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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
| commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
| tree | 871fc667d2de662f171103ce5ec067014ef85e61 /testing/web-platform/tests/webrtc-extensions | |
| parent | Adding upstream version 124.0.1. (diff) | |
| download | firefox-upstream/125.0.1.tar.xz firefox-upstream/125.0.1.zip | |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-extensions')
4 files changed, 106 insertions, 96 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 new file mode 100644 index 0000000000..d728ec5a9c --- /dev/null +++ b/testing/web-platform/tests/webrtc-extensions/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-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js new file mode 100644 index 0000000000..31d80926d3 --- /dev/null +++ b/testing/web-platform/tests/webrtc-extensions/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-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html b/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html deleted file mode 100644 index 33f71800bd..0000000000 --- a/testing/web-platform/tests/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html +++ /dev/null @@ -1,96 +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> -<body> -<script> -'use strict' - -function async_promise_test(func, name, properties) { - async_test(t => { - Promise.resolve(func(t)) - .catch(t.step_func(e => { throw e; })) - .then(() => t.done()); - }, name, properties); -} - -async_promise_test(t => applyJitterBufferTarget(t, "video", 4000), - "measure raising and lowering video jitterBufferTarget"); -async_promise_test(t => applyJitterBufferTarget(t, "audio", 4000), - "measure raising and lowering audio jitterBufferTarget"); - -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]}); - caller.addTransceiver(stream.getTracks()[0], {streams: [stream]}); - - exchangeIceCandidates(caller, callee); - await exchangeOffer(caller, callee); - const [unconstrainedReceiver, constrainedReceiver] = callee.getReceivers(); - const haveRtp = Promise.all([ - new Promise(r => constrainedReceiver.track.onunmute = r), - new Promise(r => unconstrainedReceiver.track.onunmute = r) - ]); - await exchangeAnswer(caller, callee); - const chromeTimeout = new Promise(r => t.step_timeout(r, 1000)); // crbug.com/1295295 - await Promise.race([haveRtp, chromeTimeout]); - - // Allow some data to be processed to let the jitter buffer to stabilize a bit before measuring - await new Promise(r => t.step_timeout(r, 5000)); - - t.step(() => assert_equals(constrainedReceiver.jitterBufferTarget, null, - `jitterBufferTarget supported for ${kind}`)); - - constrainedReceiver.jitterBufferTarget = target; - t.step(() => assert_equals(constrainedReceiver.jitterBufferTarget, target, - `jitterBufferTarget increase target for ${kind}`)); - - const [increased, base] = await Promise.all([ - measureDelayFromStats(t, constrainedReceiver, 20), - measureDelayFromStats(t, unconstrainedReceiver, 20) - ]); - - t.step(() => assert_greater_than(increased , base, - `${kind} increased delay ${increased} ` + - ` greater than base delay ${base}`)); - - constrainedReceiver.jitterBufferTarget = 0; - - // Allow the jitter buffer to stabilize a bit before measuring - await new Promise(r => t.step_timeout(r, 5000)); - t.step(() => assert_equals(constrainedReceiver.jitterBufferTarget, 0, - `jitterBufferTarget decrease target for ${kind}`)); - - const decreased = await measureDelayFromStats(t, constrainedReceiver, 20); - - t.step(() => assert_less_than(decreased, increased, - `${kind} decreasedDelay ${decreased} ` + - `less than increased delay ${increased}`)); -} - -async function measureDelayFromStats(t, receiver, cycles) { - - let statsReport = await receiver.getStats(); - const oldInboundStats = [...statsReport.values()].find(({type}) => type == "inbound-rtp"); - - await new Promise(r => t.step_timeout(r, 1000 * cycles)); - - statsReport = await receiver.getStats(); - const inboundStats = [...statsReport.values()].find(({type}) => type == "inbound-rtp"); - - const delay = ((inboundStats.jitterBufferDelay - oldInboundStats.jitterBufferDelay) / - (inboundStats.jitterBufferEmittedCount - oldInboundStats.jitterBufferEmittedCount) * 1000); - - return delay; -} -</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 new file mode 100644 index 0000000000..022dbe70c5 --- /dev/null +++ b/testing/web-platform/tests/webrtc-extensions/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> |