[wpt-sync] Sync PR 30226 - Increase bandwidth of fake video signal used for testing.
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(Core :: WebRTC, task, P4)
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Tracking | Status | |
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firefox93 | --- | fixed |
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(Reporter: mozilla.org, Unassigned)
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(Whiteboard: [wptsync downstream])
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Sync web-platform-tests PR 30226 into mozilla-central (this bug is closed when the sync is complete).
PR: https://github.com/web-platform-tests/wpt/pull/30226
Details from upstream follow.
b'Harald Alvestrand <hta@chromium.org>' wrote:
Increase bandwidth of fake video signal used for testing.
This will increase the size of an encoded 640x480 video from
approximately 8 Kbits/second to approximately 64 Kbits/second, which
means that limiting the bandwidth will actually have an effect.This is done in preparation for further tests that limit the bandwidth
and expect an observable result.Bug: None
Change-Id: Ib3c63c4ae8c41fe7c608f1c06c8c61c2beecbe11
Reviewed-on: https://chromium-review.googlesource.com/3128063
WPT-Export-Revision: 1399ee9e31b028459ed1a5d75838c8ed917dddf9
Assignee | ||
Updated•3 years ago
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Assignee | ||
Comment 1•3 years ago
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Pushed to try (stability) https://treeherder.mozilla.org/#/jobs?repo=try&revision=9b465eb133286a6e46ada2c37a22dd2239eca413
Assignee | ||
Comment 2•3 years ago
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# CI Results Ran 0 Firefox configurations based on mozilla-central, and Firefox, Chrome, and Safari on GitHub CI Total 125 tests and 18 subtests ## Status Summary ### Firefox OK : 113 PASS : 1043 FAIL : 626 TIMEOUT: 17 ERROR : 5 NOTRUN : 16 ### Chrome OK : 120 PASS : 1476 FAIL : 213 TIMEOUT: 7 ERROR : 2 NOTRUN : 20 ### Safari OK : 103 PASS : 1068 FAIL : 563 TIMEOUT: 31 ERROR : 7 NOTRUN : 50 ## Links [GitHub PR Head](https://wpt.fyi/results/?sha=3c8b691b4fc045b1aca3fb431acf760e95f23584&label=pr_head) [GitHub PR Base](https://wpt.fyi/results/?sha=3c8b691b4fc045b1aca3fb431acf760e95f23584&label=pr_base) ## Details ### Firefox-only Failures /html/dom/usvstring-reflection.https.html anchor : unpaired surrogate codepoint should be replaced with U+FFFD: FAIL linked bug:Bug 1645268 area : unpaired surrogate codepoint should be replaced with U+FFFD: FAIL linked bug:Bug 1645268 source : unpaired surrogate codepoint should be replaced with U+FFFD: FAIL linked bug:Bug 1645268 storage event : unpaired surrogate codepoint should be replaced with U+FFFD: FAIL linked bug:Bug 1645268 /mst-content-hint/idlharness.window.html MediaStreamTrack interface: attribute contentHint: FAIL MediaStreamTrack interface: audioTrack must inherit property "contentHint" with the proper type: FAIL MediaStreamTrack interface: videoTrack must inherit property "contentHint" with the proper type: FAIL /webrtc/RTCConfiguration-rtcpMuxPolicy.html new RTCPeerConnection() should have default rtcpMuxPolicy require: FAIL new RTCPeerConnection({ rtcpMuxPolicy: undefined }) should have default rtcpMuxPolicy require: FAIL new RTCPeerConnection({ rtcpMuxPolicy: 'require' }) should succeed: FAIL new RTCPeerConnection({ rtcpMuxPolicy: 'negotiate' }) may succeed or throw NotSupportedError: FAIL new RTCPeerConnection(config) - with { rtcpMuxPolicy: null } should throw TypeError: FAIL setConfiguration(config) - with { rtcpMuxPolicy: null } should throw TypeError: FAIL new RTCPeerConnection(config) - with { rtcpMuxPolicy: 'invalid' } should throw TypeError: FAIL setConfiguration(config) - with { rtcpMuxPolicy: 'invalid' } should throw TypeError: FAIL setConfiguration({ rtcpMuxPolicy: 'negotiate' }) with initial rtcpMuxPolicy require should throw InvalidModificationError: FAIL setConfiguration({ rtcpMuxPolicy: 'require' }) with initial rtcpMuxPolicy negotiate should