSON operation on new RTCIceCandidate({ sdpMid: 1 })] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface object length] + expected: FAIL + + [RTCDTMFSender interface: attribute toneBuffer] + expected: FAIL + + [RTCCertificate interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "maxChannels" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onclose" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute onicecandidateerror] + expected: FAIL + + [RTCStatsReport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCTrackEvent interface: attribute track] + expected: FAIL + + [RTCIceCandidate interface object name] + expected: FAIL + + [RTCPeerConnection interface: operation addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback)] + expected: FAIL + + [RTCTrackEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: attribute onnegotiationneeded] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "oniceconnectionstatechange" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: attribute transport] + expected: FAIL + + [RTCDataChannel interface object name] + expected: FAIL + + [RTCPeerConnection interface: operation getSenders()] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "getRemoteCertificates()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getStats(MediaStreamTrack)" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "toJSON()" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCDataChannelEvent interface object name] + expected: FAIL + + [RTCDataChannelEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCIceCandidate interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCRtpSender interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpSender interface: operation getCapabilities(DOMString)] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "errorText" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute negotiated] + expected: FAIL + + [RTCRtpReceiver interface: attribute transport] + expected: FAIL + + [Stringification of new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 });] + expected: FAIL + + [RTCPeerConnection interface: attribute onicecandidate] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalParameters()" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onopen" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getStats()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: operation close()] + expected: FAIL + + [RTCDataChannel interface: attribute protocol] + expected: FAIL + + [RTCPeerConnection interface: operation addTrack(MediaStreamTrack, MediaStream)] + expected: FAIL + + [RTCPeerConnection interface: attribute currentRemoteDescription] + expected: FAIL + + [RTCSctpTransport interface: attribute onstatechange] + expected: FAIL + + [RTCIceTransport interface: attribute state] + expected: FAIL + + [RTCPeerConnectionIceEvent interface object name] + expected: FAIL + + [RTCDataChannel interface: attribute onerror] + expected: FAIL + + [RTCRtpSender interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: attribute localDescription] + expected: FAIL + + [RTCPeerConnection interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: operation createOffer(RTCOfferOptions)] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicecandidateerror" with the proper type] + expected: FAIL + + [RTCStatsReport interface: existence and properties of interface object] + expected: FAIL + + [RTCDataChannel interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addTransceiver([object Object\],[object Object\], RTCRtpTransceiverInit)" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "onstatechange" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute maxPacketLifeTime] + expected: FAIL + + [RTCIceCandidate interface: attribute type] + expected: FAIL + + [RTCTrackEvent interface: attribute receiver] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "port" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: operation createDataChannel(USVString, RTCDataChannelInit)] + expected: FAIL + + [RTCDataChannelEvent must be primary interface of new RTCDataChannelEvent('channel', {\n channel: new RTCPeerConnection().createDataChannel('')\n })] + expected: FAIL + + [RTCRtpSender must be primary interface of new RTCPeerConnection().addTransceiver('audio').sender] + expected: FAIL + + [RTCSctpTransport interface object name] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onbufferedamountlow" with the proper type] + expected: FAIL + + [RTCSessionDescription interface object length] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "mid" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createDataChannel(USVString, RTCDataChannelInit)" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute onmessage] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface object name] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCRtpTransceiver interface: attribute sender] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "protocol" with the proper type] + expected: FAIL + + [RTCStatsEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "toJSON()" with the proper type] + expected: FAIL + + [RTCStatsReport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "bufferedAmountLowThreshold" with the proper type] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [Stringification of idlTestObjects.dtlsTransport] + expected: FAIL + + [RTCPeerConnectionIceEvent interface object length] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type] + expected: FAIL + + [Stringification of new RTCDataChannelEvent('channel', {\n channel: new RTCPeerConnection().createDataChannel('')\n })] + expected: FAIL + + [RTCCertificate interface: idlTestObjects.certificate must inherit property "getFingerprints()" with the proper type] + expected: FAIL + + [RTCCertificate must be primary interface of idlTestObjects.certificate] + expected: FAIL + + [RTCCertificate interface object length] + expected: FAIL + + [RTCRtpReceiver must be primary interface of new RTCPeerConnection().addTransceiver('audio').receiver] + expected: FAIL + + [RTCTrackEvent interface: initTrackEvent() must inherit property "transceiver" with the proper type] + expected: FAIL + + [RTCDTMFSender interface object length] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "close()" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute address] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "readyState" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "close()" with the proper type] + expected: FAIL + + [RTCSctpTransport interface: existence and properties of interface object] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "canTrickleIceCandidates" with the proper type] + expected: FAIL + + [RTCTrackEvent interface: initTrackEvent() must inherit property "track" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getSelectedCandidatePair()" with the proper type] + expected: FAIL + + [Stringification of new RTCPeerConnectionIceEvent('ice')] + expected: FAIL + + [RTCDataChannel interface: operation close()] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "sdpMLineIndex" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: calling removeTrack(RTCRtpSender) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCIceCandidate interface: operation toJSON()] + expected: FAIL + + [RTCDataChannelEvent interface object length] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "candidate" with the proper type] + expected: FAIL + + [RTCErrorEvent interface: attribute error] + expected: FAIL + + [RTCPeerConnection interface: operation generateCertificate(AlgorithmIdentifier)] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "type" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: operation stop()] + expected: FAIL + + [RTCCertificate interface: idlTestObjects.certificate must inherit property "expires" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getSynchronizationSources()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [Stringification of new RTCErrorEvent('error')] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "url" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute onsignalingstatechange] + expected: FAIL + + [RTCPeerConnection interface: operation getStats(MediaStreamTrack)] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "dtmf" with the proper type] + expected: FAIL + + [RTCRtpSender interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: attribute onicegatheringstatechange] + expected: FAIL + + [RTCCertificate interface: operation getFingerprints()] + expected: FAIL + + [RTCSctpTransport interface: attribute state] + expected: FAIL + + [RTCSctpTransport interface: attribute maxChannels] + expected: FAIL + + [RTCIceTransport interface: operation getLocalParameters()] + expected: FAIL + + [RTCIceTransport interface: attribute ongatheringstatechange] + expected: FAIL + + [RTCSctpTransport interface: attribute maxMessageSize] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setLocalDescription(RTCSessionDescriptionInit)" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface object length] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getParameters()" with the proper type] + expected: FAIL + + [RTCCertificate interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCDtlsTransport interface object length] + expected: FAIL + + [RTCRtpTransceiver interface: attribute receiver] + expected: FAIL + + [RTCIceCandidate interface: attribute usernameFragment] + expected: FAIL + + [RTCDTMFToneChangeEvent interface object length] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "receiver" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "rtcpTransport" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "ontrack" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)] + expected: FAIL + + [RTCRtpTransceiver interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "priority" with the proper type] + expected: FAIL + + [RTCStatsEvent interface: attribute report] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute hostCandidate] + expected: FAIL + + [RTCRtpReceiver interface: attribute track] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "setStreams(MediaStream)" with the proper type] + expected: FAIL + + [RTCRtpReceiver interface: operation getParameters()] + expected: FAIL + + [RTCPeerConnection interface: operation getDefaultIceServers()] + expected: FAIL + + [RTCRtpTransceiver must be primary interface of new RTCPeerConnection().addTransceiver('audio')] + expected: FAIL + + [RTCDTMFSender interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface object length] + expected: FAIL + + [RTCDataChannel interface: attribute label] + expected: FAIL + + [Test driver for asyncInitMediaStreamTrack] + expected: FAIL + + [RTCPeerConnection interface: calling createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: operation setConfiguration(RTCConfiguration)] + expected: FAIL + + [RTCPeerConnection interface: calling createOffer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback, RTCOfferOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onconnectionstatechange" with the proper type] + expected: FAIL + + [RTCCertificate interface: idlTestObjects.certificate must inherit property "getSupportedAlgorithms()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute currentLocalDescription] + expected: FAIL + + [RTCDataChannel interface: operation send(Blob)] + expected: FAIL + + [RTCIceTransport interface: attribute onselectedcandidatepairchange] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "errorCode" with the proper type] + expected: FAIL + + [RTCCertificate interface: operation getSupportedAlgorithms()] + expected: FAIL + + [RTCPeerConnection interface: operation removeTrack(RTCRtpSender)] + expected: FAIL + + [RTCTrackEvent interface: attribute streams] + expected: FAIL + + [RTCPeerConnection interface: calling addIceCandidate(RTCIceCandidateInit) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpSender interface: operation getParameters()] + expected: FAIL + + [RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit)] + expected: FAIL + + [Test driver for asyncInitTransports] + expected: FAIL + + [RTCIceTransport interface: attribute onstatechange] + expected: FAIL + + [RTCPeerConnection interface: attribute iceConnectionState] + expected: FAIL + + [RTCDataChannel interface: attribute id] + expected: FAIL + + [RTCTrackEvent interface object length] + expected: FAIL + + [RTCPeerConnection interface: attribute ontrack] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "negotiated" with the proper type] + expected: FAIL + + [RTCCertificate interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCDataChannel interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCDTMFToneChangeEvent interface: attribute tone] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "localDescription" with the proper type] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "setParameters(RTCRtpSendParameters)" with the proper type] + expected: FAIL + + [RTCDTMFSender interface: attribute ontonechange] + expected: FAIL + + [RTCDataChannel interface: attribute readyState] + expected: FAIL + + [RTCError interface: attribute sctpCauseCode] + expected: FAIL + + [RTCStatsEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCStatsEvent interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "role" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: attribute candidate] + expected: FAIL + + [RTCPeerConnection interface: attribute connectionState] + expected: FAIL + + [RTCRtpReceiver interface object name] + expected: FAIL + + [RTCIceTransport interface: operation getRemoteParameters()] + expected: FAIL + + [RTCDtlsTransport interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "component" with the proper type] + expected: FAIL + + [RTCError interface: attribute httpRequestStatusCode] + expected: FAIL + + [RTCTrackEvent interface object name] + expected: FAIL + + [RTCRtpSender interface: operation getStats()] + expected: FAIL + + [RTCPeerConnection interface: operation getReceivers()] + expected: FAIL + + [RTCPeerConnection interface: operation addIceCandidate(RTCIceCandidateInit)] + expected: FAIL + + [Stringification of idlTestObjects.iceTransport] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createOffer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback, RTCOfferOptions)" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: calling setConfiguration(RTCConfiguration) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: calling createDataChannel(USVString, RTCDataChannelInit) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpReceiver interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: operation createAnswer(RTCAnswerOptions)] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getSenders()" with the proper type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalCandidates()" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setLocalDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute onopen] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "getCapabilities(DOMString)" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: attribute tcpType] + expected: FAIL + + [RTCTrackEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCRtpReceiver interface: attribute rtcpTransport] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedAddress" with the proper type] + expected: FAIL + + [RTCDataChannel interface: attribute ordered] + expected: FAIL + + [RTCPeerConnection interface: calling getStats(MediaStreamTrack) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "iceConnectionState" with the proper type] + expected: FAIL + + [RTCError interface: attribute sdpLineNumber] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "bufferedAmount" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCDtlsTransport interface: attribute state] + expected: FAIL + + [RTCPeerConnection must be primary interface of new RTCPeerConnection()] + expected: FAIL + + [RTCError interface: attribute receivedAlert] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getRemoteCandidates()" with the proper type] + expected: FAIL + + [RTCTrackEvent interface: initTrackEvent() must inherit property "receiver" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute errorCode] + expected: FAIL + + [RTCIceCandidate interface: attribute priority] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addIceCandidate(RTCIceCandidateInit)" with the proper type] + expected: FAIL + + [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "sdpMid" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "currentRemoteDescription" with the proper type] + expected: FAIL + + [RTCDataChannelEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCIceCandidate interface: attribute foundation] + expected: FAIL + + [RTCStatsReport interface object length] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "ondatachannel" with the proper type] + expected: FAIL + + [RTCDataChannel interface object length] + expected: FAIL + + [RTCSessionDescription interface: existence and properties of interface object] + expected: FAIL + + [RTCRtpSender interface: calling setParameters(RTCRtpSendParameters) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] + expected: FAIL + + [RTCStatsReport interface object name] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "gatheringState" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent must be primary interface of new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 });] + expected: FAIL + + [RTCErrorEvent interface object name] + expected: FAIL + + [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "onerror" with the proper type] + expected: FAIL + + [RTCIceTransport interface object name] + expected: FAIL + + [RTCDataChannelEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: operation setLocalDescription(RTCSessionDescriptionInit)] + expected: FAIL + + [RTCDataChannel interface: operation send(USVString)] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "pendingRemoteDescription" with the proper type] + expected: FAIL + + [RTCCertificate interface object name] + expected: FAIL + + [RTCPeerConnection interface: calling setRemoteDescription(RTCSessionDescriptionInit) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [Stringification of new RTCPeerConnection()] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getConfiguration()" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceEvent interface: attribute url] + expected: FAIL + + [RTCDataChannel must be primary interface of new RTCPeerConnection().createDataChannel('')] + expected: FAIL + + [RTCStatsEvent interface object length] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "iceGatheringState" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onsignalingstatechange" with the proper type] + expected: FAIL + + [RTCDTMFToneChangeEvent interface object name] + expected: FAIL + + [RTCDataChannel interface: attribute bufferedAmount] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface object] + expected: FAIL + + [RTCPeerConnection interface: attribute remoteDescription] + expected: FAIL + + [RTCIceTransport interface: attribute role] + expected: FAIL + + [RTCIceCandidate interface: attribute sdpMLineIndex] + expected: FAIL + + [RTCDataChannelEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCDtlsTransport interface: existence and properties of interface object] + expected: FAIL + + [RTCDtlsTransport interface: attribute iceTransport] + expected: FAIL + + [RTCPeerConnection interface: calling generateCertificate(AlgorithmIdentifier) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "currentLocalDescription" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: existence and properties of interface prototype object] + expected: FAIL + + [Stringification of new RTCPeerConnection().addTransceiver('audio').receiver] + expected: FAIL + + [RTCIceCandidate interface: attribute port] + expected: FAIL + + [RTCRtpSender interface object name] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "label" with the proper type] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "replaceTrack(MediaStreamTrack)" with the proper type] + expected: FAIL + + [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "type" with the proper type] + expected: FAIL + + [RTCDtlsTransport interface: attribute onstatechange] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "rtcpTransport" with the proper type] + expected: FAIL + + [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "track" with the proper type] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onerror" with the proper type] + expected: FAIL + + [RTCRtpTransceiver interface: operation setCodecPreferences([object Object\])] + expected: FAIL + + [RTCIceCandidate interface: attribute protocol] + expected: FAIL + + [RTCIceTransport interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCIceTransport interface: attribute component] + expected: FAIL + + [RTCPeerConnection interface: calling addTrack(MediaStreamTrack, MediaStream) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [Stringification of new RTCPeerConnection().addTransceiver('audio').sender] + expected: FAIL + + [RTCDTMFToneChangeEvent interface: existence and properties of interface object] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCPeerConnection interface: attribute oniceconnectionstatechange] + expected: FAIL + + [RTCDataChannel interface: attribute binaryType] + expected: FAIL + + [RTCSessionDescription interface: attribute type] + expected: FAIL + + [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "state" with the proper type] + expected: FAIL + + [RTCDTMFToneChangeEvent interface: existence and properties of interface prototype object's "constructor" property] + expected: FAIL + + [RTCPeerConnection interface: calling setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] + expected: FAIL + + [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getContributingSources()" with the proper type] + expected: FAIL + + [RTCDataChannel interface: calling send(ArrayBufferView) on new RTCPeerConnection().createDataChannel('') with too few arguments must throw TypeError] + expected: FAIL + + [RTCDtlsTransport interface: operation getRemoteCertificates()] + expected: FAIL + + [RTCStatsEvent interface object name] + expected: FAIL + + [RTCIceCandidate interface: attribute sdpMid] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(ArrayBufferView)" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceEvent must be primary interface of new RTCPeerConnectionIceEvent('ice')] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "hostCandidate" with the proper type] + expected: FAIL + + [RTCDataChannel interface: operation send(ArrayBuffer)] + expected: FAIL + + [RTCSessionDescription interface: operation toJSON()] + expected: FAIL + + [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(USVString)" with the proper type] + expected: FAIL + + [RTCIceTransport interface: operation getRemoteCandidates()] + expected: FAIL + + [RTCSctpTransport must be primary interface of idlTestObjects.sctpTransport] + expected: FAIL + + [RTCRtpTransceiver interface: attribute currentDirection] + expected: FAIL + + [RTCIceCandidate interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "maxMessageSize" with the proper type] + expected: FAIL + + [RTCError interface: existence and properties of interface prototype object] + expected: FAIL + + [Stringification of idlTestObjects.sctpTransport] + expected: FAIL + + [RTCIceTransport interface: operation getLocalCandidates()] + expected: FAIL + + [RTCDtlsTransport interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCPeerConnection interface: attribute canTrickleIceCandidates] + expected: FAIL + + [Stringification of initTrackEvent()] + expected: FAIL + + [RTCDataChannel interface: operation send(ArrayBufferView)] + expected: FAIL + + [RTCDTMFSender interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCRtpSender interface: attribute dtmf] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "pendingLocalDescription" with the proper type] + expected: FAIL + + [RTCPeerConnectionIceErrorEvent interface: attribute url] + expected: FAIL + + [RTCRtpSender interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCError interface: existence and properties of interface object] + expected: FAIL + + [RTCSctpTransport interface object length] + expected: FAIL + + [RTCIceCandidate interface: attribute relatedAddress] + expected: FAIL + + [RTCRtpSender interface: operation replaceTrack(MediaStreamTrack)] + expected: FAIL + + [RTCDTMFSender interface: existence and properties of interface object] + expected: FAIL + + [RTCRtpReceiver interface: existence and properties of interface prototype object's @@unscopables property] + expected: FAIL + + [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "transport" with the proper type] + expected: FAIL + + [RTCDTMFSender interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: operation setLocalDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)] + expected: FAIL + + [RTCSessionDescription interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCDtlsTransport interface: attribute onerror] + expected: FAIL + + [RTCDataChannel interface: attribute onclose] + expected: FAIL + + [Stringification of new RTCPeerConnection().createDataChannel('')] + expected: FAIL + + [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "stop()" with the proper type] + expected: FAIL + + [RTCErrorEvent interface: existence and properties of interface prototype object] + expected: FAIL + + [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "connectionState" with the proper type] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-addStream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-addStream.https.html.ini new file mode 100644 index 000000000000..491868c9613b --- /dev/null +++ b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-addStream.https.html.ini @@ -0,0 +1,4 @@ +[RTCPeerConnection-addStream.https.html] + [Legacy addStream(): Media stream stats references track stats] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini new file mode 100644 index 000000000000..874c5a94c9cc --- /dev/null +++ b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini @@ -0,0 +1,55 @@ +[RTCPeerConnection-createOffer-offerToReceive.html] + [offerToReceiveVideo option should be ignored if a non-stopped "recvonly" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveAudio should add audio line to all subsequent created offers] + expected: FAIL + + [createOffer() with offerToReceiveAudio set to false should not create a transceiver] + expected: FAIL + + [offerToReceiveVideo option should be ignored if a non-stopped "sendrecv" transceiver exists] + expected: FAIL + + [offerToReceiveAudio set to false with a track should create a "sendonly" transceiver] + expected: FAIL + + [subsequent offerToReceiveAudio set to false with a track should change the direction to "sendonly"] + expected: FAIL + + [offerToReceiveAudio option should be ignored if a non-stopped "sendrecv" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveVideo should add video line to all subsequent created offers] + expected: FAIL + + [subsequent offerToReceiveVideo set to false with a track should change the direction to "sendonly"] + expected: FAIL + + [createOffer() with offerToReceiveAudio should create a "recvonly" transceiver] + expected: FAIL + + [offerToReceiveAudio option should be ignored if a non-stopped "recvonly" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveAudio:true, then with offerToReceiveVideo:true, should have result offer with both audio and video line] + expected: FAIL + + [offerToReceiveAudio and Video should create two "recvonly" transceivers] + expected: FAIL + + [createOffer() with offerToReceiveVideo set to false should not create a transceiver] + expected: FAIL + + [offerToReceiveVideo set to false with a track should create a "sendonly" transceiver] + expected: FAIL + + [offerToReceiveAudio set to false with a "recvonly" transceiver should change the direction to "inactive"] + expected: FAIL + + [offerToReceiveVideo set to false with a "recvonly" transceiver should change the direction to "inactive"] + expected: FAIL + + [createOffer() with offerToReceiveVideo should create a "recvonly" transceiver] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini new file mode 100644 index 000000000000..841c26c5f9a3 --- /dev/null +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -0,0 +1,13 @@ +[RTCRtpTransceiver-with-OfferToReceive-options.https.html] + [checkAddTransceiverWithStream] + expected: FAIL + + [checkAddTransceiverWithOfferToReceiveVideo] + expected: FAIL + + [checkAddTransceiverWithOfferToReceiveBoth] + expected: FAIL + + [checkAddTransceiverWithOfferToReceiveAudio] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini new file mode 100644 index 000000000000..71b4cc138f65 --- /dev/null +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -0,0 +1,4 @@ +[onaddstream.https.html] + [Check onaddstream] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/no-media-call.html.ini b/tests/wpt/metadata/webrtc/no-media-call.html.ini new file mode 100644 index 000000000000..5a3fd876b999 --- /dev/null +++ b/tests/wpt/metadata/webrtc/no-media-call.html.ini @@ -0,0 +1,4 @@ +[no-media-call.html] + [Can set up a basic WebRTC call with no data.