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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-06-12 05:43:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-06-12 05:43:14 +0000
commit8dd16259287f58f9273002717ec4d27e97127719 (patch)
tree3863e62a53829a84037444beab3abd4ed9dfc7d0 /testing/web-platform/tests/webrtc-extensions
parentReleasing progress-linux version 126.0.1-1~progress7.99u1. (diff)
downloadfirefox-8dd16259287f58f9273002717ec4d27e97127719.tar.xz
firefox-8dd16259287f58f9273002717ec4d27e97127719.zip
Merging upstream version 127.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-extensions')
-rw-r--r--testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html18
-rw-r--r--testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js70
-rw-r--r--testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget.html130
-rw-r--r--testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html18
4 files changed, 0 insertions, 236 deletions
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 @@
-<!DOCTYPE html>
-<meta charset="utf-8">
-<meta name="timeout" content="long">
-<title>Tests RTCRtpReceiver-jitterBufferTarget verified with stats</title>
-<script src="/resources/testharness.js"></script>
-<script src="/resources/testharnessreport.js"></script>
-<script src="/webrtc/RTCPeerConnection-helper.js"></script>
-<script src="/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js"></script>
-<body>
-<script>
-'use strict'
-
-promise_test(async t => {
- await applyJitterBufferTarget(t, "audio", 300);
-}, `measure raising and lowering audio jitterBufferTarget`);
-
-</script>
-</body>
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 @@
-<!DOCTYPE html>
-<meta charset="utf-8">
-<title>Tests for RTCRtpReceiver-jitterBufferTarget attribute</title>
-<script src="/resources/testharness.js"></script>
-<script src="/resources/testharnessreport.js"></script>
-<body>
-<script>
-'use strict'
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
-}, 'audio jitterBufferTarget is null by default');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 500;
- assert_equals(receiver.jitterBufferTarget, 500);
-}, 'audio jitterBufferTarget accepts posititve values');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 4000;
- assert_throws_js(RangeError, () => {
- receiver.jitterBufferTarget = 4001;
- }, 'audio jitterBufferTarget doesn\'t accept values greater than 4000 milliseconds');
- assert_equals(receiver.jitterBufferTarget, 4000);
-}, 'audio jitterBufferTarget accepts values up to 4000 milliseconds');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 700;
- assert_throws_js(RangeError, () => {
- receiver.jitterBufferTarget = -500;
- }, 'audio jitterBufferTarget doesn\'t accept negative values');
- assert_equals(receiver.jitterBufferTarget, 700);
-}, 'audio jitterBufferTarget returns last valid value on throw');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 0;
- assert_equals(receiver.jitterBufferTarget, 0);
-}, 'audio jitterBufferTarget allows zero value');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('audio', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 500;
- assert_equals(receiver.jitterBufferTarget, 500);
- receiver.jitterBufferTarget = null;
- assert_equals(receiver.jitterBufferTarget, null);
-}, 'audio jitterBufferTarget allows to reset value to null');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
-}, 'video jitterBufferTarget is null by default');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 500;
- assert_equals(receiver.jitterBufferTarget, 500);
-}, 'video jitterBufferTarget accepts posititve values');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 4000;
- assert_throws_js(RangeError, () => {
- receiver.jitterBufferTarget = 4001;
- }, 'video jitterBufferTarget doesn\'t accept values greater than 4000 milliseconds');
- assert_equals(receiver.jitterBufferTarget, 4000);
-}, 'video jitterBufferTarget accepts values up to 4000 milliseconds');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 700;
- assert_throws_js(RangeError, () => {
- receiver.jitterBufferTarget = -500;
- }, 'video jitterBufferTarget doesn\'t accept negative values');
- assert_equals(receiver.jitterBufferTarget, 700);
-}, 'video jitterBufferTarget returns last valid value');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 0;
- assert_equals(receiver.jitterBufferTarget, 0);
-}, 'video jitterBufferTarget allows zero value');
-
-test(t => {
- const pc = new RTCPeerConnection();
- t.add_cleanup(() => pc.close());
- const {receiver} = pc.addTransceiver('video', {direction:'recvonly'});
- assert_equals(receiver.jitterBufferTarget, null);
- receiver.jitterBufferTarget = 500;
- assert_equals(receiver.jitterBufferTarget, 500);
- receiver.jitterBufferTarget = null;
- assert_equals(receiver.jitterBufferTarget, null);
-}, 'video jitterBufferTarget allows to reset value to null');
-</script>
-</body>
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 @@
-<!DOCTYPE html>
-<meta charset="utf-8">
-<meta name="timeout" content="long">
-<title>Tests RTCRtpReceiver-jitterBufferTarget verified with stats</title>
-<script src="/resources/testharness.js"></script>
-<script src="/resources/testharnessreport.js"></script>
-<script src="/webrtc/RTCPeerConnection-helper.js"></script>
-<script src="/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js"></script>
-<body>
-<script>
-'use strict'
-
-promise_test(async t => {
- await applyJitterBufferTarget(t, "video", 1000);
-}, `measure raising and lowering video jitterBufferTarget`);
-
-</script>
-</body>
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-21 15:03:48 +0000