Add wpt to use stats to measure jitterBufferTarget.

Differential Revision: https://phabricator.services.mozilla.com/D178484 bugzilla-url: https://bugzilla.mozilla.org/show_bug.cgi?id=1592988 gecko-commit: 38320c197609eae03222a0bf0678b364576c8c9f gecko-reviewers: jib
web-platform-tests · May 23, 2023 · a306eeb · a306eeb
1 parent 01f72bd
commit a306eeb
Showing 1 changed file with 91 additions and 0 deletions.
diff --git a/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html b/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html
@@ -0,0 +1,91 @@
+<!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", 250),
+                    "measure raising video jitterBufferTarget to 250");
+async_promise_test(t => applyJitterBufferTarget(t, "audio", 250),
+                    "measure raising audio jitterBufferTarget to 250");
+async_promise_test(t => applyJitterBufferTarget(t, "video", 500),
+                    "measure raising video jitterBufferTarget to 500");
+async_promise_test(t => applyJitterBufferTarget(t, "audio", 500),
+                    "measure raising audio jitterBufferTarget to 500");
+async_promise_test(t => applyJitterBufferTarget(t, "video", 250, 50),
+                    "measure lowering video jitterBufferTarget to 50");
+async_promise_test(t => applyJitterBufferTarget(t, "audio", 250, 50),
+                    "measure lowering audio jitterBufferTarget to 50");
+async_promise_test(t => applyJitterBufferTarget(t, "video", 500, 150),
+                    "measure lowering video jitterBufferTarget to 150");
+async_promise_test(t => applyJitterBufferTarget(t, "audio", 500, 150),
+                    "measure lowering audio jitterBufferTarget to 150");
+
+async function applyJitterBufferTarget(t, kind, target, targetToLower) {
+  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()));
+  for (const track of stream.getTracks()) {
+    caller.addTrack(track, stream);
+  }
+  exchangeIceCandidates(caller, callee);
+  await exchangeOfferAnswer(caller, callee);
+  const receiver = callee.getReceivers().find(({track}) => track.kind == kind);
+  t.step(() => assert_equals(receiver.jitterBufferTarget, null,
+   `jitterBufferTarget supported for ${kind}`));
+  receiver.jitterBufferTarget = target;
+  t.step(() => assert_equals(receiver.jitterBufferTarget, target,
+    `jitterBufferTarget increase target for ${kind}`));
+  const delay = await measureDelayFromStats(t, receiver, callee, target, kind);
+
+  if (targetToLower) {
+    t.step(() => assert_less_than(targetToLower, delay, "targetToLower represents a decrease"));
+    receiver.jitterBufferTarget = targetToLower;
+    t.step(() => assert_equals(receiver.jitterBufferTarget, targetToLower,
+      `jitterBufferTarget decrease target for ${kind}`));
+    await measureDelayFromStats(t, receiver, callee, targetToLower, kind);
+  }
+}
+
+async function measureDelayFromStats(t, receiver, callee, target, kind) {
+  const lowerBound = target * 0.7;
+  const upperBound = target * 1.3;
+  let delay, oldInboundStats;
+
+  for (let statChecks = 0; statChecks < 40; statChecks++) {
+    await new Promise(r => t.step_timeout(r, 1000));
+    const statsReport = await callee.getStats();
+    const inboundStats = [...statsReport.values()].find(({type}) => type == "inbound-rtp");
+
+    if (oldInboundStats) {
+      delay = ((inboundStats.jitterBufferDelay - oldInboundStats.jitterBufferDelay) /
+               (inboundStats.jitterBufferEmittedCount - oldInboundStats.jitterBufferEmittedCount) * 1000);
+      if (delay > lowerBound && delay < upperBound) {
+        break;
+      }
+    }
+    oldInboundStats = inboundStats;
+  }
+  t.step(() => assert_between_inclusive(delay , lowerBound, upperBound,
+   `${kind} delay is within bounds`));
+  return delay;
+}
+</script>
+</body>