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/promises-call.html.ini b/tests/wpt/metadata/webrtc/promises-call.html.ini new file mode 100644 index 000000000000..644ee144700a --- /dev/null +++ b/tests/wpt/metadata/webrtc/promises-call.html.ini @@ -0,0 +1,4 @@ +[promises-call.html] + [Can set up a basic WebRTC call with only data using promises.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini b/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini new file mode 100644 index 000000000000..db02fef95386 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini @@ -0,0 +1,10 @@ +[ice-state.https.html] + [PC should generate offer with a=ice-options:trickle] + expected: FAIL + + [PC should enter connected state when candidates are sent] + expected: FAIL + + [PC should enter disconnected state when a failing candidate is sent] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini b/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini new file mode 100644 index 000000000000..8ee2a670e815 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini @@ -0,0 +1,4 @@ +[jsep-initial-offer.https.html] + [Offer conforms to basic SDP requirements] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/missing-fields.html.ini b/tests/wpt/metadata/webrtc/protocol/missing-fields.html.ini new file mode 100644 index 000000000000..eccf5f7d6519 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/missing-fields.html.ini @@ -0,0 +1,7 @@ +[missing-fields.html] + [Offer description with no mid is accepted] + expected: FAIL + + [Answer description with no mid is accepted] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini b/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini new file mode 100644 index 000000000000..260513f58fbb --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini @@ -0,0 +1,13 @@ +[msid-parse.html] + [Description with no msid produces a track with a stream] + expected: FAIL + + [Description with two msid produces two streams] + expected: FAIL + + [Description with msid:foo bar produces a stream with id foo] + expected: FAIL + + [Description with msid:- appid produces a track with no stream] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/simulcast-answer.html.ini b/tests/wpt/metadata/webrtc/protocol/simulcast-answer.html.ini new file mode 100644 index 000000000000..641ec8fb9d79 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/simulcast-answer.html.ini @@ -0,0 +1,4 @@ +[simulcast-answer.html] + [createOffer() with multiple send encodings should create simulcast offer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/simulcast-offer.html.ini b/tests/wpt/metadata/webrtc/protocol/simulcast-offer.html.ini new file mode 100644 index 000000000000..11cd69418f11 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/simulcast-offer.html.ini @@ -0,0 +1,4 @@ +[simulcast-offer.html] + [createOffer() with multiple send encodings should create simulcast offer] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini new file mode 100644 index 000000000000..d316de1f6122 --- /dev/null +++ b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini @@ -0,0 +1,10 @@ +[video-codecs.https.html] + [H.264 and VP8 should be negotiated after handshake] + expected: FAIL + + [H.264 and VP8 should be supported in initial offer] + expected: FAIL + + [All H.264 codecs MUST include profile-level-id] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini new file mode 100644 index 000000000000..b3469e44234e --- /dev/null +++ b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini @@ -0,0 +1,4 @@ +[simplecall-no-ssrcs.https.html] + [Can set up a basic WebRTC call without announcing ssrcs.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/simplecall.https.html.ini b/tests/wpt/metadata/webrtc/simplecall.https.html.ini new file mode 100644 index 000000000000..c4522527714c --- /dev/null +++ b/tests/wpt/metadata/webrtc/simplecall.https.html.ini @@ -0,0 +1,4 @@ +[simplecall.https.html] + [Can set up a basic WebRTC call.] + expected: FAIL + From 350157fd655cb9627df434cc984fc28474fcc7d6 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 11:57:36 -0700 Subject: [PATCH 05/13] Enable webrtc pref for webrtc folder --- .../RTCConfiguration-bundlePolicy.html.ini | 6 - ...onfiguration-iceCandidatePoolSize.html.ini | 6 - .../RTCConfiguration-iceServers.html.ini | 24 -- ...CConfiguration-iceTransportPolicy.html.ini | 9 - .../RTCConfiguration-rtcpMuxPolicy.html.ini | 6 - .../RTCIceCandidate-constructor.html.ini | 48 --- .../RTCIceTransport-extension.https.html.ini | 34 +- ...ction-add-track-no-deadlock.https.html.ini | 1 + ...RTCPeerConnection-addIceCandidate.html.ini | 21 -- .../RTCPeerConnection-addTrack.https.html.ini | 17 +- ...erConnection-addTransceiver.https.html.ini | 3 +- ...rConnection-connectionState.https.html.ini | 3 +- .../RTCPeerConnection-constructor.html.ini | 39 --- .../RTCPeerConnection-createAnswer.html.ini | 7 +- ...CPeerConnection-createDataChannel.html.ini | 30 -- .../RTCPeerConnection-createOffer.html.ini | 6 - .../RTCPeerConnection-getStats.https.html.ini | 15 +- ...onnectionState-disconnected.https.html.ini | 1 + ...nnection-iceConnectionState.https.html.ini | 6 - ...CPeerConnection-iceGatheringState.html.ini | 6 +- ...eerConnection-onnegotiationneeded.html.ini | 5 +- ...ion-onsignalingstatechanged.https.html.ini | 1 + .../RTCPeerConnection-ontrack.https.html.ini | 6 +- ...onnection-remote-track-mute.https.html.ini | 7 +- ...CPeerConnection-removeTrack.https.html.ini | 25 +- ...ection-setLocalDescription-answer.html.ini | 11 +- ...nection-setLocalDescription-offer.html.ini | 11 +- ...tion-setLocalDescription-pranswer.html.ini | 3 - ...tion-setLocalDescription-rollback.html.ini | 7 +- ...eerConnection-setLocalDescription.html.ini | 3 - ...ction-setRemoteDescription-nomsid.html.ini | 3 +- ...ion-setRemoteDescription-pranswer.html.ini | 3 - ...ion-setRemoteDescription-rollback.html.ini | 9 +- ...setRemoteDescription-tracks.https.html.ini | 27 +- ...erConnection-setRemoteDescription.html.ini | 9 - ...CPeerConnection-track-stats.https.html.ini | 11 +- ...PeerConnection-transceivers.https.html.ini | 87 ++--- ...eerConnectionIceEvent-constructor.html.ini | 21 -- ...iver-getContributingSources.https.html.ini | 3 +- .../RTCRtpReceiver-getStats.https.html.ini | 3 +- ...r-getSynchronizationSources.https.html.ini | 23 +- .../RTCRtpSender-getStats.https.html.ini | 3 +- .../RTCRtpSender-replaceTrack.https.html.ini | 17 +- .../RTCRtpSender-transport.https.html.ini | 9 +- .../webrtc/RTCRtpTransceiver.https.html.ini | 69 ++-- .../webrtc/RTCTrackEvent-fire.html.ini | 5 +- tests/wpt/metadata/webrtc/__dir__.ini | 1 + tests/wpt/metadata/webrtc/historical.html.ini | 24 -- .../webrtc/idlharness.https.window.js.ini | 312 ------------------ ...with-OfferToReceive-options.https.html.ini | 7 +- .../webrtc/legacy/onaddstream.https.html.ini | 1 + .../metadata/webrtc/no-media-call.html.ini | 3 +- .../jsep-initial-offer.https.html.ini | 4 - .../webrtc/protocol/msid-parse.html.ini | 9 +- 54 files changed, 253 insertions(+), 777 deletions(-) create mode 100644 tests/wpt/metadata/webrtc/__dir__.ini delete mode 100644 tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini index 6e84784af60a..3b3b678b02ec 100644 --- a/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-bundlePolicy.html.ini @@ -35,12 +35,6 @@ [Default bundlePolicy should be balanced] expected: FAIL - [new RTCPeerConnection({ bundlePolicy: 'invalid' }) should throw TypeError] - expected: FAIL - - [new RTCPeerConnection({ bundlePolicy: null }) should throw TypeError] - expected: FAIL - [new RTCPeerConnection({ bundlePolicy: 'balanced' }) should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini index 3ed2265cbd75..9d89a1e55b50 100644 --- a/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceCandidatePoolSize.html.ini @@ -14,15 +14,9 @@ [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to 256 (Out Of Range)] expected: FAIL - [Initialize a new RTCPeerConnection with iceCandidatePoolSize: 256 (Out Of Range)] - expected: FAIL - [Initialize a new RTCPeerConnection with no iceCandidatePoolSize] expected: FAIL - [Initialize a new RTCPeerConnection with iceCandidatePoolSize: -1 (Out Of Range)] - expected: FAIL - [Reconfigure RTCPeerConnection instance iceCandidatePoolSize to -1 (Out Of Range)] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini index 9d995562d6fd..4f357ec8c1c0 100644 --- a/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceServers.html.ini @@ -38,9 +38,6 @@ [new RTCPeerConnection() should have default configuration.iceServers of undefined] expected: FAIL - [new RTCPeerConnection(config) - with credentialType token should throw TypeError] - expected: FAIL - [setConfiguration(config) - with turns server, credentialType oauth and RTCOAuthCredential credential should succeed] expected: FAIL @@ -56,9 +53,6 @@ [setConfiguration(config) - with stun server should succeed] expected: FAIL - [new RTCPeerConnection(config) - {} should succeed] - expected: FAIL - [new RTCPeerConnection(config) - with turns server, credentialType oauth and RTCOAuthCredential credential should succeed] expected: FAIL @@ -107,9 +101,6 @@ [new RTCPeerConnection(config) - with both turns and stun server, credentialType oauth and RTCOAuthCredential credential should succeed] expected: FAIL - [new RTCPeerConnection(config) - with url field should throw TypeError] - expected: FAIL - [setConfiguration(config) - {} should succeed] expected: FAIL @@ -158,12 +149,6 @@ [setConfiguration(config) - with credentialType token should throw TypeError] expected: FAIL - [new RTCPeerConnection(config) - { iceServers: [{}\] } should throw TypeError] - expected: FAIL - - [new RTCPeerConnection(config) - { iceServers: [null\] } should throw TypeError] - expected: FAIL - [setConfiguration(config) - { iceServers: [\] } should succeed] expected: FAIL @@ -197,9 +182,6 @@ [new RTCPeerConnection(config) - with turn server and empty string username, credential should succeed] expected: FAIL - [new RTCPeerConnection(config) - { iceServers: null } should throw TypeError] - expected: FAIL - [setConfiguration(config) - with http url should throw SyntaxError] expected: FAIL @@ -209,12 +191,6 @@ [new RTCPeerConnection(config) - with empty urls should throw SyntaxError] expected: FAIL - [new RTCPeerConnection(config) - { iceServers: [undefined\] } should throw TypeError] - expected: FAIL - - [new RTCPeerConnection(config) - with invalid credentialType should throw TypeError] - expected: FAIL - [setConfiguration(config) - with turns server, credentialType password, and RTCOauthCredential credential should throw InvalidAccessError] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini index b2f3e0a5c479..c51b53bb31ff 100644 --- a/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-iceTransportPolicy.html.ini @@ -2,24 +2,15 @@ [setConfiguration(config) - with null iceTransportPolicy should throw TypeError] expected: FAIL - [new RTCPeerConnection(config) - with none iceTransportPolicy should throw TypeError] - expected: FAIL - [new RTCPeerConnection({ iceTransportPolicy: undefined }) should have default iceTransportPolicy all] expected: FAIL [setConfiguration({}) with initial iceTransportPolicy relay should set new value to all] expected: FAIL - [new RTCPeerConnection(config) - with invalid iceTransportPolicy should throw TypeError] - expected: FAIL - [new RTCPeerConnection({ iceTransportPolicy: 'relay' }) should succeed] expected: FAIL - [new RTCPeerConnection(config) - with null iceTransportPolicy should throw TypeError] - expected: FAIL - [setConfiguration({ iceTransportPolicy: 'relay' }) with initial iceTransportPolicy all should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini b/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini index 2d233dfc3703..3bd8bdcb31d0 100644 --- a/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini +++ b/tests/wpt/metadata/webrtc/RTCConfiguration-rtcpMuxPolicy.html.ini @@ -14,9 +14,6 @@ [new RTCPeerConnection({ rtcpMuxPolicy: 'negotiate' }) may succeed or throw NotSupportedError] expected: FAIL - [new RTCPeerConnection(config) - with { rtcpMuxPolicy: 'invalid' } should throw TypeError] - expected: FAIL - [setConfiguration({ rtcpMuxPolicy: 'require' }) with initial rtcpMuxPolicy negotiate should throw InvalidModificationError] expected: FAIL @@ -35,6 +32,3 @@ [new RTCPeerConnection() should have default rtcpMuxPolicy require] expected: FAIL - [new RTCPeerConnection(config) - with { rtcpMuxPolicy: null } should throw TypeError] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini index e2ba0bd7c582..250ef2003cdf 100644 --- a/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini +++ b/tests/wpt/metadata/webrtc/RTCIceCandidate-constructor.html.ini @@ -1,55 +1,7 @@ [RTCIceCandidate-constructor.html] - [new RTCIceCandidate({ sdpMLineIndex: 0 })] - expected: FAIL - - [new RTCIceCandidate({ ... }) with invalid sdpMid] - expected: FAIL - - [new RTCIceCandidate({ ... }) with invalid sdpMLineIndex] - expected: FAIL - - [new RTCIceCandidate()] - expected: FAIL - - [new RTCIceCandidate({ sdpMid: null, sdpMLineIndex: null })] - expected: FAIL - - [new RTCIceCandidate({ ... }) with valid candidate string and sdpMid] - expected: FAIL - [new RTCIceCandidate({ ... }) with nondefault values for all fields, tcp candidate] expected: FAIL - [new RTCIceCandidate({ candidate: '', sdpMid: 'audio' }] - expected: FAIL - - [new RTCIceCandidate({ candidate: null })] - expected: FAIL - - [new RTCIceCandidate({})] - expected: FAIL - - [new RTCIceCandidate({ sdpMid: 'audio' })] - expected: FAIL - - [new RTCIceCandidate({ ... }) with manually filled default values] - expected: FAIL - - [new RTCIceCandidate({ candidate: '', sdpMLineIndex: 0 }] - expected: FAIL - [new RTCIceCandidate({ ... }) with nondefault values for all fields] expected: FAIL - [new RTCIceCandidate({ candidate: '' })] - expected: FAIL - - [new RTCIceCandidate({ ... }) with invalid candidate string and sdpMid] - expected: FAIL - - [new RTCIceCandidate({ ... }) with valid candidate string only] - expected: FAIL - - [new RTCIceCandidate({ sdpMid: 'audio', sdpMLineIndex: 0 })] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini index 7c58f72543c8..fbb77db1161c 100644 --- a/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini @@ -1,5 +1,4 @@ [RTCIceTransport-extension.https.html] - expected: ERROR [RTCIceTransport initial properties are set] expected: FAIL @@ -54,3 +53,36 @@ [gather() throws if closed] expected: FAIL + [addRemoteCandidate() throws if closed] + expected: FAIL + + [Two RTCIceTransports configured with the controlled role resolve the conflict in band and still connect.] + expected: FAIL + + [addRemoteCandidate() transitions state to 'checking' if start() had been called before] + expected: FAIL + + [Selected candidate pair changes once the RTCIceTransports connect.] + expected: FAIL + + [Two RTCIceTransports connect to each other] + expected: FAIL + + [start() flushes remote candidates and transitions state to 'new' if later called with different remote parameters] + expected: FAIL + + [start() throws if later called with a different role] + expected: FAIL + + [addRemoteCandidate() throws on invalid candidate] + expected: FAIL + + [Two RTCIceTransports configured with the controlling role resolve the conflict in band and still connect.] + expected: FAIL + + [start() transitions state to 'checking' if one remote candidate had been added] + expected: FAIL + + [getSelectedCandidatePair() returns null once the RTCIceTransport is stopped.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini index 3e9d2b4aaca3..033edaad84f7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-add-track-no-deadlock.https.html] + expected: ERROR [RTCPeerConnection addTrack does not deadlock.] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini index e82723288688..aa70cfeec278 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addIceCandidate.html.ini @@ -1,7 +1,4 @@ [RTCPeerConnection-addIceCandidate.html] - [Add candidate with only valid sdpMLineIndex should succeed] - expected: FAIL - [addIceCandidate({usernameFragment: usernameFragment1, sdpMid: sdpMid1}) should work, and add a=end-of-candidates to the first m-section] expected: FAIL @@ -11,18 +8,9 @@ [Add candidate with only valid sdpMid should succeed] expected: FAIL - [Add ICE candidate after setting remote description should succeed] - expected: FAIL - [addIceCandidate({usernameFragment: "no such ufrag"}) should not work] expected: FAIL - [Add candidate with invalid candidate string and both sdpMid and sdpMLineIndex null should reject with TypeError] - expected: FAIL - - [Add candidate with both sdpMid and sdpMLineIndex manually set to null should reject with TypeError] - expected: FAIL - [addIceCandidate with second sdpMid and sdpMLineIndex should add candidate to second media stream] expected: FAIL @@ -32,15 +20,9 @@ [Add candidate with sdpMid belonging to different usernameFragment should reject with OperationError] expected: FAIL - [Add candidate with only valid candidate string should reject with TypeError] - expected: FAIL - [Add candidate with invalid sdpMLineIndex should reject with OperationError] expected: FAIL - [Add ICE candidate with RTCIceCandidate should succeed] - expected: FAIL - [Add candidate for first media stream with null usernameFragment should add candidate to first media stream] expected: FAIL @@ -77,6 +59,3 @@ [addIceCandidate({"candidate":"","sdpMid":null,"sdpMLineIndex":null}) should work, and add a=end-of-candidates to both m-sections] expected: FAIL - [Invalid sdpMLineIndex should be ignored if valid sdpMid is provided] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini index e3ad91184bb7..6de0040367c7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini @@ -1,28 +1,29 @@ [RTCPeerConnection-addTrack.https.html] + expected: ERROR [addTrack with existing sender that has been used to send should create new sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and multiple streams should succeed] - expected: FAIL + expected: NOTRUN [addTrack with existing sender with null track, different kind, and recvonly direction should create new sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and no stream should succeed] - expected: FAIL + expected: NOTRUN [addTrack with existing sender that has not been used to send should reuse the sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and single stream should succeed] - expected: FAIL + expected: NOTRUN [addTrack when pc is closed should throw InvalidStateError] expected: FAIL [addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender] - expected: FAIL + expected: NOTRUN [Adding the same track multiple times should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini index 1b5424929ef5..46efcaa35623 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-addTransceiver.https.html] + expected: ERROR [addTransceiver() with direction inactive should have result transceiver.direction be the same] expected: FAIL @@ -33,5 +34,5 @@ expected: FAIL [addTransceiver(track) multiple times should create multiple transceivers] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini index 1b387aeb1f73..73eb59444599 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-connectionState.https.html] + expected: ERROR [Initial connectionState should be new] expected: FAIL @@ -6,7 +7,7 @@ expected: FAIL [connectionState transitions to connected via connecting] - expected: FAIL + expected: NOTRUN [connection with one data channel should eventually have transports in connected state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini index 0e8361a8bb5b..9aad6750938a 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-constructor.html.ini @@ -1,13 +1,4 @@ [RTCPeerConnection-constructor.html] - [signalingState initial value] - expected: FAIL - - [iceConnectionState initial value] - expected: FAIL - - [new RTCPeerConnection()] - expected: FAIL - [connectionState initial value] expected: FAIL @@ -17,51 +8,21 @@ [pendingLocalDescription initial value] expected: FAIL - [new RTCPeerConnection({ iceCandidatePoolSize: toNumberThrows })] - expected: FAIL - - [iceGatheringState initial value] - expected: FAIL - - [RTCPeerConnection.length] - expected: FAIL - - [new RTCPeerConnection({})] - expected: FAIL - [new RTCPeerConnection({ certificates: null })] expected: FAIL [canTrickleIceCandidates initial value] expected: FAIL - [localDescription initial value] - expected: FAIL - [currentLocalDescription initial value] expected: FAIL [new RTCPeerConnection({ certificates: [null\] })] expected: FAIL - [new RTCPeerConnection({ certificates: undefined })] - expected: FAIL - [currentRemoteDescription initial value] expected: FAIL - [new RTCPeerConnection(null)] - expected: FAIL - - [new RTCPeerConnection({ certificates: [\] })] - expected: FAIL - - [new RTCPeerConnection(undefined)] - expected: FAIL - - [remoteDescription initial value] - expected: FAIL - [new RTCPeerConnection({ certificates: [undefined\] })] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini index 3fc9569af419..333d2ec4454b 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -1,10 +1,11 @@ [RTCPeerConnection-createAnswer.html] + expected: TIMEOUT [createAnswer() when connection is closed reject with InvalidStateError] - expected: FAIL + expected: NOTRUN [createAnswer() after setting remote description should succeed] - expected: FAIL + expected: NOTRUN [createAnswer() with null remoteDescription should reject with InvalidStateError] - expected: FAIL + expected: TIMEOUT diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini index deea9840f2a7..f672fd7d1e72 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createDataChannel.html.ini @@ -14,15 +14,9 @@ [createDataChannel with negotiated false should succeed] expected: FAIL - [createDataChannel with negotiated true and id not defined should throw TypeError] - expected: FAIL - [createDataChannel with negotiated false and id 42 should ignore the id] expected: FAIL - [createDataChannel with both maxPacketLifeTime and maxRetransmits should throw TypeError] - expected: FAIL - [createDataChannel with label undefined should succeed] expected: FAIL @@ -41,9 +35,6 @@ [createDataChannel with ordered null/undefined should succeed] expected: FAIL - [createDataChannel with too long label should throw TypeError] - expected: FAIL - [createDataChannel with label "foo" should succeed] expected: FAIL @@ -53,33 +44,18 @@ [createDataChannel with no argument should throw TypeError] expected: FAIL - [createDataChannel with too long label (2 byte unicode) should throw TypeError] - expected: FAIL - - [createDataChannel with id -1 should throw TypeError] - expected: FAIL - - [createDataChannel with invalid priority should throw TypeError] - expected: FAIL - [createDataChannel with id 0 should succeed] expected: FAIL [createDataChannel attribute default values] expected: FAIL - [createDataChannel with too long protocol (2 byte unicode) should throw TypeError] - expected: FAIL - [Channels created (after setRemoteDescription) should have id assigned] expected: FAIL [createDataChannel with ordered false should succeed] expected: FAIL - [createDataChannel with too long protocol should throw TypeError] - expected: FAIL - [createDataChannel with label lone surrogate should succeed] expected: FAIL @@ -116,9 +92,3 @@ [createDataChannel with same label used twice should not throw] expected: FAIL - [createDataChannel with id 65535 should throw TypeError] - expected: FAIL - - [createDataChannel with id 65536 should throw TypeError] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini index 794e6de939a9..71490a1c46cb 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createOffer.html.ini @@ -1,13 +1,7 @@ [RTCPeerConnection-createOffer.html] - [createOffer() after connection is closed should reject with InvalidStateError] - expected: