1 files changed, 108 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-helper.js b/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-helper.js
new file mode 100644
index 0000000000..c8a3e45aae
--- /dev/null
+++ b/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-helper.js
@@ -0,0 +1,108 @@
+'use strict';
+
+// This file depends on `webrtc/RTCPeerConnection-helper.js`
+// which should be loaded from the main HTML file.
+
+var kAbsCaptureTime =
+    'http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time';
+
+function addHeaderExtensionToSdp(sdp, uri) {
+  // Find the highest used header extension id by sorting the extension ids used,
+  // eliminating duplicates and adding one. This is not quite correct
+  // but this code will go away with the header extension API.
+  const usedIds = sdp.split('\n')
+    .filter(line => line.startsWith('a=extmap:'))
+    .map(line => parseInt(line.split(' ')[0].substring(9), 10))
+    .sort((a, b) => a - b)
+    .filter((item, index, array) => array.indexOf(item) === index);
+  const nextId = usedIds[usedIds.length - 1] + 1;
+  const extmapLine = 'a=extmap:' + nextId + ' ' + uri + '\r\n';
+
+  const sections = sdp.split('\nm=').map((part, index) => {
+    return (index > 0 ? 'm=' + part : part).trim() + '\r\n';
+  });
+  const sessionPart = sections.shift();
+  return sessionPart + sections.map(mediaSection => mediaSection + extmapLine).join('');
+}
+
+// TODO(crbug.com/1051821): Use RTP header extension API instead of munging
+// when the RTP header extension API is implemented.
+async function addAbsCaptureTimeAndExchangeOffer(caller, callee) {
+  let offer = await caller.createOffer();
+
+  // Absolute capture time header extension may not be offered by default,
+  // in such case, munge the SDP.
+  offer.sdp = addHeaderExtensionToSdp(offer.sdp, kAbsCaptureTime);
+
+  await caller.setLocalDescription(offer);
+  return callee.setRemoteDescription(offer);
+}
+
+// TODO(crbug.com/1051821): Use RTP header extension API instead of munging
+// when the RTP header extension API is implemented.
+async function checkAbsCaptureTimeAndExchangeAnswer(caller, callee,
+                                                    absCaptureTimeAnswered) {
+  let answer = await callee.createAnswer();
+
+  const extmap = new RegExp('a=extmap:\\d+ ' + kAbsCaptureTime + '\r\n', 'g');
+  if (answer.sdp.match(extmap) == null) {
+    // We expect that absolute capture time RTP header extension is answered.
+    // But if not, there is no need to proceed with the test.
+    assert_false(absCaptureTimeAnswered, 'Absolute capture time RTP ' +
+        'header extension is not answered');
+  } else {
+    if (!absCaptureTimeAnswered) {
+      // We expect that absolute capture time RTP header extension is not
+      // answered, but it is, then we munge the answer to remove it.
+      answer.sdp = answer.sdp.replace(extmap, '');
+    }
+  }
+
+  await callee.setLocalDescription(answer);
+  return caller.setRemoteDescription(answer);
+}
+
+async function exchangeOfferAndListenToOntrack(t, caller, callee,
+                                               absCaptureTimeOffered) {
+  const ontrackPromise = addEventListenerPromise(t, callee, 'track');
+  // Absolute capture time header extension is expected not offered by default,
+  // and thus munging is needed to enable it.
+  await absCaptureTimeOffered
+      ? addAbsCaptureTimeAndExchangeOffer(caller, callee)
+      : exchangeOffer(caller, callee);
+  return ontrackPromise;
+}
+
+async function initiateSingleTrackCall(t, cap, absCaptureTimeOffered,
+                                       absCaptureTimeAnswered) {
+  const caller = new RTCPeerConnection();
+  t.add_cleanup(() => caller.close());
+  const callee = new RTCPeerConnection();
+  t.add_cleanup(() => callee.close());
+
+  const stream = await getNoiseStream(cap);
+  stream.getTracks().forEach(track => {
+    caller.addTrack(track, stream);
+    t.add_cleanup(() => track.stop());
+  });
+
+  // TODO(crbug.com/988432): `getSynchronizationSources() on the audio side
+  // needs a hardware sink for the returned dictionary entries to get updated.
+  const remoteVideo = document.getElementById('remote');
+
+  callee.ontrack = e => {
+    remoteVideo.srcObject = e.streams[0];
+  }
+
+  exchangeIceCandidates(caller, callee);
+
+  await exchangeOfferAndListenToOntrack(t, caller, callee,
+                                        absCaptureTimeOffered);
+
+  // Exchange answer and check whether the absolute capture time RTP header
+  // extension is answered.
+  await checkAbsCaptureTimeAndExchangeAnswer(caller, callee,
+                                             absCaptureTimeAnswered);
+
+  return [caller, callee];
+}