throw InvalidModificationError: FAIL setConfiguration({}) with initial rtcpMuxPolicy negotiate should throw InvalidModificationError: FAIL setRemoteDescription throws InvalidAccessError when called with an offer without rtcp-mux and rtcpMuxPolicy is set to require: FAIL setRemoteDescription throws InvalidAccessError when called with an answer without rtcp-mux and rtcpMuxPolicy is set to require: FAIL /webrtc/RTCDataChannel-iceRestart.html: ERROR /webrtc/RTCPeerConnection-connectionState.https.html: TIMEOUT Initial connectionState should be new: FAIL Closing the connection should set connectionState to closed: FAIL connection with one data channel should eventually have connected connection state: TIMEOUT connectionState transitions to connected via connecting: TIMEOUT Closing a PeerConnection should not fire connectionstatechange event: NOTRUN /webrtc/RTCPeerConnection-getStats.https.html: TIMEOUT getStats() with no argument should return stats report containing peer-connection stats on an empty PC: FAIL getStats() track with stream returns peer-connection and outbound-rtp stats: TIMEOUT getStats() track without stream returns peer-connection and outbound-rtp stats: NOTRUN getStats() audio outbound-rtp contains all mandatory stats: NOTRUN getStats() video outbound-rtp contains all mandatory stats: NOTRUN getStats() on track associated with RTCRtpSender should return stats report containing outbound-rtp stats: NOTRUN getStats() on track associated with RTCRtpReceiver should return stats report containing inbound-rtp stats: NOTRUN getStats() inbound-rtp contains all mandatory stats: NOTRUN RTCStats.timestamp increases with time passing: NOTRUN /webrtc/RTCPeerConnection-mandatory-getStats.https.html RTCRtpStreamStats's transportId: FAIL RTCRtpStreamStats's codecId: FAIL RTCInboundRtpStreamStats's framesReceived: FAIL RTCInboundRtpStreamStats's totalAudioEnergy: FAIL RTCInboundRtpStreamStats's totalSamplesDuration: FAIL RTCMediaSourceStats's trackIdentifier: FAIL RTCMediaSourceStats's kind: FAIL RTCCodecStats's payloadType: FAIL RTCCodecStats's mimeType: FAIL RTCCodecStats's clockRate: FAIL RTCCodecStats's channels: FAIL RTCCodecStats's sdpFmtpLine: FAIL RTCIceCandidatePairStats's totalRoundTripTime: FAIL RTCIceCandidatePairStats's currentRoundTripTime: FAIL RTCCertificateStats's fingerprint: FAIL RTCCertificateStats's fingerprintAlgorithm: FAIL RTCCertificateStats's base64Certificate: FAIL /webrtc/RTCPeerConnection-onnegotiationneeded.html Calling setStreams should cause negotiationneeded to fire: FAIL /webrtc/RTCPeerConnection-setDescription-transceiver.html setRemoteDescription should set transceiver inactive if its corresponding m section is rejected: FAIL /webrtc/RTCPeerConnection-setLocalDescription-pranswer.html setLocalDescription(pranswer) should succeed: FAIL setLocalDescription(pranswer) can be applied multiple times while still in have-local-pranswer: FAIL setLocalDescription(answer) from have-local-pranswer state should succeed: FAIL /webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html setRemoteDescription(pranswer) from stable state should reject with InvalidStateError: FAIL setRemoteDescription(pranswer) from have-local-offer state should succeed: FAIL setRemoteDescription(pranswer) multiple times should succeed: FAIL setRemoteDescription(answer) from have-remote-pranswer state should succeed: FAIL /webrtc/RTCPeerConnection-track-stats.https.html: TIMEOUT addTrack() without setLocalDescription() yields track stats: FAIL addTrack() with setLocalDescription() yields track stats: FAIL O/A exchange yields outbound RTP stream stats for sending track: FAIL O/A exchange yields inbound RTP stream stats for receiving track: FAIL replaceTrack() before offer: new track attachment stats present: FAIL replaceTrack() after offer, before answer: new track attachment stats present: FAIL replaceTrack() after answer: new track attachment stats present: FAIL RTCRtpSender.getStats() contains only outbound-rtp and related stats: TIMEOUT RTCRtpReceiver.getStats() contains only inbound-rtp