FAIL - [When media stream is added when createOffer() is running in parallel, the result offer should contain the new media stream] expected: FAIL [createOffer() and then setLocalDescription() should succeed] expected: FAIL - [createOffer() with no argument from newly created RTCPeerConnection should succeed] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini index 0b0f45d1843e..b7adadda25ec 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -1,21 +1,22 @@ [RTCPeerConnection-getStats.https.html] + expected: ERROR [getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats] expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats and outbound-track-stats] - expected: FAIL + expected: NOTRUN [getStats() with track associated with both sender and receiver should reject with InvalidAccessError] - expected: FAIL + expected: NOTRUN [getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats] - expected: FAIL + expected: NOTRUN [getStats() with no argument should return stats for no-stream tracks] - expected: FAIL + expected: NOTRUN [getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats] - expected: FAIL + expected: NOTRUN [getStats() with no argument should succeed] expected: FAIL @@ -27,13 +28,13 @@ expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats on an empty PC] - expected: FAIL + expected: NOTRUN [getStats() with track added via addTransceiver should succeed] expected: FAIL [getStats() with track associated with more than one sender should reject with InvalidAccessError] - expected: FAIL + expected: NOTRUN [getStats() with track added via addTrack should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini index 5914c3ea93e2..b135b4af16f2 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-iceConnectionState-disconnected.https.html] + expected: ERROR [ICE goes to disconnected if the other side goes away] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini index a6e68735fbfd..7bd5bbdf08e3 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -1,13 +1,7 @@ [RTCPeerConnection-iceConnectionState.https.html] - [Initial iceConnectionState should be new] - expected: FAIL - [ICE can connect in a recvonly usecase] expected: FAIL - [Closing the connection should set iceConnectionState to closed] - expected: FAIL - [connection with one data channel should eventually have connected connection state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini index 3423f87aabb0..ee97c9e7f325 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceGatheringState.html.ini @@ -1,9 +1,7 @@ [RTCPeerConnection-iceGatheringState.html] + expected: TIMEOUT [iceGatheringState should eventually become complete after setLocalDescription] - expected: FAIL - - [Initial iceGatheringState should be new] - expected: FAIL + expected: TIMEOUT [connection with one data channel should eventually have connected connection state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index 124c16268d9c..434392114458 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -1,8 +1,5 @@ [RTCPeerConnection-onnegotiationneeded.html] expected: ERROR - [negotiationneeded event should not fire if signaling state is not stable] - expected: FAIL - [addTransceiver() should fire negotiationneeded event] expected: FAIL @@ -19,7 +16,7 @@ expected: FAIL [Updating the direction of the transceiver should cause negotiationneeded to fire] - expected: FAIL + expected: NOTRUN [negotiationneeded event should fire only after signaling state go back to stable after setLocalDescription] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini index d564904adfb2..a81b9e2ce821 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-onsignalingstatechanged.https.html] + expected: ERROR [RTCPeerConnection onsignalingstatechanged] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini index f911ff8026ec..a880161b7dd3 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-ontrack.https.html.ini @@ -1,12 +1,10 @@ [RTCPeerConnection-ontrack.https.html] + expected: TIMEOUT [addTrack() should cause remote connection to fire ontrack when setRemoteDescription()] expected: FAIL - [setRemoteDescription() with m= line of recvonly direction should not trigger track event] - expected: FAIL - [setRemoteDescription should trigger ontrack event when the MSID of the stream is is parsed.] - expected: FAIL + expected: TIMEOUT [addTransceiver('video') should cause remote connection to fire ontrack when setRemoteDescription()] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini index b8cc12e7e538..36b8c351cb59 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -1,12 +1,13 @@ [RTCPeerConnection-remote-track-mute.https.html] + expected: ERROR [Changing transceiver direction to 'sendrecv' unmutes the remote track] - expected: FAIL + expected: NOTRUN [pc.close() mutes remote tracks] - expected: FAIL + expected: NOTRUN [Changing transceiver direction to 'inactive' mutes the remote track] - expected: FAIL + expected: NOTRUN [ontrack: track goes from muted to unmuted] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini index 03f54a272326..6875bfdb7e12 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini @@ -1,40 +1,41 @@ [RTCPeerConnection-removeTrack.https.html] + expected: ERROR [Calling removeTrack with currentDirection sendonly should set direction to inactive] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack when connection is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack with valid sender should set sender.track to null] - expected: FAIL + expected: NOTRUN [Calling removeTrack on a stopped transceiver should be a no-op] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack when connection is closed should throw InvalidStateError] expected: FAIL [addTrack - Calling removeTrack on different connection should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection inactive should not change direction] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection sendrecv should set direction to recvonly] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection recvonly should not change direction] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack with valid sender should set sender.track to null] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini index 7d6d3f228390..d4733273c51d 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-answer.html.ini @@ -1,18 +1,19 @@ [RTCPeerConnection-setLocalDescription-answer.html] + expected: TIMEOUT [Setting previously generated answer after a call to createOffer should work] - expected: FAIL + expected: NOTRUN [Calling setLocalDescription(answer) from have-local-offer state should reject with InvalidModificationError] - expected: FAIL + expected: NOTRUN [setLocalDescription() with type answer and null sdp should use lastAnswer generated from createAnswer] - expected: FAIL + expected: TIMEOUT [Calling setLocalDescription(answer) from stable state should reject with InvalidModificationError] - expected: FAIL + expected: NOTRUN [setLocalDescription() with answer not created by own createAnswer() should reject with InvalidModificationError] - expected: FAIL + expected: NOTRUN [setLocalDescription() with valid answer should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini index 415598909b2a..4db3cc7b2870 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-offer.html.ini @@ -1,19 +1,20 @@ [RTCPeerConnection-setLocalDescription-offer.html] + expected: TIMEOUT [Set created offer other than last offer should reject with InvalidModificationError] - expected: FAIL + expected: NOTRUN [setLocalDescription() with offer not created by own createOffer() should reject with InvalidModificationError] - expected: FAIL + expected: NOTRUN [setLocalDescription with valid offer should succeed] expected: FAIL [Setting previously generated offer after a call to createAnswer should work] - expected: FAIL + expected: NOTRUN [setLocalDescription with type offer and null sdp should use lastOffer generated from createOffer] - expected: FAIL + expected: TIMEOUT [Creating and setting offer multiple times should succeed] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini index f3c47f23515a..49e0f57d3a2d 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html.ini @@ -8,6 +8,3 @@ [setLocalDescription(answer) from have-local-pranswer state should succeed] expected: FAIL - [setLocalDescription(pranswer) can be applied multiple times while still in have-local-pranswer] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini index 0f1c2de4941b..1269ea7f95bd 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription-rollback.html.ini @@ -1,12 +1,13 @@ [RTCPeerConnection-setLocalDescription-rollback.html] + expected: TIMEOUT [setLocalDescription(rollback) after setting answer description should reject with InvalidStateError] - expected: FAIL + expected: NOTRUN [setLocalDescription(rollback) should ignore invalid sdp content and succeed] - expected: FAIL + expected: NOTRUN [setLocalDescription(rollback) from stable state should reject with InvalidStateError] - expected: FAIL + expected: TIMEOUT [setLocalDescription(rollback) from have-local-offer state should reset back to stable state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini index 80d16956b1ac..31b794ddd864 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setLocalDescription.html.ini @@ -2,9 +2,6 @@ [Switching role from answerer to offerer after going back to stable state should succeed] expected: FAIL - [onsignalingstatechange fires before setLocalDescription resolves] - expected: FAIL - [Calling createOffer() and setLocalDescription() again after one round of local-offer/remote-answer should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini index a26b4f69f8ba..dbc455e6a0b1 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-nomsid.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-setRemoteDescription-nomsid.html] + expected: TIMEOUT [setRemoteDescription with an SDP without a=msid lines triggers ontrack with a default stream.] - expected: FAIL + expected: TIMEOUT diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini index 830a3ac69d30..81ec65365060 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html.ini @@ -2,9 +2,6 @@ [setRemoteDescription(answer) from have-remote-pranswer state should succeed] expected: FAIL - [setRemoteDescription(pranswer) multiple times should succeed] - expected: FAIL - [setRemoteDescription(pranswer) from stable state should reject with InvalidStateError] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini index 7d8eb7285488..35321b65a089 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html.ini @@ -1,16 +1,17 @@ [RTCPeerConnection-setRemoteDescription-rollback.html] + expected: TIMEOUT [setRemoteDescription(rollback) from stable state should reject with InvalidStateError] - expected: FAIL + expected: TIMEOUT [setRemoteDescription(rollback) should ignore invalid sdp content and succeed] - expected: FAIL + expected: NOTRUN [local offer created before setRemoteDescription(remote offer) then rollback should still be usable] - expected: FAIL + expected: NOTRUN [setRemoteDescription(rollback) in have-remote-offer state should revert to stable state] expected: FAIL [local offer created before setRemoteDescription(remote offer) with different transceiver level assignments then rollback should still be usable] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini index a8cb104dff57..0f67a2850ead 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -1,43 +1,44 @@ [RTCPeerConnection-setRemoteDescription-tracks.https.html] + expected: ERROR [ontrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() for an existing stream makes stream.onaddtrack fire.] - expected: FAIL + expected: NOTRUN [ontrack's receiver matches getReceivers().] - expected: FAIL + expected: NOTRUN [track.onmute fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() with two tracks and one stream makes ontrack fire twice with the tracks and shared stream.] - expected: FAIL + expected: NOTRUN [stream.onaddtrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() with a track and no stream makes ontrack fire with a track and no stream.] expected: FAIL [addTrack() with a track and a stream makes ontrack fire with a track and a stream.] - expected: FAIL + expected: NOTRUN [removeTrack() makes track.onmute fire and the track to be muted.] - expected: FAIL + expected: NOTRUN [addTrack() with a track and two streams makes ontrack fire with a track and two streams.] - expected: FAIL + expected: NOTRUN [stream.onremovetrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [removeTrack() makes stream.onremovetrack fire and the track to be removed from the stream.] - expected: FAIL + expected: NOTRUN [removeTrack() does not remove the receiver.] - expected: FAIL + expected: NOTRUN [removeTrack() twice is safe.] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini index 5ec70d9a9320..0a0df27d0127 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription.html.ini @@ -5,12 +5,3 @@ [Switching role from offerer to answerer after going back to stable state should succeed] expected: FAIL - [Negotiation should fire signalingsstate events] - expected: FAIL - - [setRemoteDescription() with invalid SDP and stable state should reject with InvalidStateError] - expected: FAIL - - [setRemoteDescription with invalid type and invalid SDP should reject with TypeError] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini index cfca4a6ac221..404bb86a9486 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-track-stats.https.html] + expected: ERROR [replaceTrack() before offer: new track attachment stats present] expected: FAIL @@ -12,7 +13,7 @@ expected: FAIL [RTCRtpReceiver.getStats() contains only inbound-rtp and related stats] - expected: FAIL + expected: NOTRUN [O/A exchange yields inbound RTP stream stats for receiving track] expected: FAIL @@ -33,19 +34,19 @@ expected: FAIL [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are zero senders or receivers for the track] - expected: FAIL + expected: NOTRUN [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are multiple senders for the track] - expected: FAIL + expected: NOTRUN [replaceTrack() after answer: new track attachment stats present] expected: FAIL [RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats()] - expected: FAIL + expected: NOTRUN [RTCPeerConnection.getStats(sendingTrack) is the same as RTCRtpSender.getStats()] - expected: FAIL + expected: NOTRUN [addTrack() with setLocalDescription() yields track stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini index f630372bc8f0..ffce6fe8e7ad 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -1,133 +1,134 @@ [RTCPeerConnection-transceivers.https.html] + expected: ERROR [setLocalDescription(answer): transceiver.currentDirection is recvonly] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack fires with a track] - expected: FAIL + expected: NOTRUN [Can setup two-way call using a single transceiver] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.mid is the same on both ends] - expected: FAIL + expected: NOTRUN [addTransceiver(track, init): initialize sendEncodings[0\].active to false] - expected: FAIL + expected: NOTRUN [addTransceiver('video'): transceiver.receiver.track.kind == 'video'] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.stopped is false] - expected: FAIL + expected: NOTRUN [setLocalDescription(answer): transceiver.currentDirection is sendonly] - expected: FAIL + expected: NOTRUN [addTrack: transceiver is not associated with an m-section] - expected: FAIL + expected: NOTRUN [transceiver.sender.track does not revert to an old state] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.receiver has its own track] - expected: FAIL + expected: NOTRUN [addTrack: "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [setLocalDescription(offer): transceiver.mid matches the offer SDP] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.direction is recvonly] - expected: FAIL + expected: NOTRUN [Changing transceiver direction to 'sendrecv' makes ontrack fire] - expected: FAIL + expected: NOTRUN [addTrack: transceiver is not stopped] - expected: FAIL + expected: NOTRUN [addTrack(1 stream): ontrack fires with corresponding stream] - expected: FAIL + expected: NOTRUN [addTransceiver(track, init): initialize direction to inactive] - expected: FAIL + expected: NOTRUN [addTrack(0 streams): ontrack fires with no stream] - expected: FAIL + expected: NOTRUN [addTrack: transceiver's direction is sendrecv] - expected: FAIL + expected: NOTRUN [addTrack(2 streams): ontrack fires with corresponding two streams] - expected: FAIL + expected: NOTRUN [addTransceiver(0 streams): ontrack fires with no stream] - expected: FAIL + expected: NOTRUN [addTransceiver(1 stream): ontrack fires with corresponding stream] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.currentDirection is null] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.sender is associated with the track] - expected: FAIL + expected: NOTRUN [addTransceiver does not reuse reusable transceivers] - expected: FAIL + expected: NOTRUN [addTransceiver(track): "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [transceiver.direction does not revert to an old state] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.sender.track == null] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): creates a transceiver with direction sendrecv] - expected: FAIL + expected: NOTRUN [setLocalDescription(offer): transceiver gets associated with an m-section] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.currentDirection is null] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack's stream.id is the same as stream.id] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.stopped is false] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.receiver.track.kind == 'audio'] - expected: FAIL + expected: NOTRUN [addTransceiver(track): creates a transceiver for the track] - expected: FAIL + expected: NOTRUN [addTransceiver(2 streams): ontrack fires with corresponding two streams] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [addTrack: transceiver's currentDirection is null] - expected: FAIL + expected: NOTRUN [addTrack reuses reusable transceivers] - expected: FAIL + expected: NOTRUN [addTrack: creates a transceiver for the sender] expected: FAIL [Closing the PC stops the transceivers] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack fires with a transceiver.] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.receiver's track is muted] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini index dfb88485f887..043a9c242eb9 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnectionIceEvent-constructor.html.ini @@ -1,25 +1,4 @@ [RTCPeerConnectionIceEvent-constructor.html] - [RTCPeerConnectionIceEvent.candidate is null when constructed with { candidate: null }] - expected: FAIL - - [RTCPeerConnectionIceEvent with empty object as eventInitDict (default)] - expected: FAIL - - [RTCPeerConnectionIceEvent.candidate is null when constructed with { candidate: undefined }] - expected: FAIL - - [RTCPeerConnectionIceEvent with RTCIceCandidate] - expected: FAIL - - [RTCPeerConnectionIceEvent with non RTCIceCandidate object throws] - expected: FAIL - - [RTCPeerConnectionIceEvent with no arguments throws TypeError] - expected: FAIL - [RTCPeerConnectionIceEvent bubbles and cancelable] expected: FAIL - [RTCPeerConnectionIceEvent with no eventInitDict (default)] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini index ff336963390c..f8ec71028533 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini @@ -1,7 +1,8 @@ [RTCRtpReceiver-getContributingSources.https.html] + expected: ERROR [[audio\] getContributingSources() returns an empty list in loopback call] expected: FAIL [[video\] getContributingSources() returns an empty list in loopback call] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini index a69a3eda57bd..0d9402476431 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -1,6 +1,7 @@ [RTCRtpReceiver-getStats.https.html] + expected: ERROR [receiver.getStats() via addTrack should return stats report containing inbound-rtp stats] - expected: FAIL + expected: NOTRUN [receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini index a5a15242ac4c..18a8e0fef6a1 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -1,37 +1,38 @@ [RTCRtpReceiver-getSynchronizationSources.https.html] + expected: ERROR [[audio\] RTCRtpSynchronizationSource.source is a number] - expected: FAIL + expected: NOTRUN [[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: FAIL + expected: NOTRUN [[audio\] getSynchronizationSources() eventually returns a non-empty list] expected: FAIL [[video\] RTCRtpSynchronizationSource.timestamp is a number] - expected: FAIL + expected: NOTRUN [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean] - expected: FAIL + expected: NOTRUN [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: FAIL + expected: NOTRUN [[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: FAIL + expected: NOTRUN [[video\] getSynchronizationSources() eventually returns a non-empty list] - expected: FAIL + expected: NOTRUN [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: FAIL + expected: NOTRUN [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]] - expected: FAIL + expected: NOTRUN [[video\] RTCRtpSynchronizationSource.source is a number] - expected: FAIL + expected: NOTRUN [[audio\] RTCRtpSynchronizationSource.timestamp is a number] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini index 4256534ee3e2..924a89b73025 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini @@ -1,6 +1,7 @@ [RTCRtpSender-getStats.https.html] + expected: ERROR [sender.getStats() via addTrack should return stats report containing outbound-rtp stats] - expected: FAIL + expected: NOTRUN [sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini index 821925718a1d..04f02b467342 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini @@ -1,28 +1,29 @@ [RTCRtpSender-replaceTrack.https.html] + expected: ERROR [Calling replaceTrack on sender with null track and not set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack(null) on sender not set to session description should resolve with sender.track set to null] - expected: FAIL + expected: NOTRUN [Calling replaceTrack with track of different kind should reject with TypeError] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on sender with similar track and and set to session description should resolve with sender.track set to new track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on closed connection should reject with InvalidStateError] expected: FAIL [Calling replaceTrack on sender with stopped track and and set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on stopped sender should reject with InvalidStateError] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on sender not set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack(null) on sender set to session description should resolve with sender.track set to null] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini index fac8652e7aa7..200008065aef 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini @@ -1,15 +1,16 @@ [RTCRtpSender-transport.https.html] + expected: ERROR [RTCRtpSender/receiver.transport at the right time, with bundle policy balanced] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport has a value when connected] - expected: FAIL + expected: NOTRUN [RTCRtpSender.transport is null when unconnected] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini index c65b2f802864..2958e0a5949f 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -1,109 +1,110 @@ [RTCRtpTransceiver.https.html] + expected: ERROR [checkAddTransceiverWithTrack] expected: FAIL [checkNoMidAnswer] - expected: FAIL + expected: NOTRUN [checkSetDirection] - expected: FAIL + expected: NOTRUN [checkCurrentDirection] - expected: FAIL + expected: NOTRUN [checkReplaceTrackNullDoesntPreventPairing] - expected: FAIL + expected: NOTRUN [checkAddTrackExistingTransceiverThenRemove] - expected: FAIL + expected: NOTRUN [checkMsectionReuse] - expected: FAIL + expected: NOTRUN [checkRemoteRollback] - expected: FAIL + expected: NOTRUN [checkAddTransceiverNoTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkRemoveTrackNegotiation] - expected: FAIL + expected: NOTRUN [checkStopAfterClose] - expected: FAIL + expected: NOTRUN [checkNoMidOffer] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateOffer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverThenAddTrackPairs] - expected: FAIL + expected: NOTRUN [checkLocalRollback] - expected: FAIL + expected: NOTRUN [checkSendrecvWithTracklessStream] - expected: FAIL + expected: NOTRUN [checkMsidNoTrackId] - expected: FAIL + expected: NOTRUN [checkAddTransceiverNoTrack] expected: FAIL [checkAddTransceiverBadKind] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateAnswer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithDirection] - expected: FAIL + expected: NOTRUN [checkAddTrackPairs] - expected: FAIL + expected: NOTRUN [checkStopAfterSetLocalAnswer] - expected: FAIL + expected: NOTRUN [checkStopAfterSetRemoteOffer] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateOfferWithReusedMsection] - expected: FAIL + expected: NOTRUN [checkMute] - expected: FAIL + expected: NOTRUN [checkSendrecvWithNoSendTrack] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithSetRemoteOfferSending] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithAddTrack] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithSetRemoteOfferNoSend] - expected: FAIL + expected: NOTRUN [checkRemoveAndReadd] - expected: FAIL + expected: NOTRUN [checkStopAfterSetLocalOffer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkAddTransceiverThenReplaceTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkRollbackAndSetRemoteOfferWithDifferentType] - expected: FAIL + expected: NOTRUN [checkStop] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini index 2fa5e125c03e..59fbbba131fc 100644 --- a/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini +++ b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini @@ -1,7 +1,8 @@ [RTCTrackEvent-fire.html] + expected: TIMEOUT [Applying a remote description with removed msid should trigger firing a removetrack event on the corresponding stream] - expected: FAIL + expected: TIMEOUT [Applying a remote description with a new msid should trigger firing an event with populated streams] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/__dir__.ini b/tests/wpt/metadata/webrtc/__dir__.ini new file mode 100644 index 000000000000..40d8e8883654 --- /dev/null +++ b/tests/wpt/metadata/webrtc/__dir__.ini @@ -0,0 +1 @@ +prefs: ["dom.webrtc.enabled:true"] diff --git a/tests/wpt/metadata/webrtc/historical.html.ini b/tests/wpt/metadata/webrtc/historical.html.ini index 9e0a93addfa3..d4de2a34fa49 100644 --- a/tests/wpt/metadata/webrtc/historical.html.ini +++ b/tests/wpt/metadata/webrtc/historical.html.ini @@ -1,37 +1,13 @@ [historical.html] - [RTCPeerConnection member removeStream should not exist] - expected: FAIL - [RTCRtpTransceiver member setDirection should not exist] expected: FAIL [RTCDataChannel member reliable should not exist] expected: FAIL - [RTCPeerConnection member getStreamById should not exist] - expected: FAIL - - [RTCPeerConnection member onremovestream should not exist] - expected: FAIL - [RTCPeerConnection member addStream should not exist] expected: FAIL - [RTCPeerConnection member getLocalStreams should not exist] - expected: FAIL - - [RTCPeerConnection member getRemoteStreams should not exist] - expected: FAIL - - [RTCPeerConnection member updateIce should not exist] - expected: FAIL - [RTCDataChannel member maxRetransmitTime should not exist] expected: FAIL - [RTCPeerConnection member onaddstream should not exist] - expected: FAIL - - [RTCPeerConnection member createDTMFSender should not exist] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini index 0a145552b6e7..f06ce24b7e81 100644 --- a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini +++ b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini @@ -1,10 +1,4 @@ [idlharness.https.window.html] - [RTCPeerConnection interface: attribute signalingState] - expected: FAIL - - [RTCIceCandidate interface: existence and properties of interface object] - expected: FAIL - [RTCRtpSender interface: attribute track] expected: FAIL @@ -26,21 +20,12 @@ [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences([object Object\])" with the proper type] expected: FAIL - [RTCPeerConnection interface: calling createAnswer(RTCAnswerOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getReceivers()" with the proper type] expected: FAIL [RTCDataChannel interface: existence and properties of interface object] expected: FAIL - [RTCSessionDescription interface: existence and properties of interface prototype object's @@unscopables property] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getDefaultIceServers()" with the proper type] - expected: FAIL - [RTCSctpTransport interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL @@ -122,24 +107,12 @@ [RTCRtpSender interface: operation setStreams(MediaStream)] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] - expected: FAIL - [RTCPeerConnection interface: operation getConfiguration()] expected: FAIL - [RTCPeerConnectionIceEvent interface: new RTCPeerConnectionIceEvent('ice') must inherit property "candidate" with the proper type] - expected: FAIL - - [RTCIceCandidate must be primary interface of new RTCIceCandidate({ sdpMid: 1 })] - expected: FAIL - [RTCRtpTransceiver interface: existence and properties of interface object] expected: FAIL - [RTCSessionDescription must be primary interface of new RTCSessionDescription({ type: 'offer' })] - expected: FAIL - [RTCPeerConnection interface: attribute ondatachannel] expected: FAIL @@ -158,36 +131,18 @@ [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "ongatheringstatechange" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "signalingState" with the proper type] - expected: FAIL - - [RTCIceCandidate interface object length] - expected: FAIL - [RTCDataChannel interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL - [RTCIceCandidate interface: attribute candidate] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "ordered" with the proper type] expected: FAIL [RTCCertificate interface: existence and properties of interface object] expected: FAIL - [RTCPeerConnection interface: existence and properties of interface prototype object's @@unscopables property] - expected: FAIL - - [RTCTrackEvent interface: existence and properties of interface object] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "maxPacketLifeTime" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicecandidate" with the proper type] - expected: FAIL - [RTCIceCandidate interface: attribute component] expected: FAIL @@ -200,9 +155,6 @@ [Test driver for asyncInitCertificate] expected: FAIL - [RTCPeerConnection interface: existence and properties of interface object] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "id" with the proper type] expected: FAIL @@ -224,9 +176,6 @@ [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onmessage" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback)" with the proper type] - expected: FAIL - [RTCRtpTransceiver interface object length] expected: FAIL @@ -278,9 +227,6 @@ [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "onstatechange" with the proper type] expected: FAIL - [Stringification of new RTCSessionDescription({ type: 'offer' })] - expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "removeTrack(RTCRtpSender)" with the proper type] expected: FAIL @@ -296,12 +242,6 @@ [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onstatsended" with the proper type] expected: FAIL - [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "sdp" with the proper type] - expected: FAIL - - [RTCSessionDescription interface: attribute sdp] - expected: FAIL - [RTCDTMFSender interface: attribute canInsertDTMF] expected: FAIL @@ -311,12 +251,6 @@ [RTCRtpReceiver interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').receiver with too few arguments must throw TypeError] expected: FAIL - [RTCSessionDescription interface object name] - expected: FAIL - - [RTCPeerConnection interface object name] - expected: FAIL - [RTCRtpReceiver interface: operation getSynchronizationSources()] expected: FAIL @@ -326,15 +260,9 @@ [RTCPeerConnection interface: attribute onconnectionstatechange] expected: FAIL - [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object's @@unscopables property] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(ArrayBuffer)" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicegatheringstatechange" with the proper type] - expected: FAIL - [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "component" with the proper type] expected: FAIL @@ -359,18 +287,12 @@ [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getRemoteParameters()" with the proper type] expected: FAIL - [RTCPeerConnection interface: attribute iceGatheringState] - expected: FAIL - [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getCapabilities(DOMString)" with the proper type] expected: FAIL [RTCDTMFToneChangeEvent interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL - [RTCTrackEvent interface: existence and properties of interface prototype object] - expected: FAIL - [RTCRtpTransceiver interface: calling setCodecPreferences([object Object\]) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError] expected: FAIL @@ -383,18 +305,12 @@ [RTCIceTransport interface: attribute gatheringState] expected: FAIL - [RTCSessionDescription interface: default toJSON operation on new RTCSessionDescription({ type: 'offer' })] - expected: FAIL - [RTCErrorEvent interface: existence and properties of interface object] expected: FAIL [RTCPeerConnection interface: operation createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback)] expected: FAIL - [RTCPeerConnection interface: calling createOffer(RTCOfferOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCRtpReceiver interface: operation getCapabilities(DOMString)] expected: FAIL @@ -407,9 +323,6 @@ [RTCRtpTransceiver interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onnegotiationneeded" with the proper type] - expected: FAIL - [RTCPeerConnection interface: attribute pendingRemoteDescription] expected: FAIL @@ -425,18 +338,9 @@ [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "protocol" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setRemoteDescription(RTCSessionDescriptionInit)" with the proper type] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createOffer(RTCOfferOptions)" with the proper type] - expected: FAIL - [RTCRtpTransceiver interface: attribute direction] expected: FAIL - [RTCIceCandidate interface: existence and properties of interface prototype object's "constructor" property] - expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getTransceivers()" with the proper type] expected: FAIL @@ -452,33 +356,12 @@ [RTCError interface: attribute errorDetail] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "remoteDescription" with the proper type] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createAnswer(RTCAnswerOptions)" with the proper type] - expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedPort" with the proper type] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "usernameFragment" with the proper type] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "generateCertificate(AlgorithmIdentifier)" with the proper type] - expected: FAIL - - [Stringification of new RTCIceCandidate({ sdpMid: 1 })] - expected: FAIL - - [RTCPeerConnection interface: calling addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "tcpType" with the proper type] expected: FAIL - [RTCPeerConnectionIceEvent interface: new RTCPeerConnectionIceEvent('ice') must inherit property "url" with the proper type] - expected: FAIL - [RTCErrorEvent interface object length] expected: FAIL @@ -491,9 +374,6 @@ [RTCRtpSender interface object length] expected: FAIL - [RTCIceCandidate interface: toJSON operation on new RTCIceCandidate({ sdpMid: 1 })] - expected: FAIL - [RTCPeerConnectionIceErrorEvent interface object length] expected: FAIL @@ -515,24 +395,9 @@ [RTCStatsReport interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL - [RTCTrackEvent interface: attribute track] - expected: FAIL - - [RTCIceCandidate interface object name] - expected: FAIL - [RTCPeerConnection interface: operation addIceCandidate(RTCIceCandidateInit, VoidFunction, RTCPeerConnectionErrorCallback)] expected: FAIL - [RTCTrackEvent interface: existence and properties of interface prototype object's @@unscopables property] - expected: FAIL - - [RTCPeerConnection interface: attribute onnegotiationneeded] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "oniceconnectionstatechange" with the proper type] - expected: FAIL - [RTCSctpTransport interface: attribute transport] expected: FAIL @@ -551,9 +416,6 @@ [RTCDtlsTransport interface: idlTestObjects.dtlsTransport must inherit property "state" with the proper type] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "toJSON()" with the proper type] - expected: FAIL - [RTCRtpReceiver interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL @@ -563,9 +425,6 @@ [RTCDataChannelEvent interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL - [RTCIceCandidate interface: existence and properties of interface prototype object] - expected: FAIL - [RTCRtpSender interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] expected: FAIL @@ -584,9 +443,6 @@ [Stringification of new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 });] expected: FAIL - [RTCPeerConnection interface: attribute onicecandidate] - expected: FAIL - [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalParameters()" with the proper type] expected: FAIL @@ -596,9 +452,6 @@ [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getStats()" with the proper type] expected: FAIL - [RTCPeerConnection interface: operation close()] - expected: FAIL - [RTCDataChannel interface: attribute protocol] expected: FAIL @@ -614,21 +467,12 @@ [RTCIceTransport interface: attribute state] expected: FAIL - [RTCPeerConnectionIceEvent interface object name] - expected: FAIL - [RTCDataChannel interface: attribute onerror] expected: FAIL [RTCRtpSender interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL - [RTCPeerConnection interface: attribute localDescription] - expected: FAIL - - [RTCPeerConnection interface: existence and properties of interface prototype object's "constructor" property] - expected: FAIL - [RTCPeerConnection interface: operation createOffer(RTCOfferOptions)] expected: FAIL @@ -674,9 +518,6 @@ [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "onbufferedamountlow" with the proper type] expected: FAIL - [RTCSessionDescription interface object length] - expected: FAIL - [RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "mid" with the proper type] expected: FAIL @@ -704,9 +545,6 @@ [RTCStatsEvent interface: existence and properties of interface object] expected: FAIL - [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "toJSON()" with the proper type] - expected: FAIL - [RTCStatsReport interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL @@ -719,9 +557,6 @@ [Stringification of idlTestObjects.dtlsTransport] expected: FAIL - [RTCPeerConnectionIceEvent interface object length] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "binaryType" with the proper type] expected: FAIL @@ -746,9 +581,6 @@ [RTCDTMFSender interface object length] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "close()" with the proper type] - expected: FAIL - [RTCIceCandidate interface: attribute address] expected: FAIL @@ -770,27 +602,15 @@ [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getSelectedCandidatePair()" with the proper type] expected: FAIL - [Stringification of new RTCPeerConnectionIceEvent('ice')] - expected: FAIL - [RTCDataChannel interface: operation close()] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "sdpMLineIndex" with the proper type] - expected: FAIL - [RTCPeerConnection interface: calling removeTrack(RTCRtpSender) on new RTCPeerConnection() with too few arguments must throw TypeError] expected: FAIL - [RTCIceCandidate interface: operation toJSON()] - expected: FAIL - [RTCDataChannelEvent interface object length] expected: FAIL - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "candidate" with the proper type] - expected: FAIL - [RTCErrorEvent interface: attribute error] expected: FAIL @@ -809,9 +629,6 @@ [RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getSynchronizationSources()" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] - expected: FAIL - [RTCIceTransport interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL @@ -821,9 +638,6 @@ [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "url" with the proper type] expected: FAIL - [RTCPeerConnection interface: attribute onsignalingstatechange] - expected: FAIL - [RTCPeerConnection interface: operation getStats(MediaStreamTrack)] expected: FAIL @@ -833,9 +647,6 @@ [RTCRtpSender interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL - [RTCPeerConnection interface: attribute onicegatheringstatechange] - expected: FAIL - [RTCCertificate interface: operation getFingerprints()] expected: FAIL @@ -854,9 +665,6 @@ [RTCSctpTransport interface: attribute maxMessageSize] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setLocalDescription(RTCSessionDescriptionInit)" with the proper type] - expected: FAIL - [RTCRtpReceiver interface object length] expected: FAIL @@ -872,9 +680,6 @@ [RTCRtpTransceiver interface: attribute receiver] expected: FAIL - [RTCIceCandidate interface: attribute usernameFragment] - expected: FAIL - [RTCDTMFToneChangeEvent interface object length] expected: FAIL @@ -887,9 +692,6 @@ [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "rtcpTransport" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "ontrack" with the proper type] - expected: FAIL - [RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)] expected: FAIL @@ -923,27 +725,12 @@ [RTCDTMFSender interface: existence and properties of interface prototype object's "constructor" property] expected: FAIL - [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object] - expected: FAIL - - [RTCPeerConnection interface object length] - expected: FAIL - [RTCDataChannel interface: attribute label] expected: FAIL - [Test driver for asyncInitMediaStreamTrack] - expected: FAIL - - [RTCPeerConnection interface: calling createAnswer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCPeerConnection interface: operation setConfiguration(RTCConfiguration)] expected: FAIL - [RTCPeerConnection interface: calling createOffer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback, RTCOfferOptions) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object] expected: FAIL @@ -974,9 +761,6 @@ [RTCTrackEvent interface: attribute streams] expected: FAIL - [RTCPeerConnection interface: calling addIceCandidate(RTCIceCandidateInit) on new RTCPeerConnection() with too few arguments must throw TypeError] - expected: FAIL - [RTCRtpSender interface: operation getParameters()] expected: FAIL @@ -989,18 +773,9 @@ [RTCIceTransport interface: attribute onstatechange] expected: FAIL - [RTCPeerConnection interface: attribute iceConnectionState] - expected: FAIL - [RTCDataChannel interface: attribute id] expected: FAIL - [RTCTrackEvent interface object length] - expected: FAIL - - [RTCPeerConnection interface: attribute ontrack] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "negotiated" with the proper type] expected: FAIL @@ -1013,9 +788,6 @@ [RTCDTMFToneChangeEvent interface: attribute tone] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "localDescription" with the proper type] - expected: FAIL - [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "setParameters(RTCRtpSendParameters)" with the proper type] expected: FAIL @@ -1034,18 +806,9 @@ [RTCStatsEvent interface: existence and properties of interface prototype object's @@unscopables property] expected: FAIL - [RTCPeerConnectionIceEvent interface: existence and properties of interface prototype object's "constructor" property] - expected: FAIL - [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "role" with the proper type] expected: FAIL - [RTCPeerConnectionIceEvent interface: existence and properties of interface object] - expected: FAIL - - [RTCPeerConnectionIceEvent interface: attribute candidate] - expected: FAIL - [RTCPeerConnection interface: attribute connectionState] expected: FAIL @@ -1064,9 +827,6 @@ [RTCError interface: attribute httpRequestStatusCode] expected: FAIL - [RTCTrackEvent interface object name] - expected: FAIL - [RTCRtpSender interface: operation getStats()] expected: FAIL @@ -1079,9 +839,6 @@ [Stringification of idlTestObjects.iceTransport] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "createOffer(RTCSessionDescriptionCallback, RTCPeerConnectionErrorCallback, RTCOfferOptions)" with the proper type] - expected: FAIL - [RTCPeerConnection interface: calling setConfiguration(RTCConfiguration) on new RTCPeerConnection() with too few arguments must throw TypeError] expected: FAIL @@ -1100,9 +857,6 @@ [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "getLocalCandidates()" with the proper type] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setLocalDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)" with the proper type] - expected: FAIL - [RTCDataChannel interface: attribute onopen] expected: FAIL @@ -1112,9 +866,6 @@ [RTCIceCandidate interface: attribute tcpType] expected: FAIL - [RTCTrackEvent interface: existence and properties of interface prototype object's "constructor" property] - expected: FAIL - [RTCRtpReceiver interface: attribute rtcpTransport] expected: FAIL @@ -1127,24 +878,15 @@ [RTCPeerConnection interface: calling getStats(MediaStreamTrack) on new RTCPeerConnection() with too few arguments must throw TypeError] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "iceConnectionState" with the proper type] - expected: FAIL - [RTCError interface: attribute sdpLineNumber] expected: FAIL [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "bufferedAmount" with the proper type] expected: FAIL - [RTCSessionDescription interface: existence and properties of interface prototype object's "constructor" property] - expected: FAIL - [RTCDtlsTransport interface: attribute state] expected: FAIL - [RTCPeerConnection must be primary interface of new RTCPeerConnection()] - expected: FAIL - [RTCError interface: attribute receivedAlert] expected: FAIL @@ -1160,12 +902,6 @@ [RTCIceCandidate interface: attribute priority] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "addIceCandidate(RTCIceCandidateInit)" with the proper type] - expected: FAIL - - [RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "sdpMid" with the proper type] - expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "currentRemoteDescription" with the proper type] expected: FAIL @@ -1187,9 +923,6 @@ [RTCDataChannel interface object length] expected: FAIL - [RTCSessionDescription interface: existence and properties of interface object] - expected: FAIL - [RTCRtpSender interface: calling setParameters(RTCRtpSendParameters) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError] expected: FAIL @@ -1232,27 +965,15 @@ [RTCPeerConnection interface: calling setRemoteDescription(RTCSessionDescriptionInit) on new RTCPeerConnection() with too few arguments must throw TypeError] expected: FAIL - [Stringification of new RTCPeerConnection()] - expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "getConfiguration()" with the proper type] expected: FAIL - [RTCPeerConnectionIceEvent interface: attribute url] - expected: FAIL - [RTCDataChannel must be primary interface of new RTCPeerConnection().createDataChannel('')] expected: FAIL [RTCStatsEvent interface object length] expected: FAIL - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "iceGatheringState" with the proper type] - expected: FAIL - - [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onsignalingstatechange" with the proper type] - expected: FAIL - [RTCDTMFToneChangeEvent interface object name] expected: FAIL @@ -1262,15 +983,9 @@ [RTCIceTransport interface: existence and properties of interface object] expected: FAIL - [RTCPeerConnection interface: attribute remoteDescription] - expected: FAIL - [RTCIceTransport interface: attribute role] expected: FAIL - [RTCIceCandidate interface: attribute sdpMLineIndex] - expected: FAIL - [RTCDataChannelEvent interface: existence and properties of interface prototype object] expected: FAIL @@ -1286,9 +1001,6 @@ [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "currentLocalDescription" with the proper type] expected: FAIL - [RTCPeerConnection interface: existence and properties of interface prototype object] - expected: FAIL - [Stringification of new RTCPeerConnection().addTransceiver('audio').receiver] expected: FAIL @@ -1304,9 +1016,6 @@ [RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "replaceTrack(MediaStreamTrack)" with the proper type] expected: FAIL - [RTCSessionDescription interface: new RTCSessionDescription({ type: 'offer' }) must inherit property "type" with the proper type] - expected: FAIL - [RTCDtlsTransport interface: attribute onstatechange] expected: FAIL @@ -1343,15 +1052,9 @@ [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "state" with the proper type] expected: FAIL - [RTCPeerConnection interface: attribute oniceconnectionstatechange] - expected: FAIL - [RTCDataChannel interface: attribute binaryType] expected: FAIL - [RTCSessionDescription interface: attribute type] - expected: FAIL - [RTCIceTransport interface: idlTestObjects.iceTransport must inherit property "state" with the proper type] expected: FAIL @@ -1373,24 +1076,15 @@ [RTCStatsEvent interface object name] expected: FAIL - [RTCIceCandidate interface: attribute sdpMid] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(ArrayBufferView)" with the proper type] expected: FAIL - [RTCPeerConnectionIceEvent must be primary interface of new RTCPeerConnectionIceEvent('ice')] - expected: FAIL - [RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { errorCode: 701 }); must inherit property "hostCandidate" with the proper type] expected: FAIL [RTCDataChannel interface: operation send(ArrayBuffer)] expected: FAIL - [RTCSessionDescription interface: operation toJSON()] - expected: FAIL - [RTCDataChannel interface: new RTCPeerConnection().createDataChannel('') must inherit property "send(USVString)" with the proper type] expected: FAIL @@ -1403,9 +1097,6 @@ [RTCRtpTransceiver interface: attribute currentDirection] expected: FAIL - [RTCIceCandidate interface: existence and properties of interface prototype object's @@unscopables property] - expected: FAIL - [RTCSctpTransport interface: idlTestObjects.sctpTransport must inherit property "maxMessageSize" with the proper type] expected: FAIL @@ -1472,9 +1163,6 @@ [RTCPeerConnection interface: operation setLocalDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback)] expected: FAIL - [RTCSessionDescription interface: existence and properties of interface prototype object] - expected: FAIL - [RTCDtlsTransport interface: attribute onerror] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini index 841c26c5f9a3..b9dadb679850 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -1,13 +1,14 @@ [RTCRtpTransceiver-with-OfferToReceive-options.https.html] + expected: ERROR [checkAddTransceiverWithStream] expected: FAIL [checkAddTransceiverWithOfferToReceiveVideo] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithOfferToReceiveBoth] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithOfferToReceiveAudio] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini index 71b4cc138f65..7923ea02036a 100644 --- a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -1,4 +1,5 @@ [onaddstream.https.html] + expected: ERROR [Check onaddstream] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/no-media-call.html.ini b/tests/wpt/metadata/webrtc/no-media-call.html.ini index 5a3fd876b999..95345d5794b2 100644 --- a/tests/wpt/metadata/webrtc/no-media-call.html.ini +++ b/tests/wpt/metadata/webrtc/no-media-call.html.ini @@ -1,4 +1,5 @@ [no-media-call.html] + expected: TIMEOUT [Can set up a basic WebRTC call with no data.] - expected: FAIL + expected: TIMEOUT diff --git a/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini b/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini deleted file mode 100644 index 8ee2a670e815..000000000000 --- a/tests/wpt/metadata/webrtc/protocol/jsep-initial-offer.https.html.ini +++ /dev/null @@ -1,4 +0,0 @@ -[jsep-initial-offer.https.html] - [Offer conforms to basic SDP requirements] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini b/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini index 260513f58fbb..ef091feba6a3 100644 --- a/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini +++ b/tests/wpt/metadata/webrtc/protocol/msid-parse.html.ini @@ -1,13 +1,14 @@ [msid-parse.html] + expected: TIMEOUT [Description with no msid produces a