author | Philipp Hancke <philipp.hancke@googlemail.com> |
Tue, 15 May 2018 13:30:12 +0000 | |
changeset 418485 | 7142ea1ef1d646980cc33a2960ebcb0fa94a54f8 |
parent 418484 | 96819f2142ae4142d3661a49e0423c7791be6433 |
child 418486 | 816d4ef0a3dea89726812659ab06b401a4d7f94e |
push id | 103318 |
push user | wptsync@mozilla.com |
push date | Wed, 16 May 2018 15:06:09 +0000 |
treeherder | mozilla-inbound@51f1ecd79ebe [default view] [failures only] |
perfherder | [talos] [build metrics] [platform microbench] (compared to previous push) |
reviewers | testonly |
bugs | 1459903, 10893, 1047674, 836871, 1049865, 556795 |
milestone | 62.0a1 |
first release with | nightly linux32
nightly linux64
nightly mac
nightly win32
nightly win64
|
last release without | nightly linux32
nightly linux64
nightly mac
nightly win32
nightly win64
|
--- a/testing/web-platform/meta/MANIFEST.json +++ b/testing/web-platform/meta/MANIFEST.json @@ -613738,17 +613738,17 @@ "a9beda60053916185106aedac98014123d6f7105", "testharness" ], "webrtc/RTCDTMFSender-helper.js": [ "26b5336881897c93fafd1ddf32910cf6fef16987", "support" ], "webrtc/RTCDTMFSender-insertDTMF.https.html": [ - "79574cff7e0500cecaf7e3ae182e89d98f49ce72", + "a5fc15d44fe572744c07d933b9204d9319c968cd", "testharness" ], "webrtc/RTCDTMFSender-ontonechange-long.https.html": [ "d89602fd655bf032cadfc538291ccea858eb2446", "testharness" ], "webrtc/RTCDTMFSender-ontonechange.https.html": [ "ecc1e77f083cb91af78b1bcb7311fe4b5b96313e", @@ -613774,33 +613774,33 @@ "7d6ea59253879cac114e2a7b162fa67926a39635", "testharness" ], "webrtc/RTCIceCandidate-constructor.html": [ "6938c88a0167e418aa9e93416865c857cc3489c5", "testharness" ], "webrtc/RTCIceTransport.html": [ - "c145a8a34c79dd970475e77ff5bf1a363c0ac56c", + "db758cc2a744c049c291575e408dbb5f280cdf19", "testharness" ], "webrtc/RTCPeerConnection-addIceCandidate.html": [ "dd19f1d7a8d12ee85101e53bb30c553e94d67b6a", "testharness" ], "webrtc/RTCPeerConnection-addTrack.https.html": [ "c434d2cdcb134f28b203df07cecca04e11195700", "testharness" ], "webrtc/RTCPeerConnection-addTransceiver.html": [ "c2d5766daa3ea4050ccb2777d7c08af1a1bd176f", "testharness" ], "webrtc/RTCPeerConnection-canTrickleIceCandidates.html": [ - "0f585a89bd8f25aa8f83b6ec39b704cbb8e970b2", + "8401fdc22f8f8867aa361f6a83834cdeb7a2a9d1", "testharness" ], "webrtc/RTCPeerConnection-connectionState.html": [ "a733cd1ae59aace10832a7b5f98913967afb87f1", "testharness" ], "webrtc/RTCPeerConnection-constructor.html": [ "c229347757f56d239925915ed9e6227086e75b84", @@ -613830,17 +613830,17 @@ "208bb45887440df3bf1e45dd63f09d2d5b70857d", "testharness" ], "webrtc/RTCPeerConnection-getIdentityAssertion.html": [ "91b55a4f1d9a10cda7cb0e11ba42243bf94a0dfa", "testharness" ], "webrtc/RTCPeerConnection-getStats.https.html": [ - "9446d7bc1aefa7edd28b425415d983d69311e0ca", + "913cbc3d2aaf724e70108e7854f56ad5bb9b2283", "testharness" ], "webrtc/RTCPeerConnection-getTransceivers.html": [ "b4c97af4f907a3d02fe1ebd24f00ab110b387575", "testharness" ], "webrtc/RTCPeerConnection-helper.js": [ "d579dd68118d72c06455d8ccdbeb666f8f39c58a", @@ -613862,25 +613862,25 @@ "ca1cbd230de7aec4844879ae43f822941f566620", "testharness" ], "webrtc/RTCPeerConnection-ontrack.https.html": [ "3db4d8b29f4e1372055a50a279cae525f52cbb40", "testharness" ], "webrtc/RTCPeerConnection-peerIdentity.html": [ - "1cc5702e0aee887d925d2bf3471ac759d7430874", + "5aa9f3d712dd320cc85645abd39f960b5072349b", "testharness" ], "webrtc/RTCPeerConnection-removeTrack.https.html": [ "561575bea206ec1c9572e1e5e6f97d1e0bebe2d1", "testharness" ], "webrtc/RTCPeerConnection-setDescription-transceiver.html": [ - "0f998108088cee211977870f9c64f2a89bef7bf0", + "a21fe04592ad6941aa4277535d6482519b67ae74", "testharness" ], "webrtc/RTCPeerConnection-setLocalDescription-answer.html": [ "e215aa042c67a23ae776b83d662a035a22e03810", "testharness" ], "webrtc/RTCPeerConnection-setLocalDescription-offer.html": [ "117fc91599d11b63f2d232a63bace8e367dbb72a", @@ -613978,25 +613978,25 @@ "ed910cbe15534cae43b79cc008395bd62fbd0637", "testharness" ], "webrtc/RTCRtpReceiver-getStats.https.html": [ "ac5c0244fe64a5c07a7d255003c783e27a699728", "testharness" ], "webrtc/RTCRtpReceiver-getSynchronizationSources.https.html": [ - "56d0157f4ce987436c12ddff886b74549abbe682", + "11aa1d9f6833dd019ae7ade7b9ec14780f271650", "testharness" ], "webrtc/RTCRtpSender-getCapabilities.html": [ "27f083617973770f0d42efb93813f0112fc68ad2", "testharness" ], "webrtc/RTCRtpSender-getStats.https.html": [ - "64c4424e36c566294a317fb423eb02e97a9ebbca", + "ee215306e1d9d1fdcb65bd5244da09fb2005e799", "testharness" ], "webrtc/RTCRtpSender-replaceTrack.html": [ "f7b8caa578c9c818e8ca11777daf664ccb9457ec", "testharness" ], "webrtc/RTCRtpSender-setParameters.html": [ "615bd9591e0f9c873827c9ae8e787b12d2efaf0f",
--- a/testing/web-platform/tests/webrtc/RTCDTMFSender-insertDTMF.https.html +++ b/testing/web-platform/tests/webrtc/RTCDTMFSender-insertDTMF.https.html @@ -143,17 +143,17 @@ /* 7.2. insertDTMF The characters a to d MUST be normalized to uppercase on entry and are equivalent to A to D. 7. Set the object's toneBuffer attribute to tones. */ - promise_test(() => { + promise_test(t => { return createDtmfSender() .then(dtmfSender => { dtmfSender.insertDTMF('123'); assert_equals(dtmfSender.toneBuffer, '123'); dtmfSender.insertDTMF('ABC'); assert_equals(dtmfSender.toneBuffer, 'ABC');
--- a/testing/web-platform/tests/webrtc/RTCIceTransport.html +++ b/testing/web-platform/tests/webrtc/RTCIceTransport.html @@ -113,17 +113,17 @@ assert_true(candidatePair.remote instanceof RTCIceCandidate, 'Expect candidatePair.remote to be instance of RTCIceCandidate'); validateCandidateParameter(iceTransport.getLocalParameters()); validateCandidateParameter(iceTransport.getRemoteParameters()); } - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); return createDataChannelPair(pc1, pc2) .then(([channel1, channel2]) => { // Send a ping message and wait for it just to make sure // that the connection is fully working before testing channel1.send('ping'); @@ -158,17 +158,17 @@ assert_equals(iceTransport1.role, 'controlling', `Expect offerer's iceTransport to take the controlling role`); assert_equals(iceTransport2.role, 'controlled', `Expect answerer's iceTransport to take the controlled role`); }); }, 'Two connected iceTransports should has matching local/remote candidates returned'); - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); pc1.createDataChannel(''); // setRemoteDescription(answer) without the other peer // setting answer it's localDescription return pc1.createOffer() .then(offer =>
--- a/testing/web-platform/tests/webrtc/RTCPeerConnection-canTrickleIceCandidates.html +++ b/testing/web-platform/tests/webrtc/RTCPeerConnection-canTrickleIceCandidates.html @@ -30,26 +30,26 @@ 'a=ssrc:1001 cname:some\r\n' + 'a=rtpmap:111 opus/48000/2\r\n'; test(function() { var pc = new RTCPeerConnection(); assert_equals(pc.canTrickleIceCandidates, null, 'canTrickleIceCandidates property is null'); }, 'canTrickleIceCandidates property is null prior to setRemoteDescription'); - promise_test(function() { + promise_test(function(t) { var pc = new RTCPeerConnection(); return pc.setRemoteDescription(new RTCSessionDescription({type: 'offer', sdp: sdp})) .then(function() { assert_true(pc.canTrickleIceCandidates, 'canTrickleIceCandidates property is true after setRemoteDescription'); }) }, 'canTrickleIceCandidates property is true after setRemoteDescription with a=ice-options:trickle'); - promise_test(function() { + promise_test(function(t) { var pc = new RTCPeerConnection(); return pc.setRemoteDescription(new RTCSessionDescription({type: 'offer', sdp: sdp.replace('a=ice-options:trickle\r\n', '')})) .then(function() { assert_false(pc.canTrickleIceCandidates, 'canTrickleIceCandidates property is false after setRemoteDescription'); }) }, 'canTrickleIceCandidates property is false after setRemoteDescription without a=ice-options:trickle'); </script>
