Rewrite jitterBufferTarget test

Bug: 325960326 Change-Id: I19a157e7012f2b0b8b5a3b89a9d522683bd19fde Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5324009 Reviewed-by: Harald Alvestrand <[email protected]> Commit-Queue: Florent Castelli <[email protected]> Reviewed-by: Florent Castelli <[email protected]> Cr-Commit-Position: refs/heads/main@{#1270885}
Tobias710219 · Mar 11, 2024 · 15a6c2c · 15a6c2c
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Showing 4 changed files with 106 additions and 96 deletions.
diff --git a/webrtc-extensions/RTCRtpReceiver-audio-jitterBufferTarget-stats.https.html b/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/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats-helper.js b/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/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html b/webrtc-extensions/RTCRtpReceiver-jitterBufferTarget-stats.html
diff --git a/webrtc-extensions/RTCRtpReceiver-video-jitterBufferTarget-stats.html b/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>