track with a stream] - expected: FAIL + expected: TIMEOUT [Description with two msid produces two streams] - expected: FAIL + expected: NOTRUN [Description with msid:foo bar produces a stream with id foo] - expected: FAIL + expected: NOTRUN [Description with msid:- appid produces a track with no stream] - expected: FAIL + expected: NOTRUN From b6acfaca0712c8a1c39f578e8ae29d729c3905a9 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 12:11:47 -0700 Subject: [PATCH 06/13] Enable webrtc pref by default, split getUserMedia into mediadevices pref --- components/config/prefs.rs | 4 ++++ components/script/dom/webidls/MediaDevices.webidl | 4 ++-- components/script/dom/webidls/MediaStream.webidl | 3 +-- components/script/dom/webidls/MediaStreamTrack.webidl | 2 +- resources/prefs.json | 3 ++- .../webrtc/RTCPeerConnection-onnegotiationneeded.html.ini | 3 +++ tests/wpt/metadata/webrtc/__dir__.ini | 2 +- tests/wpt/mozilla/meta/MANIFEST.json | 2 +- tests/wpt/mozilla/tests/mozilla/interfaces.html | 7 +++++++ 9 files changed, 22 insertions(+), 8 deletions(-) diff --git a/components/config/prefs.rs b/components/config/prefs.rs index 76e70933bde7..3bb24fb7e11e 100644 --- a/components/config/prefs.rs +++ b/components/config/prefs.rs @@ -190,6 +190,10 @@ mod gen { gamepad: { enabled: bool, }, + mediadevices: { + #[serde(default)] + enabled: bool, + }, microdata: { testing: { enabled: bool, diff --git a/components/script/dom/webidls/MediaDevices.webidl b/components/script/dom/webidls/MediaDevices.webidl index 866e1e964d0c..44cd0a6592be 100644 --- a/components/script/dom/webidls/MediaDevices.webidl +++ b/components/script/dom/webidls/MediaDevices.webidl @@ -5,7 +5,7 @@ // https://w3c.github.io/mediacapture-main/#dom-mediadevices [Exposed=Window, -SecureContext, Pref="dom.webrtc.enabled"] +SecureContext, Pref="dom.mediadevices.enabled"] interface MediaDevices : EventTarget { // attribute EventHandler ondevicechange; // Promise> enumerateDevices(); @@ -13,7 +13,7 @@ interface MediaDevices : EventTarget { partial interface Navigator { // [SameObject, SecureContext] - [Pref="dom.webrtc.enabled"] readonly attribute MediaDevices mediaDevices; + [Pref="dom.mediadevices.enabled"] readonly attribute MediaDevices mediaDevices; }; partial interface MediaDevices { diff --git a/components/script/dom/webidls/MediaStream.webidl b/components/script/dom/webidls/MediaStream.webidl index b1d5664a7f7f..f2e700a799c9 100644 --- a/components/script/dom/webidls/MediaStream.webidl +++ b/components/script/dom/webidls/MediaStream.webidl @@ -7,8 +7,7 @@ [Exposed=Window, Constructor, Constructor(MediaStream stream), - Constructor(sequence tracks), -Pref="dom.webrtc.enabled"] + Constructor(sequence tracks)] interface MediaStream : EventTarget { // readonly attribute DOMString id; sequence getAudioTracks(); diff --git a/components/script/dom/webidls/MediaStreamTrack.webidl b/components/script/dom/webidls/MediaStreamTrack.webidl index 2f8bfb0bbece..b514360622af 100644 --- a/components/script/dom/webidls/MediaStreamTrack.webidl +++ b/components/script/dom/webidls/MediaStreamTrack.webidl @@ -4,7 +4,7 @@ // https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack -[Exposed=Window, Pref="dom.webrtc.enabled"] +[Exposed=Window] interface MediaStreamTrack : EventTarget { readonly attribute DOMString kind; readonly attribute DOMString id; diff --git a/resources/prefs.json b/resources/prefs.json index 23c42c3352da..56d11358d9bb 100644 --- a/resources/prefs.json +++ b/resources/prefs.json @@ -9,6 +9,7 @@ "dom.forcetouch.enabled": false, "dom.fullscreen.test": false, "dom.gamepad.enabled": false, + "dom.mediadevices.enabled": false, "dom.microdata.enabled": false, "dom.microdata.testing.enabled": false, "dom.mouseevent.which.enabled": false, @@ -26,7 +27,7 @@ "dom.testing.htmlinputelement.select_files.enabled": false, "dom.webgl.dom_to_texture.enabled": false, "dom.webgl2.enabled": false, - "dom.webrtc.enabled": false, + "dom.webrtc.enabled": true, "dom.webvr.enabled": false, "dom.webvr.event_polling_interval": 500, "dom.webvr.test": false, diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index 434392114458..ac85dc6b8990 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -33,3 +33,6 @@ [calling createDataChannel twice should fire negotiationneeded event once] expected: FAIL + [negotiationneeded event should not fire if signaling state is not stable] + expected: TIMEOUT + diff --git a/tests/wpt/metadata/webrtc/__dir__.ini b/tests/wpt/metadata/webrtc/__dir__.ini index 40d8e8883654..99550e100683 100644 --- a/tests/wpt/metadata/webrtc/__dir__.ini +++ b/tests/wpt/metadata/webrtc/__dir__.ini @@ -1 +1 @@ -prefs: ["dom.webrtc.enabled:true"] +prefs: ["dom.webrtc.enabled:true", "dom.mediadevices.enabled:true"] diff --git a/tests/wpt/mozilla/meta/MANIFEST.json b/tests/wpt/mozilla/meta/MANIFEST.json index f12353058f91..67f210953085 100644 --- a/tests/wpt/mozilla/meta/MANIFEST.json +++ b/tests/wpt/mozilla/meta/MANIFEST.json @@ -20306,7 +20306,7 @@ "testharness" ], "mozilla/interfaces.html": [ - "2effd46f565c4787e8632f5e898e1b43d457672f", + "c3941a5f90824f1081904a42b0d189bad73a8b73", "testharness" ], "mozilla/interfaces.js": [ diff --git a/tests/wpt/mozilla/tests/mozilla/interfaces.html b/tests/wpt/mozilla/tests/mozilla/interfaces.html index 2effd46f565c..c3941a5f9082 100644 --- a/tests/wpt/mozilla/tests/mozilla/interfaces.html +++ b/tests/wpt/mozilla/tests/mozilla/interfaces.html @@ -165,6 +165,8 @@ "MediaList", "MediaQueryList", "MediaQueryListEvent", + "MediaStream", + "MediaStreamTrack", "MessageEvent", "MimeType", "MimeTypeArray", @@ -201,6 +203,11 @@ "Range", "Request", "Response", + "RTCIceCandidate", + "RTCPeerConnection", + "RTCPeerConnectionIceEvent", + "RTCSessionDescription", + "RTCTrackEvent", "Screen", "ShadowRoot", "StereoPannerNode", From 76d72f8399d67b36dabc2a66bab21946514d70b7 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Mon, 13 May 2019 14:31:01 -0700 Subject: [PATCH 07/13] Add expectations for (disabled) mediacapture tests --- .../GUM-deny.https.html.ini | 4 + .../GUM-impossible-constraint.https.html.ini | 2 + .../GUM-invalid-facing-mode.https.html.ini | 4 + ...M-non-applicable-constraint.https.html.ini | 2 + .../MediaDevices-SecureContext.html.ini | 4 + ...diaDevices-enumerateDevices.https.html.ini | 7 ++ ...ces-getSupportedConstraints.https.html.ini | 52 +++++++++ .../MediaDevices-getUserMedia.https.html.ini | 20 ++++ ...ream-MediaElement-srcObject.https.html.ini | 5 + .../MediaStream-clone.https.html.ini | 7 ++ ...ream-default-feature-policy.https.html.ini | 37 ++++++ .../MediaStream-finished-add.https.html.ini | 4 + .../MediaStream-idl.https.html.ini | 4 + .../MediaStream-removetrack.https.html.ini | 8 ++ ...tream-supported-by-feature-policy.html.ini | 7 ++ ...t-disabled-audio-is-silence.https.html.ini | 4 + ...ent-disabled-video-is-black.https.html.ini | 4 + ...treamTrack-applyConstraints.https.html.ini | 17 +++ ...StreamTrack-getCapabilities.https.html.ini | 110 ++++++++++++++++++ ...ediaStreamTrack-getSettings.https.html.ini | 65 +++++++++++ .../MediaStreamTrack-init.https.html.ini | 4 + ...treamTrackEvent-constructor.https.html.ini | 10 ++ .../metadata/mediacapture-streams/__dir__.ini | 1 + .../idlharness.https.window.js.ini | 4 + 24 files changed, 386 insertions(+) create mode 100644 tests/wpt/metadata/mediacapture-streams/GUM-deny.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/GUM-impossible-constraint.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/GUM-invalid-facing-mode.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/GUM-non-applicable-constraint.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaDevices-SecureContext.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaDevices-enumerateDevices.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaDevices-getSupportedConstraints.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaDevices-getUserMedia.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-MediaElement-srcObject.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-clone.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-default-feature-policy.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-finished-add.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-idl.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-removetrack.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStream-supported-by-feature-policy.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-audio-is-silence.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-video-is-black.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-applyConstraints.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getCapabilities.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getSettings.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-init.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/MediaStreamTrackEvent-constructor.https.html.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/__dir__.ini create mode 100644 tests/wpt/metadata/mediacapture-streams/idlharness.https.window.js.ini diff --git a/tests/wpt/metadata/mediacapture-streams/GUM-deny.https.html.ini b/tests/wpt/metadata/mediacapture-streams/GUM-deny.https.html.ini new file mode 100644 index 000000000000..40cc862bcacf --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/GUM-deny.https.html.ini @@ -0,0 +1,4 @@ +[GUM-deny.https.html] + [Tests that the error callback is triggered when permission is denied] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/GUM-impossible-constraint.https.html.ini b/tests/wpt/metadata/mediacapture-streams/GUM-impossible-constraint.https.html.ini new file mode 100644 index 000000000000..76b4a82a7d33 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/GUM-impossible-constraint.https.html.ini @@ -0,0 +1,2 @@ +[GUM-impossible-constraint.https.html] + expected: CRASH diff --git a/tests/wpt/metadata/mediacapture-streams/GUM-invalid-facing-mode.https.html.ini b/tests/wpt/metadata/mediacapture-streams/GUM-invalid-facing-mode.https.html.ini new file mode 100644 index 000000000000..578ae295db7f --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/GUM-invalid-facing-mode.https.html.ini @@ -0,0 +1,4 @@ +[GUM-invalid-facing-mode.https.html] + [Tests that setting an invalid facingMode constraint in getUserMedia fails] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/GUM-non-applicable-constraint.https.html.ini b/tests/wpt/metadata/mediacapture-streams/GUM-non-applicable-constraint.https.html.ini new file mode 100644 index 000000000000..5e9ced8d8821 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/GUM-non-applicable-constraint.https.html.ini @@ -0,0 +1,2 @@ +[GUM-non-applicable-constraint.https.html] + expected: CRASH diff --git a/tests/wpt/metadata/mediacapture-streams/MediaDevices-SecureContext.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaDevices-SecureContext.html.ini new file mode 100644 index 000000000000..75a1f24462cc --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaDevices-SecureContext.html.ini @@ -0,0 +1,4 @@ +[MediaDevices-SecureContext.html] + [MediaDevices and SecureContext] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaDevices-enumerateDevices.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaDevices-enumerateDevices.https.html.ini new file mode 100644 index 000000000000..f5217123dae3 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaDevices-enumerateDevices.https.html.ini @@ -0,0 +1,7 @@ +[MediaDevices-enumerateDevices.https.html] + [mediaDevices.enumerateDevices() is present and working] + expected: FAIL + + [InputDeviceInfo is supported] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaDevices-getSupportedConstraints.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaDevices-getSupportedConstraints.https.html.ini new file mode 100644 index 000000000000..4da71c4a1eb9 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaDevices-getSupportedConstraints.https.html.ini @@ -0,0 +1,52 @@ +[MediaDevices-getSupportedConstraints.https.html] + [echoCancellation is supported] + expected: FAIL + + [sampleSize is supported] + expected: FAIL + + [autoGainControl is supported] + expected: FAIL + + [aspectRatio is supported] + expected: FAIL + + [frameRate is supported] + expected: FAIL + + [latency is supported] + expected: FAIL + + [width is supported] + expected: FAIL + + [noiseSuppression is supported] + expected: FAIL + + [navigator.mediaDevices.getSupportedConstraints exists] + expected: FAIL + + [deviceId is supported] + expected: FAIL + + [channelCount is supported] + expected: FAIL + + [resizeMode is supported] + expected: FAIL + + [groupId is supported] + expected: FAIL + + [height is supported] + expected: FAIL + + [facingMode is supported] + expected: FAIL + + [sampleRate is supported] + expected: FAIL + + [volume is supported] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaDevices-getUserMedia.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaDevices-getUserMedia.https.html.ini new file mode 100644 index 000000000000..8c3620f1c787 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaDevices-getUserMedia.https.html.ini @@ -0,0 +1,20 @@ +[MediaDevices-getUserMedia.https.html] + expected: ERROR + [getUserMedia() fails with exact invalid resizeMode.] + expected: NOTRUN + + [getUserMedia() supports setting none as resizeMode.] + expected: FAIL + + [mediaDevices.getUserMedia() is present on navigator] + expected: FAIL + + [groupId is correctly supported by getUserMedia() for video devices] + expected: FAIL + + [groupId is correctly supported by getUserMedia() for audio devices] + expected: FAIL + + [getUserMedia() supports setting crop-and-scale as resizeMode.] + expected: NOTRUN + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-MediaElement-srcObject.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-MediaElement-srcObject.https.html.ini new file mode 100644 index 000000000000..5d3ebc18f352 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-MediaElement-srcObject.https.html.ini @@ -0,0 +1,5 @@ +[MediaStream-MediaElement-srcObject.https.html] + expected: ERROR + [Tests that a MediaStream can be assigned to a video element with srcObject] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-clone.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-clone.https.html.ini new file mode 100644 index 000000000000..f49c8f01ce4c --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-clone.https.html.ini @@ -0,0 +1,7 @@ +[MediaStream-clone.https.html] + [Tests that cloning MediaStreamTrack objects works as expected] + expected: FAIL + + [Tests that cloning MediaStream objects works as expected] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-default-feature-policy.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-default-feature-policy.https.html.ini new file mode 100644 index 000000000000..342d7b8376fb --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-default-feature-policy.https.html.ini @@ -0,0 +1,37 @@ +[MediaStream-default-feature-policy.https.html] + [Default "microphone" feature policy ["self"\] disallows cross-origin iframes.] + expected: FAIL + + [Default "camera;microphone" feature policy ["self"\] allows same-origin iframes.] + expected: FAIL + + [Feature policy "microphone" can be enabled in cross-origin iframes using "allow" attribute.] + expected: FAIL + + [Feature policy "camera;microphone" can be enabled in cross-origin iframes using "allow" attribute.] + expected: FAIL + + [Default "camera" feature policy ["self"\] allows same-origin iframes.] + expected: FAIL + + [Default "camera" feature policy ["self"\] disallows cross-origin iframes.] + expected: FAIL + + [Default "microphone" feature policy ["self"\] allows same-origin iframes.] + expected: FAIL + + [Default "camera;microphone" feature policy ["self"\] allows the top-level document.] + expected: FAIL + + [Feature policy "camera" can be enabled in cross-origin iframes using "allow" attribute.] + expected: FAIL + + [Default "camera" feature policy ["self"\] allows the top-level document.] + expected: FAIL + + [Default "camera;microphone" feature policy ["self"\] disallows cross-origin iframes.] + expected: FAIL + + [Default "microphone" feature policy ["self"\] allows the top-level document.] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-finished-add.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-finished-add.https.html.ini new file mode 100644 index 000000000000..914569f57c0b --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-finished-add.https.html.ini @@ -0,0 +1,4 @@ +[MediaStream-finished-add.https.html] + [Tests that adding a track to an inactive MediaStream is allowed] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-idl.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-idl.https.html.ini new file mode 100644 index 000000000000..24a7872d38f2 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-idl.https.html.ini @@ -0,0 +1,4 @@ +[MediaStream-idl.https.html] + [Tests that a MediaStream constructor follows the algorithm set in the spec] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-removetrack.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-removetrack.https.html.ini new file mode 100644 index 000000000000..18243eb923c6 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-removetrack.https.html.ini @@ -0,0 +1,8 @@ +[MediaStream-removetrack.https.html] + expected: ERROR + [Test that removal from a MediaStream fires ended on media elements (audio first)] + expected: NOTRUN + + [Test that removal from a MediaStream fires ended on media elements (video first)] + expected: NOTRUN + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStream-supported-by-feature-policy.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStream-supported-by-feature-policy.html.ini new file mode 100644 index 000000000000..ce475a2b1413 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStream-supported-by-feature-policy.html.ini @@ -0,0 +1,7 @@ +[MediaStream-supported-by-feature-policy.html] + [document.featurePolicy.features should advertise microphone.] + expected: FAIL + + [document.featurePolicy.features should advertise camera.] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-audio-is-silence.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-audio-is-silence.https.html.ini new file mode 100644 index 000000000000..35a491747d24 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-audio-is-silence.https.html.ini @@ -0,0 +1,4 @@ +[MediaStreamTrack-MediaElement-disabled-audio-is-silence.https.html] + [Tests that a disabled audio track in a MediaStream is rendered as silence] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-video-is-black.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-video-is-black.https.html.ini new file mode 100644 index 000000000000..9739742128c5 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-MediaElement-disabled-video-is-black.https.html.ini @@ -0,0 +1,4 @@ +[MediaStreamTrack-MediaElement-disabled-video-is-black.https.html] + [Tests that a disabled video track in a MediaStream is rendered as blackness] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-applyConstraints.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-applyConstraints.https.html.ini new file mode 100644 index 000000000000..dd20b416a4c6 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-applyConstraints.https.html.ini @@ -0,0 +1,17 @@ +[MediaStreamTrack-applyConstraints.https.html] + expected: ERROR + [applyConstraints rejects attempt to switch device using groupId] + expected: FAIL + + [applyConstraints rejects invalid groupID] + expected: FAIL + + [applyConstraints rejects invalid resizeMode] + expected: FAIL + + [applyConstraints accepts invalid ideal resizeMode, does not change setting] + expected: NOTRUN + + [applyConstraints accepts invalid ideal groupID, does not change setting] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getCapabilities.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getCapabilities.https.html.ini new file mode 100644 index 000000000000..4c2916c913f2 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getCapabilities.https.html.ini @@ -0,0 +1,110 @@ +[MediaStreamTrack-getCapabilities.https.html] + expected: ERROR + [Setup video MediaStreamTrack getCapabilities() test for frameRate] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for facingMode] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for sampleSize] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for height] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for latency] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for resizeMode] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for frameRate] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for groupId] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for sampleSize] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for volume] + expected: FAIL + + [Setup video MediaStreamTrack getCapabilities() test for groupId] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for height] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for sampleRate] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for channelCount] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for aspectRatio] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for echoCancellation] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for autoGainControl] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for facingMode] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for noiseSuppression] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for deviceId] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for groupId] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for resizeMode] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for volume] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for echoCancellation] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for noiseSuppression] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for deviceId] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for width] + expected: NOTRUN + + [Setup video MediaStreamTrack getCapabilities() test for deviceId] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for sampleRate] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for channelCount] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for latency] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for groupId] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for width] + expected: NOTRUN + + [Setup audio InputDeviceInfo getCapabilities() test for deviceId] + expected: NOTRUN + + [Setup audio MediaStreamTrack getCapabilities() test for autoGainControl] + expected: NOTRUN + + [Setup video InputDeviceInfo getCapabilities() test for aspectRatio] + expected: NOTRUN + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getSettings.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getSettings.https.html.ini new file mode 100644 index 000000000000..ffab24b4e950 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-getSettings.https.html.ini @@ -0,0 +1,65 @@ +[MediaStreamTrack-getSettings.https.html] + expected: ERROR + [A device can be opened twice with different resolutions] + expected: FAIL + + [channelCount is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [autoGainControl is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [sampleSize is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [frameRate is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [sampleRate is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [volume is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [width is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [A device can be opened twice and have the same device ID] + expected: FAIL + + [resizeMode is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [groupId is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [facingMode is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [aspectRatio is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [deviceId is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [latency is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [height is reported by getSettings() for getUserMedia() video tracks] + expected: NOTRUN + + [groupId is correctly reported by getSettings() for all devices] + expected: FAIL + + [echoCancellation is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [noiseSuppression is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [groupId is reported by getSettings() for getUserMedia() audio tracks] + expected: NOTRUN + + [deviceId is reported by getSettings() for getUserMedia() audio tracks] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-init.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-init.https.html.ini new file mode 100644 index 000000000000..4b19f4724244 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrack-init.https.html.ini @@ -0,0 +1,4 @@ +[MediaStreamTrack-init.https.html] + [getUserMedia({video:true}) creates a stream with a properly initialized video track] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/MediaStreamTrackEvent-constructor.https.html.ini b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrackEvent-constructor.https.html.ini new file mode 100644 index 000000000000..543a18c43f27 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/MediaStreamTrackEvent-constructor.https.html.ini @@ -0,0 +1,10 @@ +[MediaStreamTrackEvent-constructor.https.html] + [The MediaStreamTrackEvent instance's track attribute is set.] + expected: FAIL + + [The eventInitDict's track member is required.] + expected: FAIL + + [The eventInitDict argument is required] + expected: FAIL + diff --git a/tests/wpt/metadata/mediacapture-streams/__dir__.ini b/tests/wpt/metadata/mediacapture-streams/__dir__.ini new file mode 100644 index 000000000000..6993fb302400 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/__dir__.ini @@ -0,0 +1 @@ +prefs: ["dom.mediadevices.enabled:true"] diff --git a/tests/wpt/metadata/mediacapture-streams/idlharness.https.window.js.ini b/tests/wpt/metadata/mediacapture-streams/idlharness.https.window.js.ini new file mode 100644 index 000000000000..4fd5d804f1cf --- /dev/null +++ b/tests/wpt/metadata/mediacapture-streams/idlharness.https.window.js.ini @@ -0,0 +1,4 @@ +[idlharness.https.window.html] + [mediacapture-streams interfaces.] + expected: FAIL + From 4d29aa581a4a18db442d6c9cde03ee0d0ab4f79b Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Wed, 15 May 2019 11:42:04 -0700 Subject: [PATCH 08/13] Disable mediadevices for webrtc --- .../metadata/mediacapture-record/__dir__.ini | 1 + ...ction-add-track-no-deadlock.https.html.ini | 1 - .../RTCPeerConnection-addTrack.https.html.ini | 17 ++-- ...erConnection-addTransceiver.https.html.ini | 3 +- ...rConnection-connectionState.https.html.ini | 3 +- .../RTCPeerConnection-getStats.https.html.ini | 15 ++-- ...onnectionState-disconnected.https.html.ini | 1 - ...eerConnection-onnegotiationneeded.html.ini | 5 +- ...ion-onsignalingstatechanged.https.html.ini | 1 - ...onnection-remote-track-mute.https.html.ini | 7 +- ...CPeerConnection-removeTrack.https.html.ini | 25 +++--- ...setRemoteDescription-tracks.https.html.ini | 27 +++--- ...CPeerConnection-track-stats.https.html.ini | 11 ++- ...PeerConnection-transceivers.https.html.ini | 87 +++++++++---------- ...iver-getContributingSources.https.html.ini | 3 +- .../RTCRtpReceiver-getStats.https.html.ini | 3 +- ...r-getSynchronizationSources.https.html.ini | 23 +++-- .../RTCRtpSender-getStats.https.html.ini | 3 +- .../RTCRtpSender-replaceTrack.https.html.ini | 17 ++-- .../RTCRtpSender-transport.https.html.ini | 9 +- .../webrtc/RTCRtpTransceiver.https.html.ini | 69 ++++++++------- tests/wpt/metadata/webrtc/__dir__.ini | 2 +- .../webrtc/idlharness.https.window.js.ini | 3 + ...with-OfferToReceive-options.https.html.ini | 7 +- .../webrtc/legacy/onaddstream.https.html.ini | 1 - 25 files changed, 162 insertions(+), 182 deletions(-) create mode 100644 tests/wpt/metadata/mediacapture-record/__dir__.ini diff --git a/tests/wpt/metadata/mediacapture-record/__dir__.ini b/tests/wpt/metadata/mediacapture-record/__dir__.ini new file mode 100644 index 000000000000..6993fb302400 --- /dev/null +++ b/tests/wpt/metadata/mediacapture-record/__dir__.ini @@ -0,0 +1 @@ +prefs: ["dom.mediadevices.enabled:true"] diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini index 033edaad84f7..3e9d2b4aaca3 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-add-track-no-deadlock.https.html] - expected: ERROR [RTCPeerConnection addTrack does not deadlock.] