Add WPT.
Differential Revision: https://phabricator.services.mozilla.com/D137041
bugzilla-url: https://bugzilla.mozilla.org/show_bug.cgi?id=1751532
gecko-commit: 0e7a4bf4ff326a1fa29ba95200eb55366c1b69a1
gecko-reviewers: jib
diff --git a/webrtc-stats/rtp-stats-creation.html b/webrtc-stats/rtp-stats-creation.html
new file mode 100644
index 0000000..3141bc0
--- /dev/null
+++ b/webrtc-stats/rtp-stats-creation.html
@@ -0,0 +1,85 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>No RTCRtpStreamStats should exist prior to RTP/RTCP packet flow</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script src="../webrtc/RTCPeerConnection-helper.js"></script>
+<script>
+'use strict';
+
+promise_test(async (test) => {
+ const localPc = createPeerConnectionWithCleanup(test);
+ const remotePc = createPeerConnectionWithCleanup(test);
+
+ localPc.addTransceiver("audio");
+ localPc.addTransceiver("video");
+ await exchangeOfferAndListenToOntrack(test, localPc, remotePc);
+ const report = await remotePc.getStats();
+ const rtp = [...report.values()].filter(({type}) => type.endsWith("rtp"));
+ assert_equals(rtp.length, 0, "no rtp stats with only remote description");
+}, "No RTCRtpStreamStats exist when only remote description is set");
+
+promise_test(async (test) => {
+ const localPc = createPeerConnectionWithCleanup(test);
+ const remotePc = createPeerConnectionWithCleanup(test);
+
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio"));
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video"));
+ await exchangeOfferAndListenToOntrack(test, localPc, remotePc);
+ const report = await localPc.getStats();
+ const rtp = [...report.values()].filter(({type}) => type.endsWith("rtp"));
+ assert_equals(rtp.length, 0, "no rtp stats with only local description");
+}, "No RTCRtpStreamStats exist when only local description is set");
+
+promise_test(async (test) => {
+ const localPc = createPeerConnectionWithCleanup(test);
+ const remotePc = createPeerConnectionWithCleanup(test);
+
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio"));
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video"));
+ exchangeIceCandidates(localPc, remotePc);
+ await exchangeOfferAnswer(localPc, remotePc);
+ const start = performance.now();
+ while (true) {
+ const report = await localPc.getStats();
+ const outbound =
+ [...report.values()].filter(({type}) => type == "outbound-rtp");
+ assert_true(outbound.every(({packetsSent}) => packetsSent > 0),
+ "no outbound rtp stats before packets sent");
+ if (outbound.length == 2) {
+ // One outbound stat for each track is present. We're done.
+ break;
+ }
+ if (performance.now() > start + 5000) {
+ assert_unreached("outbound stats should become available");
+ }
+ await new Promise(r => test.step_timeout(r, 100));
+ }
+}, "No RTCOutboundRtpStreamStats exist until packets have been sent");
+
+promise_test(async (test) => {
+ const localPc = createPeerConnectionWithCleanup(test);
+ const remotePc = createPeerConnectionWithCleanup(test);
+
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "audio"));
+ localPc.addTrack(...await createTrackAndStreamWithCleanup(test, "video"));
+ exchangeIceCandidates(localPc, remotePc);
+ await exchangeOfferAnswer(localPc, remotePc);
+ const start = performance.now();
+ while (true) {
+ const report = await remotePc.getStats();
+ const inbound =
+ [...report.values()].filter(({type}) => type == "inbound-rtp");
+ assert_true(inbound.every(({packetsReceived}) => packetsReceived > 0),
+ "no inbound rtp stats before packets received");
+ if (inbound.length == 2) {
+ // One inbound stat for each track is present. We're done.
+ break;
+ }
+ if (performance.now() > start + 5000) {
+ assert_unreached("inbound stats should become available");
+ }
+ await new Promise(r => test.step_timeout(r, 100));
+ }
+}, "No RTCInboundRtpStreamStats exist until packets have been received");
+</script>