--- a/testing/web-platform/tests/webrtc/RTCPeerConnection-getStats.https.html +++ b/testing/web-platform/tests/webrtc/RTCPeerConnection-getStats.https.html @@ -34,22 +34,22 @@ If no such sender or receiver exists, or if more than one sender or receiver fit this criteria, return a promise rejected with a newly created InvalidAccessError. 5. Let p be a new promise. 6. Run the following steps in parallel: 1. Gather the stats indicated by selector according to the stats selection algorithm. 2. Resolve p with the resulting RTCStatsReport object, containing the gathered stats. */ - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); return pc.getStats(); }, 'getStats() with no argument should succeed'); - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); return pc.getStats(null); }, 'getStats(null) should succeed'); /* 8.2. getStats 4. If selectorArg is a MediaStreamTrack let selector be an RTCRtpSender or RTCRtpReceiver on connection which track member matches selectorArg. @@ -155,17 +155,17 @@ /* 8.5. The stats selection algorithm 3. If selector is an RTCRtpSender, gather stats for and add the following objects to result: - All RTCOutboundRTPStreamStats objects corresponding to selector. - All stats objects referenced directly or indirectly by the RTCOutboundRTPStreamStats objects added. */ - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); return getTrackFromUserMedia('audio') .then(([track, mediaStream]) => { pc.addTrack(track, mediaStream); return pc.getStats(track) .then(statsReport => { validateStatsReport(statsReport); @@ -178,17 +178,17 @@ /* 8.5. The stats selection algorithm 4. If selector is an RTCRtpReceiver, gather stats for and add the following objects to result: - All RTCInboundRTPStreamStats objects corresponding to selector. - All stats objects referenced directly or indirectly by the RTCInboundRTPStreamStats added. */ - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); const transceiver = pc.addTransceiver('audio'); return pc.getStats(transceiver.receiver.track) .then(statsReport => { validateStatsReport(statsReport); assert_stats_report_has_stats(statsReport, ['inbound-rtp']); });
--- a/testing/web-platform/tests/webrtc/RTCPeerConnection-peerIdentity.html +++ b/testing/web-platform/tests/webrtc/RTCPeerConnection-peerIdentity.html @@ -49,17 +49,17 @@ validates the identity assertion.. If the "peerIdentity" configuration is applied to the RTCPeerConnection, this establishes a target peer identity of the provided value. Alternatively, if the RTCPeerConnection has previously authenticated the identity of the peer (that is, there is a current value for peerIdentity ), then this also establishes a target peer identity. */ - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); const port = window.location.port; const [idpDomain] = getIdpDomains(); const idpHost = hostString(idpDomain, port); pc1.setIdentityProvider(idpHost, {
--- a/testing/web-platform/tests/webrtc/RTCPeerConnection-setDescription-transceiver.html +++ b/testing/web-platform/tests/webrtc/RTCPeerConnection-setDescription-transceiver.html @@ -59,17 +59,17 @@ 7. If description is set as a local description, then run the following steps for each media description in description that is not yet associated with an RTCRtpTransceiver object: 1. Let transceiver be the RTCRtpTransceiver used to create the media description. 2. Set transceiver's mid value to the mid of the corresponding media description. */ - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); const transceiver = pc.addTransceiver('audio'); assert_equals(transceiver.mid, null); return pc.createOffer() .then(offer => { assert_equals(transceiver.mid, null, 'Expect transceiver.mid to still be null after createOffer'); @@ -92,17 +92,17 @@ 2. If no suitable transceiver is found (transceiver is unset), run the following steps: 1. Create an RTCRtpSender, sender, from the media description. 2. Create an RTCRtpReceiver, receiver, from the media description. 3. Create an RTCRtpTransceiver with sender, receiver and direction, and let transceiver be the result. 