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini index 6de0040367c7..e3ad91184bb7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini @@ -1,29 +1,28 @@ [RTCPeerConnection-addTrack.https.html] - expected: ERROR [addTrack with existing sender that has been used to send should create new sender] - expected: NOTRUN + expected: FAIL [addTrack with single track argument and multiple streams should succeed] - expected: NOTRUN + expected: FAIL [addTrack with existing sender with null track, different kind, and recvonly direction should create new sender] - expected: NOTRUN + expected: FAIL [addTrack with single track argument and no stream should succeed] - expected: NOTRUN + expected: FAIL [addTrack with existing sender that has not been used to send should reuse the sender] - expected: NOTRUN + expected: FAIL [addTrack with single track argument and single stream should succeed] - expected: NOTRUN + expected: FAIL [addTrack when pc is closed should throw InvalidStateError] expected: FAIL [addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender] - expected: NOTRUN + expected: FAIL [Adding the same track multiple times should throw InvalidAccessError] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini index 46efcaa35623..1b5424929ef5 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-addTransceiver.https.html] - expected: ERROR [addTransceiver() with direction inactive should have result transceiver.direction be the same] expected: FAIL @@ -34,5 +33,5 @@ expected: FAIL [addTransceiver(track) multiple times should create multiple transceivers] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini index 73eb59444599..1b387aeb1f73 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-connectionState.https.html] - expected: ERROR [Initial connectionState should be new] expected: FAIL @@ -7,7 +6,7 @@ expected: FAIL [connectionState transitions to connected via connecting] - expected: NOTRUN + expected: FAIL [connection with one data channel should eventually have transports in connected state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini index b7adadda25ec..0b0f45d1843e 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -1,22 +1,21 @@ [RTCPeerConnection-getStats.https.html] - expected: ERROR [getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats] expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats and outbound-track-stats] - expected: NOTRUN + expected: FAIL [getStats() with track associated with both sender and receiver should reject with InvalidAccessError] - expected: NOTRUN + expected: FAIL [getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats] - expected: NOTRUN + expected: FAIL [getStats() with no argument should return stats for no-stream tracks] - expected: NOTRUN + expected: FAIL [getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats] - expected: NOTRUN + expected: FAIL [getStats() with no argument should succeed] expected: FAIL @@ -28,13 +27,13 @@ expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats on an empty PC] - expected: NOTRUN + expected: FAIL [getStats() with track added via addTransceiver should succeed] expected: FAIL [getStats() with track associated with more than one sender should reject with InvalidAccessError] - expected: NOTRUN + expected: FAIL [getStats() with track added via addTrack should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini index b135b4af16f2..5914c3ea93e2 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-iceConnectionState-disconnected.https.html] - expected: ERROR [ICE goes to disconnected if the other side goes away] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index ac85dc6b8990..2bd50710962a 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -16,7 +16,7 @@ expected: FAIL [Updating the direction of the transceiver should cause negotiationneeded to fire] - expected: NOTRUN + expected: FAIL [negotiationneeded event should fire only after signaling state go back to stable after setLocalDescription] expected: FAIL @@ -33,6 +33,3 @@ [calling createDataChannel twice should fire negotiationneeded event once] expected: FAIL - [negotiationneeded event should not fire if signaling state is not stable] - expected: TIMEOUT - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini index a81b9e2ce821..d564904adfb2 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-onsignalingstatechanged.https.html] - expected: ERROR [RTCPeerConnection onsignalingstatechanged] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini index 36b8c351cb59..b8cc12e7e538 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -1,13 +1,12 @@ [RTCPeerConnection-remote-track-mute.https.html] - expected: ERROR [Changing transceiver direction to 'sendrecv' unmutes the remote track] - expected: NOTRUN + expected: FAIL [pc.close() mutes remote tracks] - expected: NOTRUN + expected: FAIL [Changing transceiver direction to 'inactive' mutes the remote track] - expected: NOTRUN + expected: FAIL [ontrack: track goes from muted to unmuted] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini index 6875bfdb7e12..03f54a272326 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini @@ -1,41 +1,40 @@ [RTCPeerConnection-removeTrack.https.html] - expected: ERROR [Calling removeTrack with currentDirection sendonly should set direction to inactive] - expected: NOTRUN + expected: FAIL [addTrack - Calling removeTrack when connection is closed should throw InvalidStateError] - expected: NOTRUN + expected: FAIL [addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError] - expected: NOTRUN + expected: FAIL [addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: NOTRUN + expected: FAIL [addTransceiver - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: NOTRUN + expected: FAIL [addTransceiver - Calling removeTrack with valid sender should set sender.track to null] - expected: NOTRUN + expected: FAIL [Calling removeTrack on a stopped transceiver should be a no-op] - expected: NOTRUN + expected: FAIL [addTransceiver - Calling removeTrack when connection is closed should throw InvalidStateError] expected: FAIL [addTrack - Calling removeTrack on different connection should throw InvalidAccessError] - expected: NOTRUN + expected: FAIL [Calling removeTrack with currentDirection inactive should not change direction] - expected: NOTRUN + expected: FAIL [Calling removeTrack with currentDirection sendrecv should set direction to recvonly] - expected: NOTRUN + expected: FAIL [Calling removeTrack with currentDirection recvonly should not change direction] - expected: NOTRUN + expected: FAIL [addTrack - Calling removeTrack with valid sender should set sender.track to null] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini index 0f67a2850ead..a8cb104dff57 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -1,44 +1,43 @@ [RTCPeerConnection-setRemoteDescription-tracks.https.html] - expected: ERROR [ontrack fires before setRemoteDescription resolves.] - expected: NOTRUN + expected: FAIL [addTrack() for an existing stream makes stream.onaddtrack fire.] - expected: NOTRUN + expected: FAIL [ontrack's receiver matches getReceivers().] - expected: NOTRUN + expected: FAIL [track.onmute fires before setRemoteDescription resolves.] - expected: NOTRUN + expected: FAIL [addTrack() with two tracks and one stream makes ontrack fire twice with the tracks and shared stream.] - expected: NOTRUN + expected: FAIL [stream.onaddtrack fires before setRemoteDescription resolves.] - expected: NOTRUN + expected: FAIL [addTrack() with a track and no stream makes ontrack fire with a track and no stream.] expected: FAIL [addTrack() with a track and a stream makes ontrack fire with a track and a stream.] - expected: NOTRUN + expected: FAIL [removeTrack() makes track.onmute fire and the track to be muted.] - expected: NOTRUN + expected: FAIL [addTrack() with a track and two streams makes ontrack fire with a track and two streams.] - expected: NOTRUN + expected: FAIL [stream.onremovetrack fires before setRemoteDescription resolves.] - expected: NOTRUN + expected: FAIL [removeTrack() makes stream.onremovetrack fire and the track to be removed from the stream.] - expected: NOTRUN + expected: FAIL [removeTrack() does not remove the receiver.] - expected: NOTRUN + expected: FAIL [removeTrack() twice is safe.] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini index 404bb86a9486..cfca4a6ac221 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini @@ -1,5 +1,4 @@ [RTCPeerConnection-track-stats.https.html] - expected: ERROR [replaceTrack() before offer: new track attachment stats present] expected: FAIL @@ -13,7 +12,7 @@ expected: FAIL [RTCRtpReceiver.getStats() contains only inbound-rtp and related stats] - expected: NOTRUN + expected: FAIL [O/A exchange yields inbound RTP stream stats for receiving track] expected: FAIL @@ -34,19 +33,19 @@ expected: FAIL [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are zero senders or receivers for the track] - expected: NOTRUN + expected: FAIL [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are multiple senders for the track] - expected: NOTRUN + expected: FAIL [replaceTrack() after answer: new track attachment stats present] expected: FAIL [RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats()] - expected: NOTRUN + expected: FAIL [RTCPeerConnection.getStats(sendingTrack) is the same as RTCRtpSender.getStats()] - expected: NOTRUN + expected: FAIL [addTrack() with setLocalDescription() yields track stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini index ffce6fe8e7ad..f630372bc8f0 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -1,134 +1,133 @@ [RTCPeerConnection-transceivers.https.html] - expected: ERROR [setLocalDescription(answer): transceiver.currentDirection is recvonly] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): ontrack fires with a track] - expected: NOTRUN + expected: FAIL [Can setup two-way call using a single transceiver] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): transceiver.mid is the same on both ends] - expected: NOTRUN + expected: FAIL [addTransceiver(track, init): initialize sendEncodings[0\].active to false] - expected: NOTRUN + expected: FAIL [addTransceiver('video'): transceiver.receiver.track.kind == 'video'] - expected: NOTRUN + expected: FAIL [addTransceiver('audio'): transceiver.stopped is false] - expected: NOTRUN + expected: FAIL [setLocalDescription(answer): transceiver.currentDirection is sendonly] - expected: NOTRUN + expected: FAIL [addTrack: transceiver is not associated with an m-section] - expected: NOTRUN + expected: FAIL [transceiver.sender.track does not revert to an old state] - expected: NOTRUN + expected: FAIL [addTrack: transceiver.receiver has its own track] - expected: NOTRUN + expected: FAIL [addTrack: "transceiver == {sender,receiver}"] - expected: NOTRUN + expected: FAIL [setLocalDescription(offer): transceiver.mid matches the offer SDP] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): transceiver.direction is recvonly] - expected: NOTRUN + expected: FAIL [Changing transceiver direction to 'sendrecv' makes ontrack fire] - expected: NOTRUN + expected: FAIL [addTrack: transceiver is not stopped] - expected: NOTRUN + expected: FAIL [addTrack(1 stream): ontrack fires with corresponding stream] - expected: NOTRUN + expected: FAIL [addTransceiver(track, init): initialize direction to inactive] - expected: NOTRUN + expected: FAIL [addTrack(0 streams): ontrack fires with no stream] - expected: NOTRUN + expected: FAIL [addTrack: transceiver's direction is sendrecv] - expected: NOTRUN + expected: FAIL [addTrack(2 streams): ontrack fires with corresponding two streams] - expected: NOTRUN + expected: FAIL [addTransceiver(0 streams): ontrack fires with no stream] - expected: NOTRUN + expected: FAIL [addTransceiver(1 stream): ontrack fires with corresponding stream] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): transceiver.currentDirection is null] - expected: NOTRUN + expected: FAIL [addTrack: transceiver.sender is associated with the track] - expected: NOTRUN + expected: FAIL [addTransceiver does not reuse reusable transceivers] - expected: NOTRUN + expected: FAIL [addTransceiver(track): "transceiver == {sender,receiver}"] - expected: NOTRUN + expected: FAIL [transceiver.direction does not revert to an old state] - expected: NOTRUN + expected: FAIL [addTransceiver('audio'): transceiver.sender.track == null] - expected: NOTRUN + expected: FAIL [addTransceiver('audio'): creates a transceiver with direction sendrecv] - expected: NOTRUN + expected: FAIL [setLocalDescription(offer): transceiver gets associated with an m-section] - expected: NOTRUN + expected: FAIL [addTransceiver('audio'): transceiver.currentDirection is null] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): ontrack's stream.id is the same as stream.id] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): transceiver.stopped is false] - expected: NOTRUN + expected: FAIL [addTransceiver('audio'): transceiver.receiver.track.kind == 'audio'] - expected: NOTRUN + expected: FAIL [addTransceiver(track): creates a transceiver for the track] - expected: NOTRUN + expected: FAIL [addTransceiver(2 streams): ontrack fires with corresponding two streams] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): "transceiver == {sender,receiver}"] - expected: NOTRUN + expected: FAIL [addTrack: transceiver's currentDirection is null] - expected: NOTRUN + expected: FAIL [addTrack reuses reusable transceivers] - expected: NOTRUN + expected: FAIL [addTrack: creates a transceiver for the sender] expected: FAIL [Closing the PC stops the transceivers] - expected: NOTRUN + expected: FAIL [setRemoteDescription(offer): ontrack fires with a transceiver.] - expected: NOTRUN + expected: FAIL [addTrack: transceiver.receiver's track is muted] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini index f8ec71028533..ff336963390c 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini @@ -1,8 +1,7 @@ [RTCRtpReceiver-getContributingSources.https.html] - expected: ERROR [[audio\] getContributingSources() returns an empty list in loopback call] expected: FAIL [[video\] getContributingSources() returns an empty list in loopback call] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini index 0d9402476431..a69a3eda57bd 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -1,7 +1,6 @@ [RTCRtpReceiver-getStats.https.html] - expected: ERROR [receiver.getStats() via addTrack should return stats report containing inbound-rtp stats] - expected: NOTRUN + expected: FAIL [receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini index 18a8e0fef6a1..a5a15242ac4c 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -1,38 +1,37 @@ [RTCRtpReceiver-getSynchronizationSources.https.html] - expected: ERROR [[audio\] RTCRtpSynchronizationSource.source is a number] - expected: NOTRUN + expected: FAIL [[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: NOTRUN + expected: FAIL [[audio\] getSynchronizationSources() eventually returns a non-empty list] expected: FAIL [[video\] RTCRtpSynchronizationSource.timestamp is a number] - expected: NOTRUN + expected: FAIL [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean] - expected: NOTRUN + expected: FAIL [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: NOTRUN + expected: FAIL [[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: NOTRUN + expected: FAIL [[video\] getSynchronizationSources() eventually returns a non-empty list] - expected: NOTRUN + expected: FAIL [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: NOTRUN + expected: FAIL [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]] - expected: NOTRUN + expected: FAIL [[video\] RTCRtpSynchronizationSource.source is a number] - expected: NOTRUN + expected: FAIL [[audio\] RTCRtpSynchronizationSource.timestamp is a number] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini index 924a89b73025..4256534ee3e2 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini @@ -1,7 +1,6 @@ [RTCRtpSender-getStats.https.html] - expected: ERROR [sender.getStats() via addTrack should return stats report containing outbound-rtp stats] - expected: NOTRUN + expected: FAIL [sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini index 04f02b467342..821925718a1d 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini @@ -1,29 +1,28 @@ [RTCRtpSender-replaceTrack.https.html] - expected: ERROR [Calling replaceTrack on sender with null track and not set to session description should resolve with sender.track set to given track] - expected: NOTRUN + expected: FAIL [Calling replaceTrack(null) on sender not set to session description should resolve with sender.track set to null] - expected: NOTRUN + expected: FAIL [Calling replaceTrack with track of different kind should reject with TypeError] - expected: NOTRUN + expected: FAIL [Calling replaceTrack on sender with similar track and and set to session description should resolve with sender.track set to new track] - expected: NOTRUN + expected: FAIL [Calling replaceTrack on closed connection should reject with InvalidStateError] expected: FAIL [Calling replaceTrack on sender with stopped track and and set to session description should resolve with sender.track set to given track] - expected: NOTRUN + expected: FAIL [Calling replaceTrack on stopped sender should reject with InvalidStateError] - expected: NOTRUN + expected: FAIL [Calling replaceTrack on sender not set to session description should resolve with sender.track set to given track] - expected: NOTRUN + expected: FAIL [Calling replaceTrack(null) on sender set to session description should resolve with sender.track set to null] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini index 200008065aef..fac8652e7aa7 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini @@ -1,16 +1,15 @@ [RTCRtpSender-transport.https.html] - expected: ERROR [RTCRtpSender/receiver.transport at the right time, with bundle policy balanced] - expected: NOTRUN + expected: FAIL [RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle] - expected: NOTRUN + expected: FAIL [RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat] - expected: NOTRUN + expected: FAIL [RTCRtpSender/receiver.transport has a value when connected] - expected: NOTRUN + expected: FAIL [RTCRtpSender.transport is null when unconnected] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini index 2958e0a5949f..c65b2f802864 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -1,110 +1,109 @@ [RTCRtpTransceiver.https.html] - expected: ERROR [checkAddTransceiverWithTrack] expected: FAIL [checkNoMidAnswer] - expected: NOTRUN + expected: FAIL [checkSetDirection] - expected: NOTRUN + expected: FAIL [checkCurrentDirection] - expected: NOTRUN + expected: FAIL [checkReplaceTrackNullDoesntPreventPairing] - expected: NOTRUN + expected: FAIL [checkAddTrackExistingTransceiverThenRemove] - expected: NOTRUN + expected: FAIL [checkMsectionReuse] - expected: NOTRUN + expected: FAIL [checkRemoteRollback] - expected: NOTRUN + expected: FAIL [checkAddTransceiverNoTrackDoesntPair] - expected: NOTRUN + expected: FAIL [checkRemoveTrackNegotiation] - expected: NOTRUN + expected: FAIL [checkStopAfterClose] - expected: NOTRUN + expected: FAIL [checkNoMidOffer] - expected: NOTRUN + expected: FAIL [checkStopAfterCreateOffer] - expected: NOTRUN + expected: FAIL [checkAddTransceiverThenAddTrackPairs] - expected: NOTRUN + expected: FAIL [checkLocalRollback] - expected: NOTRUN + expected: FAIL [checkSendrecvWithTracklessStream] - expected: NOTRUN + expected: FAIL [checkMsidNoTrackId] - expected: NOTRUN + expected: FAIL [checkAddTransceiverNoTrack] expected: FAIL [checkAddTransceiverBadKind] - expected: NOTRUN + expected: FAIL [checkStopAfterCreateAnswer] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithDirection] - expected: NOTRUN + expected: FAIL [checkAddTrackPairs] - expected: NOTRUN + expected: FAIL [checkStopAfterSetLocalAnswer] - expected: NOTRUN + expected: FAIL [checkStopAfterSetRemoteOffer] - expected: NOTRUN + expected: FAIL [checkStopAfterCreateOfferWithReusedMsection] - expected: NOTRUN + expected: FAIL [checkMute] - expected: NOTRUN + expected: FAIL [checkSendrecvWithNoSendTrack] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithSetRemoteOfferSending] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithAddTrack] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithSetRemoteOfferNoSend] - expected: NOTRUN + expected: FAIL [checkRemoveAndReadd] - expected: NOTRUN + expected: FAIL [checkStopAfterSetLocalOffer] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithTrackDoesntPair] - expected: NOTRUN + expected: FAIL [checkAddTransceiverThenReplaceTrackDoesntPair] - expected: NOTRUN + expected: FAIL [checkRollbackAndSetRemoteOfferWithDifferentType] - expected: NOTRUN + expected: FAIL [checkStop] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/__dir__.ini b/tests/wpt/metadata/webrtc/__dir__.ini index 99550e100683..7c0f6049724b 100644 --- a/tests/wpt/metadata/webrtc/__dir__.ini +++ b/tests/wpt/metadata/webrtc/__dir__.ini @@ -1 +1 @@ -prefs: ["dom.webrtc.enabled:true", "dom.mediadevices.enabled:true"] +prefs: ["dom.webrtc.enabled:true", "dom.mediadevices.enabled:false"] diff --git a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini index f06ce24b7e81..51a1f678cd96 100644 --- a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini +++ b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini @@ -1181,3 +1181,6 @@ [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "connectionState" with the proper type] expected: FAIL + [Test driver for asyncInitMediaStreamTrack] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini index b9dadb679850..841c26c5f9a3 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -1,14 +1,13 @@ [RTCRtpTransceiver-with-OfferToReceive-options.https.html] - expected: ERROR [checkAddTransceiverWithStream] expected: FAIL [checkAddTransceiverWithOfferToReceiveVideo] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithOfferToReceiveBoth] - expected: NOTRUN + expected: FAIL [checkAddTransceiverWithOfferToReceiveAudio] - expected: NOTRUN + expected: FAIL diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini index 7923ea02036a..71b4cc138f65 100644 --- a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -1,5 +1,4 @@ [onaddstream.https.html] - expected: ERROR [Check onaddstream] expected: FAIL From c7546a5b02abb2c67d8dc7f0efdfb6c39785e8fb Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Wed, 15 May 2019 14:08:31 -0700 Subject: [PATCH 09/13] Update servo-media --- Cargo.lock | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/Cargo.lock b/Cargo.lock index c33b7821577e..944fc188a6fd 100644 --- a/Cargo.lock +++ b/Cargo.lock @@ -3863,7 +3863,7 @@ dependencies = [ [[package]] name = "servo-media" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "servo-media-audio 0.1.0 (git+https://github.com/servo/media)", "servo-media-player 0.1.0 (git+https://github.com/servo/media)", @@ -3874,7 +3874,7 @@ dependencies = [ [[package]] name = "servo-media-audio" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "boxfnonce 0.1.0 (registry+https://github.com/rust-lang/crates.io-index)", "byte-slice-cast 0.2.0 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3890,7 +3890,7 @@ dependencies = [ [[package]] name = "servo-media-dummy" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "boxfnonce 0.1.0 (registry+https://github.com/rust-lang/crates.io-index)", "ipc-channel 0.11.2 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3904,7 +3904,7 @@ dependencies = [ [[package]] name = "servo-media-gstreamer" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "boxfnonce 0.1.0 (registry+https://github.com/rust-lang/crates.io-index)", "byte-slice-cast 0.2.0 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3938,7 +3938,7 @@ dependencies = [ [[package]] name = "servo-media-gstreamer-render" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "gstreamer 0.13.0 (registry+https://github.com/rust-lang/crates.io-index)", "gstreamer-video 0.13.0 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3948,7 +3948,7 @@ dependencies = [ [[package]] name = "servo-media-gstreamer-render-unix" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "glib 0.7.1 (registry+https://github.com/rust-lang/crates.io-index)", "gstreamer 0.13.0 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3961,7 +3961,7 @@ dependencies = [ [[package]] name = "servo-media-player" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "ipc-channel 0.11.2 (registry+https://github.com/rust-lang/crates.io-index)", "serde 1.0.80 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3972,7 +3972,7 @@ dependencies = [ [[package]] name = "servo-media-streams" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "lazy_static 