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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
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')
-rw-r--r--testing/web-platform/tests/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html18
-rw-r--r--testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget-stats-helper.js70
-rw-r--r--testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget.html130
-rw-r--r--testing/web-platform/tests/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html18
-rw-r--r--testing/web-platform/tests/webrtc/WEB_FEATURES.yml4
-rw-r--r--testing/web-platform/tests/webrtc/back-forward-cache-with-open-webrtc-connection.https.window.js1
6 files changed, 241 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html b/testing/web-platform/tests/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html
new file mode 100644
index 0000000000..d728ec5a9c
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html
@@ -0,0 +1,18 @@
+<!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/RTCRtpReceiver-jitterBufferTarget-stats-helper.js b/testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget-stats-helper.js
new file mode 100644
index 0000000000..31d80926d3
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget-stats-helper.js
@@ -0,0 +1,70 @@
+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/RTCRtpReceiver-jitterBufferTarget.html b/testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget.html
new file mode 100644
index 0000000000..448162d3a2
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-jitterBufferTarget.html
@@ -0,0 +1,130 @@
+<!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/RTCRtpReceiver-video-jitterBufferTarget-stats.html b/testing/web-platform/tests/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html
new file mode 100644
index 0000000000..022dbe70c5
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/RTCRtpReceiver-video-jitterBufferTarget-stats.html
@@ -0,0 +1,18 @@
+<!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>
diff --git a/testing/web-platform/tests/webrtc/WEB_FEATURES.yml b/testing/web-platform/tests/webrtc/WEB_FEATURES.yml
new file mode 100644
index 0000000000..117b04f81f
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/WEB_FEATURES.yml
@@ -0,0 +1,4 @@
+features:
+- name: webrtc-sctp
+ files:
+ - RTCSctpTransport-*
diff --git a/testing/web-platform/tests/webrtc/back-forward-cache-with-open-webrtc-connection.https.window.js b/testing/web-platform/tests/webrtc/back-forward-cache-with-open-webrtc-connection.https.window.js
index fe41a9cfd5..de797b3f2c 100644
--- a/testing/web-platform/tests/webrtc/back-forward-cache-with-open-webrtc-connection.https.window.js
+++ b/testing/web-platform/tests/webrtc/back-forward-cache-with-open-webrtc-connection.https.window.js
@@ -4,6 +4,7 @@
// META: script=/html/browsers/browsing-the-web/back-forward-cache/resources/rc-helper.js
// META: script=/html/browsers/browsing-the-web/remote-context-helper/resources/remote-context-helper.js
// META: script=resources/webrtc-test-helpers.sub.js
+// META: timeout=long
'use strict';
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 12:49:58 +0000