зеркало из https://github.com/mozilla/gecko-dev.git
Bug 1733656: Clean up wpt meta files. r=jib
Differential Revision: https://phabricator.services.mozilla.com/D128622
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92536e77f0
Коммит
fa4a762db3
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@ -1,58 +1,4 @@
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[RTCRtpReceiver-getSynchronizationSources.https.html]
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[[audio-only\] RTCRtpSynchronizationSource.voiceActivityFlag is a boolean]
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1525394
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expected:
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if (processor == "x86") and not debug: [FAIL, NOTRUN]
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FAIL
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[[audio\] getSynchronizationSources() eventually returns a non-empty list]
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expected:
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if (processor == "x86") and not debug: [PASS, TIMEOUT]
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[[audio\] RTCRtpSynchronizationSource.timestamp is a number]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[audio\] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1\]]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[audio\] getSynchronizationSources() does not contain SSRCs older than 10 seconds]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[audio\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[audio\] RTCRtpSynchronizationSource.source is a number]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] getSynchronizationSources() eventually returns a non-empty list]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] RTCRtpSynchronizationSource.timestamp is a number]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] RTCRtpSynchronizationSource.rtpTimestamp is a number [0, 2^32-1\]]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] getSynchronizationSources() does not contain SSRCs older than 10 seconds]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] RTCRtpSynchronizationSource.timestamp is comparable to performance.timeOrigin + performance.now()]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[video\] RTCRtpSynchronizationSource.source is a number]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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[[audio-only\] RTCRtpSynchronizationSource.audioLevel is a number [0, 1\]]
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expected:
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if (processor == "x86") and not debug: [PASS, NOTRUN]
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expected: FAIL
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@ -1,7 +1,9 @@
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[RTCRtpTransceiver-stop.html]
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[If a transceiver is stopped, transceivers should end up in state stopped]
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296
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expected: FAIL
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[If a transceiver is stopped, transceivers, senders and receivers should disappear after offer/answer]
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1568296
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expected: FAIL
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@ -1,4 +0,0 @@
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[simplecall-no-ssrcs.https.html]
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expected:
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if (os == "linux") and not debug and webrender and not fission: [OK, ERROR, CRASH]
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if (os == "linux") and not debug and not webrender: [OK, ERROR, CRASH]
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@ -1,11 +0,0 @@
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[simplecall.https.html]
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expected:
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if (os == "linux") and not fission and not debug and (processor == "x86"): [OK, ERROR, TIMEOUT, CRASH]
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if (os == "linux") and not fission and not debug and (processor == "x86_64"): [OK, TIMEOUT, ERROR, CRASH]
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if (os == "linux") and fission: [OK, ERROR, CRASH]
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if (os == "mac") and not debug: [OK, TIMEOUT]
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[Can set up a basic WebRTC call.]
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expected:
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if (os == "linux") and not fission and not debug and (processor == "x86"): [PASS, TIMEOUT]
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if (os == "linux") and not fission and not debug and (processor == "x86_64"): [PASS, TIMEOUT]
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if (os == "mac") and not debug: [PASS, TIMEOUT]
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