1.2.0 (registry+https://github.com/rust-lang/crates.io-index)", "uuid 0.7.1 (registry+https://github.com/rust-lang/crates.io-index)", @@ -3981,7 +3981,7 @@ dependencies = [ [[package]] name = "servo-media-webrtc" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "boxfnonce 0.1.0 (registry+https://github.com/rust-lang/crates.io-index)", "log 0.4.6 (registry+https://github.com/rust-lang/crates.io-index)", @@ -4080,7 +4080,7 @@ dependencies = [ [[package]] name = "servo_media_derive" version = "0.1.0" -source = "git+https://github.com/servo/media#8691ca03a37c45bf7efa9430e434dd5fbf0abfa3" +source = "git+https://github.com/servo/media#705e4cb84cc0216c009f1a27fefdf61f8f3797be" dependencies = [ "proc-macro2 0.4.26 (registry+https://github.com/rust-lang/crates.io-index)", "quote 0.6.3 (registry+https://github.com/rust-lang/crates.io-index)", From 5f1020bb83b87e31375afe25f6d4e7055d02e499 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Wed, 15 May 2019 13:53:23 -0700 Subject: [PATCH 10/13] Mock mediadevices on WPT --- components/config/prefs.rs | 2 + components/script/dom/mediadevices.rs | 5 ++ resources/prefs.json | 1 + ...ction-add-track-no-deadlock.https.html.ini | 1 + .../RTCPeerConnection-addTrack.https.html.ini | 17 ++-- ...erConnection-addTransceiver.https.html.ini | 3 +- ...rConnection-connectionState.https.html.ini | 3 +- .../RTCPeerConnection-createAnswer.html.ini | 2 +- .../RTCPeerConnection-getStats.https.html.ini | 15 ++-- ...onnectionState-disconnected.https.html.ini | 1 + ...eerConnection-onnegotiationneeded.html.ini | 5 +- ...ion-onsignalingstatechanged.https.html.ini | 1 + ...onnection-remote-track-mute.https.html.ini | 7 +- ...CPeerConnection-removeTrack.https.html.ini | 25 +++--- ...setRemoteDescription-tracks.https.html.ini | 27 +++--- ...CPeerConnection-track-stats.https.html.ini | 11 +-- ...PeerConnection-transceivers.https.html.ini | 87 ++++++++++--------- ...iver-getContributingSources.https.html.ini | 3 +- .../RTCRtpReceiver-getStats.https.html.ini | 3 +- ...r-getSynchronizationSources.https.html.ini | 23 ++--- .../RTCRtpSender-getStats.https.html.ini | 3 +- .../RTCRtpSender-replaceTrack.https.html.ini | 17 ++-- .../RTCRtpSender-transport.https.html.ini | 9 +- .../webrtc/RTCRtpTransceiver.https.html.ini | 69 +++++++-------- tests/wpt/metadata/webrtc/__dir__.ini | 2 +- .../webrtc/idlharness.https.window.js.ini | 3 - ...with-OfferToReceive-options.https.html.ini | 7 +- .../webrtc/legacy/onaddstream.https.html.ini | 1 + 28 files changed, 191 insertions(+), 162 deletions(-) diff --git a/components/config/prefs.rs b/components/config/prefs.rs index 3bb24fb7e11e..4b8bee0397eb 100644 --- a/components/config/prefs.rs +++ b/components/config/prefs.rs @@ -193,6 +193,8 @@ mod gen { mediadevices: { #[serde(default)] enabled: bool, + #[serde(default)] + mock: bool, }, microdata: { testing: { diff --git a/components/script/dom/mediadevices.rs b/components/script/dom/mediadevices.rs index bcc575a2ef6d..26b6ccb7eb8c 100644 --- a/components/script/dom/mediadevices.rs +++ b/components/script/dom/mediadevices.rs @@ -17,6 +17,7 @@ use crate::dom::mediastream::MediaStream; use crate::dom::mediastreamtrack::MediaStreamTrack; use crate::dom::promise::Promise; use dom_struct::dom_struct; +use servo_config::prefs; use servo_media::streams::capture::{Constrain, ConstrainRange, MediaTrackConstraintSet}; use servo_media::streams::MediaStreamType; use servo_media::ServoMedia; @@ -35,6 +36,10 @@ impl MediaDevices { } pub fn new(global: &GlobalScope) -> DomRoot { + let mock = prefs::pref_map().get("dom.mediadevices.mock").as_bool(); + if let Some(true) = mock { + ServoMedia::get().unwrap().set_capture_mocking(true); + } reflect_dom_object( Box::new(MediaDevices::new_inherited()), global, diff --git a/resources/prefs.json b/resources/prefs.json index 56d11358d9bb..ed0a262e3900 100644 --- a/resources/prefs.json +++ b/resources/prefs.json @@ -10,6 +10,7 @@ "dom.fullscreen.test": false, "dom.gamepad.enabled": false, "dom.mediadevices.enabled": false, + "dom.mediadevices.mock": false, "dom.microdata.enabled": false, "dom.microdata.testing.enabled": false, "dom.mouseevent.which.enabled": false, diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini index 3e9d2b4aaca3..033edaad84f7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-add-track-no-deadlock.https.html] + expected: ERROR [RTCPeerConnection addTrack does not deadlock.] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini index e3ad91184bb7..6de0040367c7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTrack.https.html.ini @@ -1,28 +1,29 @@ [RTCPeerConnection-addTrack.https.html] + expected: ERROR [addTrack with existing sender that has been used to send should create new sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and multiple streams should succeed] - expected: FAIL + expected: NOTRUN [addTrack with existing sender with null track, different kind, and recvonly direction should create new sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and no stream should succeed] - expected: FAIL + expected: NOTRUN [addTrack with existing sender that has not been used to send should reuse the sender] - expected: FAIL + expected: NOTRUN [addTrack with single track argument and single stream should succeed] - expected: FAIL + expected: NOTRUN [addTrack when pc is closed should throw InvalidStateError] expected: FAIL [addTrack with existing sender with null track, same kind, and recvonly direction should reuse sender] - expected: FAIL + expected: NOTRUN [Adding the same track multiple times should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini index 1b5424929ef5..46efcaa35623 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-addTransceiver.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-addTransceiver.https.html] + expected: ERROR [addTransceiver() with direction inactive should have result transceiver.direction be the same] expected: FAIL @@ -33,5 +34,5 @@ expected: FAIL [addTransceiver(track) multiple times should create multiple transceivers] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini index 1b387aeb1f73..73eb59444599 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-connectionState.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-connectionState.https.html] + expected: ERROR [Initial connectionState should be new] expected: FAIL @@ -6,7 +7,7 @@ expected: FAIL [connectionState transitions to connected via connecting] - expected: FAIL + expected: NOTRUN [connection with one data channel should eventually have transports in connected state] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini index 333d2ec4454b..625c956f86ac 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -1,5 +1,5 @@ [RTCPeerConnection-createAnswer.html] - expected: TIMEOUT + expected: CRASH [createAnswer() when connection is closed reject with InvalidStateError] expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini index 0b0f45d1843e..b7adadda25ec 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-getStats.https.html.ini @@ -1,21 +1,22 @@ [RTCPeerConnection-getStats.https.html] + expected: ERROR [getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats] expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats and outbound-track-stats] - expected: FAIL + expected: NOTRUN [getStats() with track associated with both sender and receiver should reject with InvalidAccessError] - expected: FAIL + expected: NOTRUN [getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats] - expected: FAIL + expected: NOTRUN [getStats() with no argument should return stats for no-stream tracks] - expected: FAIL + expected: NOTRUN [getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats] - expected: FAIL + expected: NOTRUN [getStats() with no argument should succeed] expected: FAIL @@ -27,13 +28,13 @@ expected: FAIL [getStats() with no argument should return stats report containing peer-connection stats on an empty PC] - expected: FAIL + expected: NOTRUN [getStats() with track added via addTransceiver should succeed] expected: FAIL [getStats() with track associated with more than one sender should reject with InvalidAccessError] - expected: FAIL + expected: NOTRUN [getStats() with track added via addTrack should succeed] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini index 5914c3ea93e2..b135b4af16f2 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState-disconnected.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-iceConnectionState-disconnected.https.html] + expected: ERROR [ICE goes to disconnected if the other side goes away] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index 2bd50710962a..ac85dc6b8990 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -16,7 +16,7 @@ expected: FAIL [Updating the direction of the transceiver should cause negotiationneeded to fire] - expected: FAIL + expected: NOTRUN [negotiationneeded event should fire only after signaling state go back to stable after setLocalDescription] expected: FAIL @@ -33,3 +33,6 @@ [calling createDataChannel twice should fire negotiationneeded event once] expected: FAIL + [negotiationneeded event should not fire if signaling state is not stable] + expected: TIMEOUT + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini index d564904adfb2..a81b9e2ce821 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-onsignalingstatechanged.https.html] + expected: ERROR [RTCPeerConnection onsignalingstatechanged] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini index b8cc12e7e538..36b8c351cb59 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -1,12 +1,13 @@ [RTCPeerConnection-remote-track-mute.https.html] + expected: ERROR [Changing transceiver direction to 'sendrecv' unmutes the remote track] - expected: FAIL + expected: NOTRUN [pc.close() mutes remote tracks] - expected: FAIL + expected: NOTRUN [Changing transceiver direction to 'inactive' mutes the remote track] - expected: FAIL + expected: NOTRUN [ontrack: track goes from muted to unmuted] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini index 03f54a272326..6875bfdb7e12 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-removeTrack.https.html.ini @@ -1,40 +1,41 @@ [RTCPeerConnection-removeTrack.https.html] + expected: ERROR [Calling removeTrack with currentDirection sendonly should set direction to inactive] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack when connection is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack on different connection that is closed should throw InvalidStateError] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack with valid sender should set sender.track to null] - expected: FAIL + expected: NOTRUN [Calling removeTrack on a stopped transceiver should be a no-op] - expected: FAIL + expected: NOTRUN [addTransceiver - Calling removeTrack when connection is closed should throw InvalidStateError] expected: FAIL [addTrack - Calling removeTrack on different connection should throw InvalidAccessError] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection inactive should not change direction] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection sendrecv should set direction to recvonly] - expected: FAIL + expected: NOTRUN [Calling removeTrack with currentDirection recvonly should not change direction] - expected: FAIL + expected: NOTRUN [addTrack - Calling removeTrack with valid sender should set sender.track to null] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini index a8cb104dff57..0f67a2850ead 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html.ini @@ -1,43 +1,44 @@ [RTCPeerConnection-setRemoteDescription-tracks.https.html] + expected: ERROR [ontrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() for an existing stream makes stream.onaddtrack fire.] - expected: FAIL + expected: NOTRUN [ontrack's receiver matches getReceivers().] - expected: FAIL + expected: NOTRUN [track.onmute fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() with two tracks and one stream makes ontrack fire twice with the tracks and shared stream.] - expected: FAIL + expected: NOTRUN [stream.onaddtrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [addTrack() with a track and no stream makes ontrack fire with a track and no stream.] expected: FAIL [addTrack() with a track and a stream makes ontrack fire with a track and a stream.] - expected: FAIL + expected: NOTRUN [removeTrack() makes track.onmute fire and the track to be muted.] - expected: FAIL + expected: NOTRUN [addTrack() with a track and two streams makes ontrack fire with a track and two streams.] - expected: FAIL + expected: NOTRUN [stream.onremovetrack fires before setRemoteDescription resolves.] - expected: FAIL + expected: NOTRUN [removeTrack() makes stream.onremovetrack fire and the track to be removed from the stream.] - expected: FAIL + expected: NOTRUN [removeTrack() does not remove the receiver.] - expected: FAIL + expected: NOTRUN [removeTrack() twice is safe.] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini index cfca4a6ac221..404bb86a9486 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-track-stats.https.html.ini @@ -1,4 +1,5 @@ [RTCPeerConnection-track-stats.https.html] + expected: ERROR [replaceTrack() before offer: new track attachment stats present] expected: FAIL @@ -12,7 +13,7 @@ expected: FAIL [RTCRtpReceiver.getStats() contains only inbound-rtp and related stats] - expected: FAIL + expected: NOTRUN [O/A exchange yields inbound RTP stream stats for receiving track] expected: FAIL @@ -33,19 +34,19 @@ expected: FAIL [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are zero senders or receivers for the track] - expected: FAIL + expected: NOTRUN [RTCPeerConnection.getStats(track) throws InvalidAccessError when there are multiple senders for the track] - expected: FAIL + expected: NOTRUN [replaceTrack() after answer: new track attachment stats present] expected: FAIL [RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats()] - expected: FAIL + expected: NOTRUN [RTCPeerConnection.getStats(sendingTrack) is the same as RTCRtpSender.getStats()] - expected: FAIL + expected: NOTRUN [addTrack() with setLocalDescription() yields track stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini index f630372bc8f0..ffce6fe8e7ad 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-transceivers.https.html.ini @@ -1,133 +1,134 @@ [RTCPeerConnection-transceivers.https.html] + expected: ERROR [setLocalDescription(answer): transceiver.currentDirection is recvonly] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack fires with a track] - expected: FAIL + expected: NOTRUN [Can setup two-way call using a single transceiver] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.mid is the same on both ends] - expected: FAIL + expected: NOTRUN [addTransceiver(track, init): initialize sendEncodings[0\].active to false] - expected: FAIL + expected: NOTRUN [addTransceiver('video'): transceiver.receiver.track.kind == 'video'] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.stopped is false] - expected: FAIL + expected: NOTRUN [setLocalDescription(answer): transceiver.currentDirection is sendonly] - expected: FAIL + expected: NOTRUN [addTrack: transceiver is not associated with an m-section] - expected: FAIL + expected: NOTRUN [transceiver.sender.track does not revert to an old state] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.receiver has its own track] - expected: FAIL + expected: NOTRUN [addTrack: "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [setLocalDescription(offer): transceiver.mid matches the offer SDP] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.direction is recvonly] - expected: FAIL + expected: NOTRUN [Changing transceiver direction to 'sendrecv' makes ontrack fire] - expected: FAIL + expected: NOTRUN [addTrack: transceiver is not stopped] - expected: FAIL + expected: NOTRUN [addTrack(1 stream): ontrack fires with corresponding stream] - expected: FAIL + expected: NOTRUN [addTransceiver(track, init): initialize direction to inactive] - expected: FAIL + expected: NOTRUN [addTrack(0 streams): ontrack fires with no stream] - expected: FAIL + expected: NOTRUN [addTrack: transceiver's direction is sendrecv] - expected: FAIL + expected: NOTRUN [addTrack(2 streams): ontrack fires with corresponding two streams] - expected: FAIL + expected: NOTRUN [addTransceiver(0 streams): ontrack fires with no stream] - expected: FAIL + expected: NOTRUN [addTransceiver(1 stream): ontrack fires with corresponding stream] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.currentDirection is null] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.sender is associated with the track] - expected: FAIL + expected: NOTRUN [addTransceiver does not reuse reusable transceivers] - expected: FAIL + expected: NOTRUN [addTransceiver(track): "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [transceiver.direction does not revert to an old state] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.sender.track == null] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): creates a transceiver with direction sendrecv] - expected: FAIL + expected: NOTRUN [setLocalDescription(offer): transceiver gets associated with an m-section] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.currentDirection is null] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack's stream.id is the same as stream.id] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): transceiver.stopped is false] - expected: FAIL + expected: NOTRUN [addTransceiver('audio'): transceiver.receiver.track.kind == 'audio'] - expected: FAIL + expected: NOTRUN [addTransceiver(track): creates a transceiver for the track] - expected: FAIL + expected: NOTRUN [addTransceiver(2 streams): ontrack fires with corresponding two streams] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): "transceiver == {sender,receiver}"] - expected: FAIL + expected: NOTRUN [addTrack: transceiver's currentDirection is null] - expected: FAIL + expected: NOTRUN [addTrack reuses reusable transceivers] - expected: FAIL + expected: NOTRUN [addTrack: creates a transceiver for the sender] expected: FAIL [Closing the PC stops the transceivers] - expected: FAIL + expected: NOTRUN [setRemoteDescription(offer): ontrack fires with a transceiver.] - expected: FAIL + expected: NOTRUN [addTrack: transceiver.receiver's track is muted] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini index ff336963390c..f8ec71028533 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getContributingSources.https.html.ini @@ -1,7 +1,8 @@ [RTCRtpReceiver-getContributingSources.https.html] + expected: ERROR [[audio\] getContributingSources() returns an empty list in loopback call] expected: FAIL [[video\] getContributingSources() returns an empty list in loopback call] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini index a69a3eda57bd..0d9402476431 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getStats.https.html.ini @@ -1,6 +1,7 @@ [RTCRtpReceiver-getStats.https.html] + expected: ERROR [receiver.getStats() via addTrack should return stats report containing inbound-rtp stats] - expected: FAIL + expected: NOTRUN [receiver.getStats() via addTransceiver should return stats report containing inbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini index a5a15242ac4c..18a8e0fef6a1 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html.ini @@ -1,37 +1,38 @@ [RTCRtpReceiver-getSynchronizationSources.https.html] + expected: ERROR [[audio\] RTCRtpSynchronizationSource.source is a number] - expected: FAIL + expected: NOTRUN [[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: FAIL + expected: NOTRUN [[audio\] getSynchronizationSources() eventually returns a non-empty list] expected: FAIL [[video\] RTCRtpSynchronizationSource.timestamp is a number] - expected: FAIL + expected: NOTRUN [[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean] - expected: FAIL + expected: NOTRUN [[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: FAIL + expected: NOTRUN [[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds] - expected: FAIL + expected: NOTRUN [[video\] getSynchronizationSources() eventually returns a non-empty list] - expected: FAIL + expected: NOTRUN [[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()] - expected: FAIL + expected: NOTRUN [[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]] - expected: FAIL + expected: NOTRUN [[video\] RTCRtpSynchronizationSource.source is a number] - expected: FAIL + expected: NOTRUN [[audio\] RTCRtpSynchronizationSource.timestamp is a number] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini index 4256534ee3e2..924a89b73025 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-getStats.https.html.ini @@ -1,6 +1,7 @@ [RTCRtpSender-getStats.https.html] + expected: ERROR [sender.getStats() via addTrack should return stats report containing outbound-rtp stats] - expected: FAIL + expected: NOTRUN [sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini index 821925718a1d..04f02b467342 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-replaceTrack.https.html.ini @@ -1,28 +1,29 @@ [RTCRtpSender-replaceTrack.https.html] + expected: ERROR [Calling replaceTrack on sender with null track and not set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack(null) on sender not set to session description should resolve with sender.track set to null] - expected: FAIL + expected: NOTRUN [Calling replaceTrack with track of different kind should reject with TypeError] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on sender with similar track and and set to session description should resolve with sender.track set to new track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on closed connection should reject with InvalidStateError] expected: FAIL [Calling replaceTrack on sender with stopped track and and set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on stopped sender should reject with InvalidStateError] - expected: FAIL + expected: NOTRUN [Calling replaceTrack on sender not set to session description should resolve with sender.track set to given track] - expected: FAIL + expected: NOTRUN [Calling replaceTrack(null) on sender set to session description should resolve with sender.track set to null] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini index fac8652e7aa7..200008065aef 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini @@ -1,15 +1,16 @@ [RTCRtpSender-transport.https.html] + expected: ERROR [RTCRtpSender/receiver.transport at the right time, with bundle policy balanced] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport at the right time, with bundle policy max-bundle] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport at the right time, with bundle policy max-compat] - expected: FAIL + expected: NOTRUN [RTCRtpSender/receiver.transport has a value when connected] - expected: FAIL + expected: NOTRUN [RTCRtpSender.transport is null when unconnected] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini index c65b2f802864..2958e0a5949f 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -1,109 +1,110 @@ [RTCRtpTransceiver.https.html] + expected: ERROR [checkAddTransceiverWithTrack] expected: FAIL [checkNoMidAnswer] - expected: FAIL + expected: NOTRUN [checkSetDirection] - expected: FAIL + expected: NOTRUN [checkCurrentDirection] - expected: FAIL + expected: NOTRUN [checkReplaceTrackNullDoesntPreventPairing] - expected: FAIL + expected: NOTRUN [checkAddTrackExistingTransceiverThenRemove] - expected: FAIL + expected: NOTRUN [checkMsectionReuse] - expected: FAIL + expected: NOTRUN [checkRemoteRollback] - expected: FAIL + expected: NOTRUN [checkAddTransceiverNoTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkRemoveTrackNegotiation] - expected: FAIL + expected: NOTRUN [checkStopAfterClose] - expected: FAIL + expected: NOTRUN [checkNoMidOffer] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateOffer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverThenAddTrackPairs] - expected: FAIL + expected: NOTRUN [checkLocalRollback] - expected: FAIL + expected: NOTRUN [checkSendrecvWithTracklessStream] - expected: FAIL + expected: NOTRUN [checkMsidNoTrackId] - expected: FAIL + expected: NOTRUN [checkAddTransceiverNoTrack] expected: FAIL [checkAddTransceiverBadKind] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateAnswer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithDirection] - expected: FAIL + expected: NOTRUN [checkAddTrackPairs] - expected: FAIL + expected: NOTRUN [checkStopAfterSetLocalAnswer] - expected: FAIL + expected: NOTRUN [checkStopAfterSetRemoteOffer] - expected: FAIL + expected: NOTRUN [checkStopAfterCreateOfferWithReusedMsection] - expected: FAIL + expected: NOTRUN [checkMute] - expected: FAIL + expected: NOTRUN [checkSendrecvWithNoSendTrack] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithSetRemoteOfferSending] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithAddTrack] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithSetRemoteOfferNoSend] - expected: FAIL + expected: NOTRUN [checkRemoveAndReadd] - expected: FAIL + expected: NOTRUN [checkStopAfterSetLocalOffer] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkAddTransceiverThenReplaceTrackDoesntPair] - expected: FAIL + expected: NOTRUN [checkRollbackAndSetRemoteOfferWithDifferentType] - expected: FAIL + expected: NOTRUN [checkStop] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/__dir__.ini b/tests/wpt/metadata/webrtc/__dir__.ini index 7c0f6049724b..1b83059072f3 100644 --- a/tests/wpt/metadata/webrtc/__dir__.ini +++ b/tests/wpt/metadata/webrtc/__dir__.ini @@ -1 +1 @@ -prefs: ["dom.webrtc.enabled:true", "dom.mediadevices.enabled:false"] +prefs: ["dom.webrtc.enabled:true", "dom.mediadevices.enabled:true", "dom.mediadevices.mock:true"] diff --git a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini index 51a1f678cd96..f06ce24b7e81 100644 --- a/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini +++ b/tests/wpt/metadata/webrtc/idlharness.https.window.js.ini @@ -1181,6 +1181,3 @@ [RTCPeerConnection interface: new RTCPeerConnection() must inherit property "connectionState" with the proper type] expected: FAIL - [Test driver for asyncInitMediaStreamTrack] - expected: FAIL - diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini index 841c26c5f9a3..b9dadb679850 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -1,13 +1,14 @@ [RTCRtpTransceiver-with-OfferToReceive-options.https.html] + expected: ERROR [checkAddTransceiverWithStream] expected: FAIL [checkAddTransceiverWithOfferToReceiveVideo] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithOfferToReceiveBoth] - expected: FAIL + expected: NOTRUN [checkAddTransceiverWithOfferToReceiveAudio] - expected: FAIL + expected: NOTRUN diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini index 71b4cc138f65..7923ea02036a 100644 --- a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -1,4 +1,5 @@ [onaddstream.https.html] + expected: ERROR [Check onaddstream] expected: FAIL From 7b220fac2c3beca531ec951ae1f73cd1bc4f2adf Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Thu, 16 May 2019 09:05:10 -0700 Subject: [PATCH 11/13] Use correct plugin search path for locally cloned gstreamer --- python/servo/command_base.py | 3 +++ 1 file changed, 3 insertions(+) diff --git a/python/servo/command_base.py b/python/servo/command_base.py index 0babff052f3d..a7e6ead3bf07 100644 --- a/python/servo/command_base.py +++ b/python/servo/command_base.py @@ -566,6 +566,9 @@ def set_run_env(self, android=False): if not android and self.needs_gstreamer_env(None): gstpath = self.get_gstreamer_path() os.environ["LD_LIBRARY_PATH"] = path.join(gstpath, "lib") + os.environ["GST_PLUGIN_SYSTEM_PATH"] = path.join(gstpath, "lib", "gstreamer-1.0") + os.environ["PKG_CONFIG_PATH"] = path.join(gstpath, "lib", "pkgconfig") + os.environ["GST_PLUGIN_SCANNER"] = path.join(gstpath, "libexec", "gstreamer-1.0", "gst-plugin-scanner") def build_env(self, hosts_file_path=None, target=None, is_build=False, test_unit=False): """Return an extended environment dictionary.""" From 00c9cc2359322c0213483568d4adfdb0d545e699 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Thu, 16 May 2019 09:08:33 -0700 Subject: [PATCH 12/13] Update metadata --- .../RTCIceTransport-extension.https.html.ini | 58 ++++----- ...ction-add-track-no-deadlock.https.html.ini | 5 +- .../RTCPeerConnection-createAnswer.html.ini | 9 -- ...eerConnection-onnegotiationneeded.html.ini | 3 - ...ion-onsignalingstatechanged.https.html.ini | 5 +- .../webrtc/RTCRtpTransceiver.https.html.ini | 110 +----------------- ...ection-createOffer-offerToReceive.html.ini | 55 +-------- ...with-OfferToReceive-options.https.html.ini | 14 +-- .../webrtc/legacy/onaddstream.https.html.ini | 5 +- .../webrtc/protocol/ice-state.https.html.ini | 10 +- .../protocol/video-codecs.https.html.ini | 10 +- .../webrtc/simplecall-no-ssrcs.https.html.ini | 4 +- .../metadata/webrtc/simplecall.https.html.ini | 4 +- 13 files changed, 39 insertions(+), 253 deletions(-) diff --git a/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini index fbb77db1161c..5675564cb5bb 100644 --- a/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCIceTransport-extension.https.html.ini @@ -1,88 +1,88 @@ [RTCIceTransport-extension.https.html] - [RTCIceTransport initial properties are set] + [gather() returns at least one host candidate] expected: FAIL - [gather() with { iceServers: null } should throw TypeError] + [gather() returns no candidates with { gatherPolicy: 'relay'} and no turn servers] expected: FAIL - [gather() returns no candidates with { gatherPolicy: 'relay'} and no turn servers] + [addRemoteCandidate() throws if closed] expected: FAIL - [gather() throws if called twice] + [Two RTCIceTransports configured with the controlled role resolve the conflict in band and still connect.] expected: FAIL - [gather() returns at least one host candidate] + [onicecandidate fires with null candidate before gatheringState transitions to 'complete'] expected: FAIL - [gather() with one turns server, one turn server, username, credential should succeed] + [addRemoteCandidate() transitions state to 'checking' if start() had been called before] expected: FAIL - [onicecandidate fires with null candidate before gatheringState transitions to 'complete'] + [gather() with 2 stun servers should succeed] expected: FAIL - [start() does not transition state to 'checking' if no remote candidates added] + [start() sets role attribute to 'controlling'] expected: FAIL - [gather() transitions gatheringState to 'gathering'] + [start() throws if closed] expected: FAIL - [start() with default role sets role attribute to 'controlled'] + [Selected candidate pair changes once the RTCIceTransports connect.] expected: FAIL - [eventually transition gatheringState to 'complete'] + [Two RTCIceTransports connect to each other] expected: FAIL - [gather() with { iceServers: undefined } should succeed] + [RTCIceTransport initial properties are set] expected: FAIL - [RTCIceTransport constructor does not throw] + [start() flushes remote candidates and transitions state to 'new' if later called with different remote parameters] expected: FAIL - [gather() with 2 stun servers should succeed] + [start() throws if later called with a different role] expected: FAIL - [start() throws if usernameFragment or password not set] + [gather() with { iceServers: undefined } should succeed] expected: FAIL - [start() sets role attribute to 'controlling'] + [addRemoteCandidate() throws on invalid candidate] expected: FAIL - [start() throws if closed] + [start() with default role sets role attribute to 'controlled'] expected: FAIL - [gather() throws if closed] + [gather() with { iceServers: null } should throw TypeError] expected: FAIL - [addRemoteCandidate() throws if closed] + [gather() throws if called twice] expected: FAIL - [Two RTCIceTransports configured with the controlled role resolve the conflict in band and still connect.] + [Two RTCIceTransports configured with the controlling role resolve the conflict in band and still connect.] expected: FAIL - [addRemoteCandidate() transitions state to 'checking' if start() had been called before] + [gather() transitions gatheringState to 'gathering'] expected: FAIL - [Selected candidate pair changes once the RTCIceTransports connect.] + [start() throws if usernameFragment or password not set] expected: FAIL - [Two RTCIceTransports connect to each other] + [start() transitions state to 'checking' if one remote candidate had been added] expected: FAIL - [start() flushes remote candidates and transitions state to 'new' if later called with different remote parameters] + [gather() with one turns server, one turn server, username, credential should succeed] expected: FAIL - [start() throws if later called with a different role] + [getSelectedCandidatePair() returns null once the RTCIceTransport is stopped.] expected: FAIL - [addRemoteCandidate() throws on invalid candidate] + [start() does not transition state to 'checking' if no remote candidates added] expected: FAIL - [Two RTCIceTransports configured with the controlling role resolve the conflict in band and still connect.] + [gather() throws if closed] expected: FAIL - [start() transitions state to 'checking' if one remote candidate had been added] + [RTCIceTransport constructor does not throw] expected: FAIL - [getSelectedCandidatePair() returns null once the RTCIceTransport is stopped.] + [eventually transition gatheringState to 'complete'] expected: FAIL diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini index 033edaad84f7..f63c63359053 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -1,5 +1,2 @@ [RTCPeerConnection-add-track-no-deadlock.https.html] - expected: ERROR - [RTCPeerConnection addTrack does not deadlock.] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini index 625c956f86ac..7c7fe40e7a9a 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -1,11 +1,2 @@ [RTCPeerConnection-createAnswer.html] expected: CRASH - [createAnswer() when connection is closed reject with InvalidStateError] - expected: NOTRUN - - [createAnswer() after setting remote description should succeed] - expected: NOTRUN - - [createAnswer() with null remoteDescription should reject with InvalidStateError] - expected: TIMEOUT - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index ac85dc6b8990..434392114458 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -33,6 +33,3 @@ [calling createDataChannel twice should fire negotiationneeded event once] expected: FAIL - [negotiationneeded event should not fire if signaling state is not stable] - expected: TIMEOUT - diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini index a81b9e2ce821..2c789d950374 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -1,5 +1,2 @@ [RTCPeerConnection-onsignalingstatechanged.https.html] - expected: ERROR - [RTCPeerConnection onsignalingstatechanged] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini index 2958e0a5949f..6f1426830388 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -1,110 +1,2 @@ [RTCRtpTransceiver.https.html] - expected: ERROR - [checkAddTransceiverWithTrack] - expected: FAIL - - [checkNoMidAnswer] - expected: NOTRUN - - [checkSetDirection] - expected: NOTRUN - - [checkCurrentDirection] - expected: NOTRUN - - [checkReplaceTrackNullDoesntPreventPairing] - expected: NOTRUN - - [checkAddTrackExistingTransceiverThenRemove] - expected: NOTRUN - - [checkMsectionReuse] - expected: NOTRUN - - [checkRemoteRollback] - expected: NOTRUN - - [checkAddTransceiverNoTrackDoesntPair] - expected: NOTRUN - - [checkRemoveTrackNegotiation] - expected: NOTRUN - - [checkStopAfterClose] - expected: NOTRUN - - [checkNoMidOffer] - expected: NOTRUN - - [checkStopAfterCreateOffer] - expected: NOTRUN - - [checkAddTransceiverThenAddTrackPairs] - expected: NOTRUN - - [checkLocalRollback] - expected: NOTRUN - - [checkSendrecvWithTracklessStream] - expected: NOTRUN - - [checkMsidNoTrackId] - expected: NOTRUN - - [checkAddTransceiverNoTrack] - expected: FAIL - - [checkAddTransceiverBadKind] - expected: NOTRUN - - [checkStopAfterCreateAnswer] - expected: NOTRUN - - [checkAddTransceiverWithDirection] - expected: NOTRUN - - [checkAddTrackPairs] - expected: NOTRUN - - [checkStopAfterSetLocalAnswer] - expected: NOTRUN - - [checkStopAfterSetRemoteOffer] - expected: NOTRUN - - [checkStopAfterCreateOfferWithReusedMsection] - expected: NOTRUN - - [checkMute] - expected: NOTRUN - - [checkSendrecvWithNoSendTrack] - expected: NOTRUN - - [checkAddTransceiverWithSetRemoteOfferSending] - expected: NOTRUN - - [checkAddTransceiverWithAddTrack] - expected: NOTRUN - - [checkAddTransceiverWithSetRemoteOfferNoSend] - expected: NOTRUN - - [checkRemoveAndReadd] - expected: NOTRUN - - [checkStopAfterSetLocalOffer] - expected: NOTRUN - - [checkAddTransceiverWithTrackDoesntPair] - expected: NOTRUN - - [checkAddTransceiverThenReplaceTrackDoesntPair] - expected: NOTRUN - - [checkRollbackAndSetRemoteOfferWithDifferentType] - expected: NOTRUN - - [checkStop] - expected: NOTRUN - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini index 874c5a94c9cc..23e34c28583a 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini @@ -1,55 +1,2 @@ [RTCPeerConnection-createOffer-offerToReceive.html] - [offerToReceiveVideo option should be ignored if a non-stopped "recvonly" transceiver exists] - expected: FAIL - - [createOffer() with offerToReceiveAudio should add audio line to all subsequent created offers] - expected: FAIL - - [createOffer() with offerToReceiveAudio set to false should not create a transceiver] - expected: FAIL - - [offerToReceiveVideo option should be ignored if a non-stopped "sendrecv" transceiver exists] - expected: FAIL - - [offerToReceiveAudio set to false with a track should create a "sendonly" transceiver] - expected: FAIL - - [subsequent offerToReceiveAudio set to false with a track should change the direction to "sendonly"] - expected: FAIL - - [offerToReceiveAudio option should be ignored if a non-stopped "sendrecv" transceiver exists] - expected: FAIL - - [createOffer() with offerToReceiveVideo should add video line to all subsequent created offers] - expected: FAIL - - [subsequent offerToReceiveVideo set to false with a track should change the direction to "sendonly"] - expected: FAIL - - [createOffer() with offerToReceiveAudio should create a "recvonly" transceiver] - expected: FAIL - - [offerToReceiveAudio option should be ignored if a non-stopped "recvonly" transceiver exists] - expected: FAIL - - [createOffer() with offerToReceiveAudio:true, then with offerToReceiveVideo:true, should have result offer with both audio and video line] - expected: FAIL - - [offerToReceiveAudio and Video should create two "recvonly" transceivers] - expected: FAIL - - [createOffer() with offerToReceiveVideo set to false should not create a transceiver] - expected: FAIL - - [offerToReceiveVideo set to false with a track should create a "sendonly" transceiver] - expected: FAIL - - [offerToReceiveAudio set to false with a "recvonly" transceiver should change the direction to "inactive"] - expected: FAIL - - [offerToReceiveVideo set to false with a "recvonly" transceiver should change the direction to "inactive"] - expected: FAIL - - [createOffer() with offerToReceiveVideo should create a "recvonly" transceiver] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini index b9dadb679850..e849bfdc474f 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -1,14 +1,2 @@ [RTCRtpTransceiver-with-OfferToReceive-options.https.html] - expected: ERROR - [checkAddTransceiverWithStream] - expected: FAIL - - [checkAddTransceiverWithOfferToReceiveVideo] - expected: NOTRUN - - [checkAddTransceiverWithOfferToReceiveBoth] - expected: NOTRUN - - [checkAddTransceiverWithOfferToReceiveAudio] - expected: NOTRUN - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini index 7923ea02036a..88d269fc834f 100644 --- a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -1,5 +1,2 @@ [onaddstream.https.html] - expected: ERROR - [Check onaddstream] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini b/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini index db02fef95386..c32b2dedfc78 100644 --- a/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini +++ b/tests/wpt/metadata/webrtc/protocol/ice-state.https.html.ini @@ -1,10 +1,2 @@ [ice-state.https.html] - [PC should generate offer with a=ice-options:trickle] - expected: FAIL - - [PC should enter connected state when candidates are sent] - expected: FAIL - - [PC should enter disconnected state when a failing candidate is sent] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini index d316de1f6122..282a32d3182e 100644 --- a/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini +++ b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini @@ -1,10 +1,2 @@ [video-codecs.https.html] - [H.264 and VP8 should be negotiated after handshake] - expected: FAIL - - [H.264 and VP8 should be supported in initial offer] - expected: FAIL - - [All H.264 codecs MUST include profile-level-id] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini index b3469e44234e..430c4586eafa 100644 --- a/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini +++ b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini @@ -1,4 +1,2 @@ [simplecall-no-ssrcs.https.html] - [Can set up a basic WebRTC call without announcing ssrcs.] - expected: FAIL - + expected: CRASH diff --git a/tests/wpt/metadata/webrtc/simplecall.https.html.ini b/tests/wpt/metadata/webrtc/simplecall.https.html.ini index c4522527714c..fd6a475676ac 100644 --- a/tests/wpt/metadata/webrtc/simplecall.https.html.ini +++ b/tests/wpt/metadata/webrtc/simplecall.https.html.ini @@ -1,4 +1,2 @@ [simplecall.https.html] - [Can set up a basic WebRTC call.] - expected: FAIL - + expected: CRASH From 02ce6b0b269f474328ff296f64a04e97d8e85cf2 Mon Sep 17 00:00:00 2001 From: Manish Goregaokar Date: Thu, 16 May 2019 18:02:50 -0700 Subject: [PATCH 13/13] another update? --- ...ction-add-track-no-deadlock.https.html.ini | 5 +- .../RTCPeerConnection-createAnswer.html.ini | 9 +++ ...nnection-iceConnectionState.https.html.ini | 9 ++- ...eerConnection-onnegotiationneeded.html.ini | 3 + ...ion-onsignalingstatechanged.https.html.ini | 5 +- ...onnection-remote-track-mute.https.html.ini | 6 ++ .../RTCRtpSender-transport.https.html.ini | 9 +++ ...tpTransceiver-setCodecPreferences.html.ini | 24 ++++++++ .../webrtc/RTCRtpTransceiver.https.html.ini | 4 +- .../webrtc/RTCTrackEvent-fire.html.ini | 20 ++++++- ...ection-createOffer-offerToReceive.html.ini | 55 ++++++++++++++++++- ...with-OfferToReceive-options.https.html.ini | 14 ++++- .../webrtc/legacy/onaddstream.https.html.ini | 5 +- .../protocol/video-codecs.https.html.ini | 10 +++- .../webrtc/simplecall-no-ssrcs.https.html.ini | 4 +- .../metadata/webrtc/simplecall.https.html.ini | 4 +- 16 files changed, 175 insertions(+), 11 deletions(-) diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini index f63c63359053..033edaad84f7 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html.ini @@ -1,2 +1,5 @@ [RTCPeerConnection-add-track-no-deadlock.https.html] - expected: CRASH + expected: ERROR + [RTCPeerConnection addTrack does not deadlock.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini index 7c7fe40e7a9a..625c956f86ac 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-createAnswer.html.ini @@ -1,2 +1,11 @@ [RTCPeerConnection-createAnswer.html] expected: CRASH + [createAnswer() when connection is closed reject with InvalidStateError] + expected: NOTRUN + + [createAnswer() after setting remote description should succeed] + expected: NOTRUN + + [createAnswer() with null remoteDescription should reject with InvalidStateError] + expected: TIMEOUT + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini index 7bd5bbdf08e3..861d4a1dbce4 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-iceConnectionState.https.html.ini @@ -1,6 +1,7 @@ [RTCPeerConnection-iceConnectionState.https.html] + expected: ERROR [ICE can connect in a recvonly usecase] - expected: FAIL + expected: NOTRUN [connection with one data channel should eventually have connected connection state] expected: FAIL @@ -8,3 +9,9 @@ [connection with one data channel should eventually have connected or completed connection state] expected: FAIL + [connection with audio track should eventually have connected connection state] + expected: FAIL + + [connection with audio and video tracks should eventually have connected connection state] + expected: NOTRUN + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini index 434392114458..ac85dc6b8990 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onnegotiationneeded.html.ini @@ -33,3 +33,6 @@ [calling createDataChannel twice should fire negotiationneeded event once] expected: FAIL + [negotiationneeded event should not fire if signaling state is not stable] + expected: TIMEOUT + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini index 2c789d950374..a81b9e2ce821 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-onsignalingstatechanged.https.html.ini @@ -1,2 +1,5 @@ [RTCPeerConnection-onsignalingstatechanged.https.html] - expected: CRASH + expected: ERROR + [RTCPeerConnection onsignalingstatechanged] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini index 36b8c351cb59..8902a67d99cb 100644 --- a/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCPeerConnection-remote-track-mute.https.html.ini @@ -12,3 +12,9 @@ [ontrack: track goes from muted to unmuted] expected: FAIL + [pc.close() on one side causes mute events on the other] + expected: NOTRUN + + [transceiver.stop() on one side (without renegotiation) causes mute events on the other] + expected: NOTRUN + diff --git a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini index 200008065aef..a2ca0dd2b522 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpSender-transport.https.html.ini @@ -15,3 +15,12 @@ [RTCRtpSender.transport is null when unconnected] expected: FAIL + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy max-compat] + expected: NOTRUN + + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy balanced] + expected: NOTRUN + + [RTCRtpSender/receiver/SCTP transport at the right time, with bundle policy max-bundle] + expected: NOTRUN + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini index e8cfff091f01..edbadaa9d164 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver-setCodecPreferences.html.ini @@ -26,3 +26,27 @@ [setCodecPreferences() on audio transceiver with codecs returned from RTCRtpSender.getCapabilities('audio') should succeed] expected: FAIL + [setCodecPreferences() with user defined codec should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() on audio transceiver with codecs returned from getCapabilities('video') should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with modified codec channel count should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with user defined codec with invalid mimeType should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with modified codecs returned from getCapabilities() should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with modified codec clock rate should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with modified codec parameters should throw InvalidModificationError] + expected: FAIL + + [setCodecPreferences() with user defined codec together with codecs returned from getCapabilities() should throw InvalidModificationError] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini index 6f1426830388..6bedeb73a102 100644 --- a/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini +++ b/tests/wpt/metadata/webrtc/RTCRtpTransceiver.https.html.ini @@ -1,2 +1,4 @@ [RTCRtpTransceiver.https.html] - expected: CRASH + [RTCRtpTransceiver] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini index 59fbbba131fc..aeeed15a1cc5 100644 --- a/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini +++ b/tests/wpt/metadata/webrtc/RTCTrackEvent-fire.html.ini @@ -1,8 +1,26 @@ [RTCTrackEvent-fire.html] expected: TIMEOUT [Applying a remote description with removed msid should trigger firing a removetrack event on the corresponding stream] - expected: TIMEOUT + expected: NOTRUN [Applying a remote description with a new msid should trigger firing an event with populated streams] expected: NOTRUN + [Source-level msid should be ignored, or an error should be thrown, if a different media-level msid is present] + expected: NOTRUN + + [When a=msid is absent, the track should still be associated with a stream] + expected: TIMEOUT + + [stream ids should be found even if msid-semantic is absent] + expected: NOTRUN + + [a=msid:- should result in a track event with no streams] + expected: NOTRUN + + [Source-level msid should be parsed if media-level msid is absent] + expected: NOTRUN + + [Source-level msid should be ignored if media-level msid is present] + expected: NOTRUN + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini index 23e34c28583a..874c5a94c9cc 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCPeerConnection-createOffer-offerToReceive.html.ini @@ -1,2 +1,55 @@ [RTCPeerConnection-createOffer-offerToReceive.html] - expected: CRASH + [offerToReceiveVideo option should be ignored if a non-stopped "recvonly" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveAudio should add audio line to all subsequent created offers] + expected: FAIL + + [createOffer() with offerToReceiveAudio set to false should not create a transceiver] + expected: FAIL + + [offerToReceiveVideo option should be ignored if a non-stopped "sendrecv" transceiver exists] + expected: FAIL + + [offerToReceiveAudio set to false with a track should create a "sendonly" transceiver] + expected: FAIL + + [subsequent offerToReceiveAudio set to false with a track should change the direction to "sendonly"] + expected: FAIL + + [offerToReceiveAudio option should be ignored if a non-stopped "sendrecv" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveVideo should add video line to all subsequent created offers] + expected: FAIL + + [subsequent offerToReceiveVideo set to false with a track should change the direction to "sendonly"] + expected: FAIL + + [createOffer() with offerToReceiveAudio should create a "recvonly" transceiver] + expected: FAIL + + [offerToReceiveAudio option should be ignored if a non-stopped "recvonly" transceiver exists] + expected: FAIL + + [createOffer() with offerToReceiveAudio:true, then with offerToReceiveVideo:true, should have result offer with both audio and video line] + expected: FAIL + + [offerToReceiveAudio and Video should create two "recvonly" transceivers] + expected: FAIL + + [createOffer() with offerToReceiveVideo set to false should not create a transceiver] + expected: FAIL + + [offerToReceiveVideo set to false with a track should create a "sendonly" transceiver] + expected: FAIL + + [offerToReceiveAudio set to false with a "recvonly" transceiver should change the direction to "inactive"] + expected: FAIL + + [offerToReceiveVideo set to false with a "recvonly" transceiver should change the direction to "inactive"] + expected: FAIL + + [createOffer() with offerToReceiveVideo should create a "recvonly" transceiver] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini index e849bfdc474f..b9dadb679850 100644 --- a/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/RTCRtpTransceiver-with-OfferToReceive-options.https.html.ini @@ -1,2 +1,14 @@ [RTCRtpTransceiver-with-OfferToReceive-options.https.html] - expected: CRASH + expected: ERROR + [checkAddTransceiverWithStream] + expected: FAIL + + [checkAddTransceiverWithOfferToReceiveVideo] + expected: NOTRUN + + [checkAddTransceiverWithOfferToReceiveBoth] + expected: NOTRUN + + [checkAddTransceiverWithOfferToReceiveAudio] + expected: NOTRUN + diff --git a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini index 88d269fc834f..7923ea02036a 100644 --- a/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini +++ b/tests/wpt/metadata/webrtc/legacy/onaddstream.https.html.ini @@ -1,2 +1,5 @@ [onaddstream.https.html] - expected: CRASH + expected: ERROR + [Check onaddstream] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini index 282a32d3182e..d316de1f6122 100644 --- a/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini +++ b/tests/wpt/metadata/webrtc/protocol/video-codecs.https.html.ini @@ -1,2 +1,10 @@ [video-codecs.https.html] - expected: CRASH + [H.264 and VP8 should be negotiated after handshake] + expected: FAIL + + [H.264 and VP8 should be supported in initial offer] + expected: FAIL + + [All H.264 codecs MUST include profile-level-id] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini index 430c4586eafa..b3469e44234e 100644 --- a/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini +++ b/tests/wpt/metadata/webrtc/simplecall-no-ssrcs.https.html.ini @@ -1,2 +1,4 @@ [simplecall-no-ssrcs.https.html] - expected: CRASH + [Can set up a basic WebRTC call without announcing ssrcs.] + expected: FAIL + diff --git a/tests/wpt/metadata/webrtc/simplecall.https.html.ini b/tests/wpt/metadata/webrtc/simplecall.https.html.ini index fd6a475676ac..c4522527714c 100644 --- a/tests/wpt/metadata/webrtc/simplecall.https.html.ini +++ b/tests/wpt/metadata/webrtc/simplecall.https.html.ini @@ -1,2 +1,4 @@ [simplecall.https.html] - expected: CRASH + [Can set up a basic WebRTC call.] + expected: FAIL +