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Bug 1531094: Fix a couple of test-cases that want to see rtp stats before RTP has started. Update meta file to reflect what is still missing. r=jib
Depends on D21557 Differential Revision: https://phabricator.services.mozilla.com/D21986 --HG-- extra : moz-landing-system : lando
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@ -25,11 +25,11 @@
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[getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats]
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expected: FAIL
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720, https://bugzilla.mozilla.org/show_bug.cgi?id=1531094
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225720
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[getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats]
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expected: FAIL
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1531094
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bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1225723
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[getStats() with connected peer connections having tracks and data channel should return all mandatory to implement stats]
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expected: FAIL
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@ -173,19 +173,18 @@
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- All stats objects referenced directly or indirectly by the RTCOutboundRTPStreamStats
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objects added.
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*/
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promise_test(t => {
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const pc = new RTCPeerConnection();
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t.add_cleanup(() => pc.close());
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return getTrackFromUserMedia('audio')
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.then(([track, mediaStream]) => {
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pc.addTrack(track, mediaStream);
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promise_test(async t => {
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const pc = createPeerConnectionWithCleanup(t);
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const pc2 = createPeerConnectionWithCleanup(t);
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return pc.getStats(track)
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.then(statsReport => {
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validateStatsReport(statsReport);
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assert_stats_report_has_stats(statsReport, ['outbound-rtp']);
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});
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});
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let [track, mediaStream] = await getTrackFromUserMedia('audio');
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pc.addTrack(track, mediaStream);
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exchangeIceCandidates(pc, pc2);
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await doSignalingHandshake(pc, pc2);
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await listenToIceConnected(pc);
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const stats = await pc.getStats(track);
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validateStatsReport(stats);
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assert_stats_report_has_stats(stats, ['outbound-rtp']);
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}, `getStats() on track associated with RtpSender should return stats report containing outbound-rtp stats`);
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@ -197,16 +196,20 @@
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- All stats objects referenced directly or indirectly by the RTCInboundRTPStreamStats
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added.
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*/
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promise_test(t => {
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const pc = new RTCPeerConnection();
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t.add_cleanup(() => pc.close());
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const transceiver = pc.addTransceiver('audio');
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promise_test(async t => {
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const pc = createPeerConnectionWithCleanup(t);
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const pc2 = createPeerConnectionWithCleanup(t);
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return pc.getStats(transceiver.receiver.track)
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.then(statsReport => {
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validateStatsReport(statsReport);
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assert_stats_report_has_stats(statsReport, ['inbound-rtp']);
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let [track, mediaStream] = await getTrackFromUserMedia('audio');
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pc.addTrack(track, mediaStream);
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exchangeIceCandidates(pc, pc2);
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await doSignalingHandshake(pc, pc2);
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await new Promise(resolve => {
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pc2.getReceivers()[0].track.addEventListener('unmute', resolve);
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});
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const stats = await pc2.getStats(track);
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validateStatsReport(stats);
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assert_stats_report_has_stats(stats, ['inbound-rtp']);
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}, `getStats() on track associated with RtpReceiver should return stats report containing inbound-rtp stats`);
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/*
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