1 files changed, 140 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..10cfd65155
--- /dev/null
+++ b/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-helper.js
@@ -0,0 +1,140 @@
+'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) {
+  const extmap = new RegExp('a=extmap:(\\d+)');
+  let sdpLines = sdp.split('\r\n');
+
+  // This assumes at most one audio m= section and one video m= section.
+  // If more are present, only the first section of each kind is munged.
+  for (const section of ['audio', 'video']) {
+    let found_section = false;
+    let maxId = undefined;
+    let maxIdLine = undefined;
+    let extmapAllowMixed = false;
+
+    // find the largest header extension id for section.
+    for (let i = 0; i < sdpLines.length; ++i) {
+      if (!found_section) {
+        if (sdpLines[i].startsWith('m=' + section)) {
+          found_section = true;
+        }
+        continue;
+      } else {
+        if (sdpLines[i].startsWith('m=')) {
+          // end of section
+          break;
+        }
+      }
+
+      if (sdpLines[i] === 'a=extmap-allow-mixed') {
+        extmapAllowMixed = true;
+      }
+      let result = sdpLines[i].match(extmap);
+      if (result && result.length === 2) {
+        if (maxId == undefined || result[1] > maxId) {
+          maxId = parseInt(result[1]);
+          maxIdLine = i;
+        }
+      }
+    }
+
+    if (maxId == 14 && !extmapAllowMixed) {
+      // Reaching the limit of one byte header extension. Adding two byte header
+      // extension support.
+      sdpLines.splice(maxIdLine + 1, 0, 'a=extmap-allow-mixed');
+    }
+    if (maxIdLine !== undefined) {
+      sdpLines.splice(maxIdLine + 1, 0,
+                      'a=extmap:' + (maxId + 1).toString() + ' ' + uri);
+    }
+  }
+  return sdpLines.join('\r\n');
+}
+
+// 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];
+}