From 8dd16259287f58f9273002717ec4d27e97127719 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 12 Jun 2024 07:43:14 +0200 Subject: Merging upstream version 127.0. Signed-off-by: Daniel Baumann --- ...eiver-audio-jitterBufferTarget-stats.https.html | 18 --- ...CRtpReceiver-jitterBufferTarget-stats-helper.js | 70 ----------- .../RTCRtpReceiver-jitterBufferTarget.html | 130 --------------------- ...RtpReceiver-video-jitterBufferTarget-stats.html | 18 --- 4 files changed, 236 deletions(-) delete mode 100644 testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html delete mode 100644 testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js delete mode 100644 testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget.html delete mode 100644 testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html (limited to 'testing/web-platform/tests/webrtc-extensions') diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html deleted file mode 100644 index d728ec5a9c..0000000000 --- a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html +++ /dev/null @@ -1,18 +0,0 @@ - - - -Tests RTCRtpReceiver-jitterBufferTarget verified with stats - - - - - - - diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js deleted file mode 100644 index 31d80926d3..0000000000 --- a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js +++ /dev/null @@ -1,70 +0,0 @@ -async function measureDelayFromStats(t, receiver, cycles, targetDelay, tolerance) { - let oldInboundStats; - - for (let i = 0; i < cycles; i++) { - const statsReport = await receiver.getStats(); - const inboundStats = [...statsReport.values()].find(({type}) => type == "inbound-rtp"); - - if (inboundStats) { - if (oldInboundStats) { - const emittedCount = inboundStats.jitterBufferEmittedCount - oldInboundStats.jitterBufferEmittedCount; - - if (emittedCount) { - const delay = 1000 * (inboundStats.jitterBufferDelay - oldInboundStats.jitterBufferDelay) / emittedCount; - - if (Math.abs(delay - targetDelay) < tolerance) { - return true; - } - } - } - oldInboundStats = inboundStats; - } - await new Promise(r => t.step_timeout(r, 1000)); - } - - return false; -} - -async function applyJitterBufferTarget(t, kind, target) { - const caller = new RTCPeerConnection(); - t.add_cleanup(() => caller.close()); - const callee = new RTCPeerConnection(); - t.add_cleanup(() => callee.close()); - - const stream = await getNoiseStream({[kind]:true}); - t.add_cleanup(() => stream.getTracks().forEach(track => track.stop())); - caller.addTransceiver(stream.getTracks()[0], {streams: [stream]}); - - exchangeIceCandidates(caller, callee); - await exchangeOffer(caller, callee); - await exchangeAnswer(caller, callee); - - const receiver = callee.getReceivers()[0]; - - // Workaround for Chromium to pull audio from jitter buffer. - if (kind === "audio") { - const audio = document.createElement("audio"); - - audio.srcObject = new MediaStream([receiver.track]); - audio.play(); - } - assert_equals(receiver.jitterBufferTarget, null, - `jitterBufferTarget supported for ${kind}`); - - let result = await measureDelayFromStats(t, receiver, 5, 0, 100); - assert_true(result, 'jitter buffer is not stabilised'); - - receiver.jitterBufferTarget = target; - assert_equals(receiver.jitterBufferTarget, target, - `jitterBufferTarget increase target for ${kind}`); - - result = await measureDelayFromStats(t, receiver, 10, target, 20); - assert_true(result, 'jitterBuffer does not reach target'); - - receiver.jitterBufferTarget = 0; - assert_equals(receiver.jitterBufferTarget, 0, - `jitterBufferTarget decrease target for ${kind}`); - - result = await measureDelayFromStats(t, receiver, 10, 0, 100); - assert_true(result, 'jitter buffer delay is not back to normal'); -} diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget.html b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget.html deleted file mode 100644 index 448162d3a2..0000000000 --- a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget.html +++ /dev/null @@ -1,130 +0,0 @@ - - -Tests for RTCRtpReceiver-jitterBufferTarget attribute - - - - - diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html deleted file mode 100644 index 022dbe70c5..0000000000 --- a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html +++ /dev/null @@ -1,18 +0,0 @@ - - - -Tests RTCRtpReceiver-jitterBufferTarget verified with stats - - - - - - - -- cgit v1.2.3