3. Set transceiver's mid value to the mid of the corresponding media description. */ - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); const transceiver1 = pc1.addTransceiver('audio'); assert_array_equals(pc1.getTransceivers(), [transceiver1]); assert_array_equals(pc2.getTransceivers(), []); return pc1.createOffer() @@ -132,17 +132,17 @@ /* 4.3.1.6. Set the RTCSessionSessionDescription 9. If description is of type "rollback", then run the following steps: 1. If the mid value of an RTCRtpTransceiver was set to a non-null value by the RTCSessionDescription that is being rolled back, set the mid value of that transceiver to null, as described by [JSEP] (section 4.1.8.2.). */ - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); const transceiver = pc.addTransceiver('audio'); assert_equals(transceiver.mid, null); return pc.createOffer() .then(offer => { assert_equals(transceiver.mid, null); return pc.setLocalDescription(offer); @@ -152,17 +152,17 @@ return pc.setLocalDescription({ type: 'rollback' }); }) .then(() => { assert_equals(transceiver.mid, null, 'Expect transceiver.mid to become null again after rollback'); }); }, 'setLocalDescription(rollback) should unset transceiver.mid'); - promise_test(() => { + promise_test(t => { const pc = new RTCPeerConnection(); const transceiver1 = pc.addTransceiver('audio'); assert_equals(transceiver1.mid, null); return pc.createOffer() .then(offer => pc.setLocalDescription(offer) .then(() => generateAnswer(offer))) @@ -197,17 +197,17 @@ /* 4.3.1.6. Set the RTCSessionSessionDescription 9. If description is of type "rollback", then run the following steps: 2. If an RTCRtpTransceiver was created by applying the RTCSessionDescription that is being rolled back, and a track has not been attached to it via addTrack, remove that transceiver from connection's set of transceivers, as described by [JSEP] (section 4.1.8.2.). */ - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); pc1.addTransceiver('audio'); return pc1.createOffer() .then(offer => pc2.setRemoteDescription(offer)) .then(() => {
--- a/testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html +++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html @@ -20,17 +20,17 @@ interface RTCRtpSynchronizationSource { readonly attribute DOMHighResTimeStamp timestamp; readonly attribute unsigned long source; readonly attribute byte audioLevel; readonly attribute boolean? voiceActivityFlag; }; */ - promise_test(() => { + promise_test(t => { const pc1 = new RTCPeerConnection(); const pc2 = new RTCPeerConnection(); const ontrackPromise = new Promise(resolve => { pc2.addEventListener('track', trackEvent => { const { receiver } = trackEvent; assert_true(receiver instanceof RTCRtpReceiver, 'Expect trackEvent.receiver to be instance of RTCRtpReceiver');
--- a/testing/web-platform/tests/webrtc/RTCRtpSender-getStats.https.html +++ b/testing/web-platform/tests/webrtc/RTCRtpSender-getStats.https.html @@ -34,28 +34,28 @@ 8.5. The stats selection algorithm 3. If selector is an RTCRtpSender, gather stats for and add the following objects to result: - All RTCOutboundRTPStreamStats objects corresponding to selector. - All stats objects referenced directly or indirectly by the RTCOutboundRTPStreamStats objects added. */ - promise_test(async () => { + promise_test(async t => { const caller = new RTCPeerConnection(); const callee = new RTCPeerConnection(); const { sender } = caller.addTransceiver('audio'); await doSignalingHandshake(caller, callee); const statsReport = await sender.getStats(); validateStatsReport(statsReport); assert_stats_report_has_stats(statsReport, ['outbound-rtp']); }, 'sender.getStats() via addTransceiver should return stats report containing outbound-rtp stats'); - promise_test(async () => { + promise_test(async t => { const caller = new RTCPeerConnection(); const callee = new RTCPeerConnection(); const stream = await navigator.mediaDevices.getUserMedia({audio:true}); const [track] = stream.getTracks(); const sender = caller.addTrack(track, stream); await doSignalingHandshake(caller, callee); const statsReport = await sender.getStats();