and related stats: NOTRUN RTCPeerConnection.getStats(sendingTrack) is the same as RTCRtpSender.getStats(): NOTRUN RTCPeerConnection.getStats(receivingTrack) is the same as RTCRtpReceiver.getStats(): NOTRUN /webrtc/RTCPeerConnection-transceivers.https.html addTransceiver(track, init): initialize sendEncodings[0].active to false: FAIL /webrtc/RTCRtpSender-setStreams.https.html setStreams causes streams to be reported via ontrack on callee: FAIL setStreams can be used to reconstruct a stream with a track on the remote side: FAIL Adding streams and changing direction causes new streams to be reported via ontrack on callee: FAIL setStreams() fires InvalidStateError on a closed peer connection.: FAIL /webrtc/idlharness.https.window.html RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit): FAIL RTCPeerConnection interface: attribute connectionState: FAIL RTCPeerConnection interface: attribute onicecandidateerror: FAIL RTCPeerConnection interface: attribute onconnectionstatechange: FAIL RTCPeerConnection interface: operation setRemoteDescription(RTCSessionDescriptionInit, VoidFunction, RTCPeerConnectionErrorCallback): FAIL RTCPeerConnection interface: new RTCPeerConnection() must inherit property "connectionState" with the proper type: FAIL RTCPeerConnection interface: new RTCPeerConnection() must inherit property "setConfiguration(optional RTCConfiguration)" with the proper type: FAIL RTCPeerConnection interface: calling setConfiguration(optional RTCConfiguration) on new RTCPeerConnection() with too few arguments must throw TypeError: FAIL RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onicecandidateerror" with the proper type: FAIL RTCPeerConnection interface: new RTCPeerConnection() must inherit property "onconnectionstatechange" with the proper type: FAIL RTCIceCandidate interface: attribute candidate: FAIL RTCIceCandidate interface: attribute sdpMid: FAIL RTCIceCandidate interface: attribute sdpMLineIndex: FAIL RTCIceCandidate interface: attribute foundation: FAIL RTCIceCandidate interface: attribute component: FAIL RTCIceCandidate interface: attribute priority: FAIL RTCIceCandidate interface: attribute address: FAIL RTCIceCandidate interface: attribute protocol: FAIL RTCIceCandidate interface: attribute port: FAIL RTCIceCandidate interface: attribute type: FAIL RTCIceCandidate interface: attribute tcpType: FAIL RTCIceCandidate interface: attribute relatedAddress: FAIL RTCIceCandidate interface: attribute relatedPort: FAIL RTCIceCandidate interface: attribute usernameFragment: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "foundation" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "component" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "priority" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "address" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "protocol" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "port" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "type" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "tcpType" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedAddress" with the proper type: FAIL RTCIceCandidate interface: new RTCIceCandidate({ sdpMid: 1 }) must inherit property "relatedPort" with the proper type: FAIL RTCPeerConnectionIceErrorEvent interface: existence and properties of interface object: FAIL RTCPeerConnectionIceErrorEvent interface object length: FAIL RTCPeerConnectionIceErrorEvent interface object name: FAIL RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object: FAIL RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's "constructor" property: FAIL RTCPeerConnectionIceErrorEvent interface: existence and properties of interface prototype object's @@unscopables property: FAIL RTCPeerConnectionIceErrorEvent interface: attribute address: FAIL RTCPeerConnectionIceErrorEvent interface: attribute port: FAIL RTCPeerConnectionIceErrorEvent interface: attribute url: FAIL RTCPeerConnectionIceErrorEvent interface: attribute errorCode: FAIL RTCPeerConnectionIceErrorEvent interface: attribute errorText: FAIL RTCPeerConnectionIceErrorEvent must be primary interface of new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 });: FAIL Stringification of new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 });: FAIL RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "address" with the proper type: FAIL RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "port" with the proper type: FAIL RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "url" with the proper type: FAIL RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "errorCode" with the proper type: FAIL RTCPeerConnectionIceErrorEvent interface: new RTCPeerConnectionIceErrorEvent('ice-error', { port: 0, errorCode: 701 }); must inherit property "errorText" with the proper type: FAIL RTCCertificate interface: operation getFingerprints(): FAIL RTCCertificate interface: idlTestObjects.certificate must inherit property "getFingerprints()" with the proper type: FAIL RTCRtpSender interface: operation getCapabilities(DOMString): FAIL RTCRtpSender interface: operation setParameters(RTCRtpSendParameters): FAIL RTCRtpSender interface: operation setStreams(MediaStream...): FAIL RTCRtpSender interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError: FAIL RTCRtpSender interface: calling setParameters(RTCRtpSendParameters) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError: FAIL RTCRtpSender interface: new RTCPeerConnection().addTransceiver('audio').sender must inherit property "setStreams(MediaStream...)" with the proper type: FAIL RTCRtpSender interface: calling setStreams(MediaStream...) on new RTCPeerConnection().addTransceiver('audio').sender with too few arguments must throw TypeError: FAIL RTCRtpReceiver interface: operation getCapabilities(DOMString): FAIL RTCRtpReceiver interface: operation getParameters(): FAIL RTCRtpReceiver interface: calling getCapabilities(DOMString) on new RTCPeerConnection().addTransceiver('audio').receiver with too few arguments must throw TypeError: FAIL RTCRtpReceiver interface: new RTCPeerConnection().addTransceiver('audio').receiver must inherit property "getParameters()" with the proper type: FAIL RTCRtpTransceiver interface: operation setCodecPreferences(sequence<RTCRtpCodecCapability>): FAIL RTCRtpTransceiver interface: new RTCPeerConnection().addTransceiver('audio') must inherit property "setCodecPreferences(sequence<RTCRtpCodecCapability>)" with the proper type: FAIL RTCRtpTransceiver interface: calling setCodecPreferences(sequence<RTCRtpCodecCapability>) on new RTCPeerConnection().addTransceiver('audio') with too few arguments must throw TypeError: FAIL RTCIceTransport interface object length: FAIL RTCIceTransport interface object name: FAIL RTCIceTransport interface: existence and properties of interface prototype object's "constructor" property: FAIL RTCIceTransport interface: existence and properties of interface prototype object's @@unscopables property: FAIL RTCIceTransport interface: attribute state: FAIL RTCIceTransport interface: attribute gatheringState: FAIL RTCDTMFSender interface: attribute canInsertDTMF: FAIL /webrtc/protocol/handover.html Negotiation of handover initiated at caller works: FAIL Negotiation of handover initiated at callee works: FAIL /webrtc/protocol/ice-ufragpwd.html setRemoteDescription with a ice-ufrag containing a non-ice-char fails: FAIL setRemoteDescription with a ice-pwd containing a non-ice-char fails: FAIL /webrtc/protocol/rtp-clockrate.html: TIMEOUT video rtp timestamps increase by approximately 90000 per second: TIMEOUT /webrtc/protocol/rtp-demuxing.html Can demux two video tracks with different payload types on a bundled connection: FAIL /webrtc/protocol/video-codecs.https.html H.264 and VP8 should be supported in initial offer: FAIL linked bug:Bug 1534688 H.264 and VP8 should be negotiated after handshake: FAIL linked bug:Bug 1534687 All H.264 codecs MUST include profile-level-id: FAIL linked bug:Bug 1534687 /webrtc/simulcast/basic.https.html Basic simulcast setup with two spatial layers: FAIL /webrtc/simulcast/getStats.https.html Simulcast getStats results: FAIL /webrtc/simulcast/h264.https.html H264 simulcast setup with two spatial layers: FAIL /webrtc/simulcast/setParameters-active.https.html Simulcast setParameters active=false stops sending frames: FAIL /webrtc/simulcast/vp8.https.html VP8 simulcast setup with two spatial layers: FAIL /webrtc-stats/supported-stats.html codec's payloadType: FAIL codec's mimeType: FAIL codec's clockRate: FAIL codec's channels: FAIL codec's sdpFmtpLine: FAIL codec's timestamp: FAIL codec's type: FAIL codec's id: FAIL inbound-rtp's keyFramesDecoded: FAIL inbound-rtp's frameWidth: FAIL inbound-rtp's frameHeight: FAIL inbound-rtp's framesPerSecond: FAIL inbound-rtp's totalDecodeTime: FAIL inbound-rtp's totalInterFrameDelay: FAIL inbound-rtp's totalSquaredInterFrameDelay: FAIL inbound-rtp's lastPacketReceivedTimestamp: FAIL inbound-rtp's headerBytesReceived: FAIL inbound-rtp's fecPacketsReceived: FAIL inbound-rtp's fecPacketsDiscarded: FAIL inbound-rtp's jitterBufferDelay: FAIL inbound-rtp's jitterBufferEmittedCount: FAIL inbound-rtp's totalSamplesReceived: FAIL inbound-rtp's concealedSamples: FAIL inbound-rtp's silentConcealedSamples: FAIL inbound-rtp's concealmentEvents: FAIL inbound-rtp's insertedSamplesForDeceleration: FAIL inbound-rtp's removedSamplesForAcceleration: FAIL inbound-rtp's audioLevel: FAIL inbound-rtp's totalAudioEnergy: FAIL inbound-rtp's totalSamplesDuration: FAIL inbound-rtp's framesReceived: FAIL inbound-rtp's transportId: FAIL inbound-rtp's codecId: FAIL outbound-rtp's mediaSourceId: FAIL outbound-rtp's headerBytesSent: FAIL outbound-rtp's retransmittedPacketsSent: FAIL outbound-rtp's retransmittedBytesSent: FAIL outbound-rtp's totalEncodedBytesTarget: FAIL outbound-rtp's frameWidth: FAIL outbound-rtp's frameHeight: FAIL outbound-rtp's framesPerSecond: FAIL outbound-rtp's framesSent: FAIL outbound-rtp's hugeFramesSent: FAIL outbound-rtp's keyFramesEncoded: FAIL outbound-rtp's totalEncodeTime: FAIL outbound-rtp's totalPacketSendDelay: FAIL outbound-rtp's qualityLimitationResolutionChanges: FAIL outbound-rtp's transportId: FAIL outbound-rtp's codecId: FAIL remote-inbound-rtp's transportId: FAIL remote-inbound-rtp's codecId: FAIL peer-connection's dataChannelsOpened: FAIL peer-connection's dataChannelsClosed: FAIL peer-connection's timestamp: FAIL peer-connection's type: FAIL peer-connection's id: FAIL media-source's trackIdentifier: FAIL media-source's kind: FAIL media-source's timestamp: FAIL media-source's type: FAIL media-source's id: FAIL transport's bytesSent: FAIL transport's bytesReceived: FAIL transport's dtlsState: FAIL transport's selectedCandidatePairId: FAIL transport's localCertificateId: FAIL transport's remoteCertificateId: FAIL transport's tlsVersion: FAIL transport's dtlsCipher: FAIL transport's srtpCipher: FAIL transport's timestamp: FAIL transport's type: FAIL transport's id: FAIL candidate-pair's totalRoundTripTime: FAIL candidate-pair's currentRoundTripTime: FAIL candidate-pair's availableOutgoingBitrate: FAIL candidate-pair's requestsReceived: FAIL candidate-pair's requestsSent: FAIL candidate-pair's responsesReceived: FAIL candidate-pair's responsesSent: FAIL local-candidate's transportId: FAIL remote-candidate's transportId: FAIL certificate's fingerprint: FAIL certificate's fingerprintAlgorithm: FAIL certificate's base64Certificate: FAIL certificate's timestamp: FAIL certificate's type: FAIL certificate's id: FAIL /webrtc-svc/RTCRtpParameters-scalability.html: ERROR (See attachment for full changes)
Pushed by wptsync@mozilla.com: https://hg.mozilla.org/integration/autoland/rev/ed61a5397257 [wpt PR 30226] - Increase bandwidth of fake video signal used for testing., a=testonly https://hg.mozilla.org/integration/autoland/rev/80b0ae7f36a1 [wpt PR 30226] - Update wpt metadata, a=testonly
Comment 4•3 years ago
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bugherder |
https://hg.mozilla.org/mozilla-central/rev/ed61a5397257
https://hg.mozilla.org/mozilla-central/rev/80b0ae7f36a1
Description
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