Update web-platform-tests to revision 60220357131c65146444da1f54624d5b54d0975d

tests/wpt/web-platform-tests/webrtc/META.yml

@@ -1,3 +1,4 @@
+spec: https://w3c.github.io/webrtc-pc/
 suggested_reviewers:
   - snuggs
   - agouaillard

tests/wpt/web-platform-tests/webrtc/RTCDTMFSender-insertDTMF.https.html

@@ -119,7 +119,9 @@
     const offer = await caller.createOffer();
     await caller.setLocalDescription(offer);
     await callee.setRemoteDescription(offer);
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     callee.addTrack(track);
     const answer = await callee.createAnswer();
     await callee.setLocalDescription(answer);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-add-track-no-deadlock.https.html

@@ -3,6 +3,7 @@
 <title>RTCPeerConnection addTrack does not deadlock</title>
 <script src="/resources/testharness.js"></script>
 <script src="/resources/testharnessreport.js"></script>
+<script src="RTCPeerConnection-helper.js"></script>
 <script>
   'use strict';
 
@@ -12,7 +13,7 @@
   promise_test(async t => {
     const pc1 = new RTCPeerConnection();
     t.add_cleanup(() => pc1.close());
-    const stream = await navigator.mediaDevices.getUserMedia(
+    const stream = await getNoiseStream(
       {audio: false, video: true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const videoTrack = stream.getVideoTracks()[0];

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-addTrack.https.html

@@ -11,7 +11,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
-  // generateMediaStreamTrack
+  //   getNoiseStream()
 
   /*
     5.1.  RTCPeerConnection Interface Extensions
@@ -41,7 +41,7 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
 
@@ -64,7 +64,7 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
 
@@ -96,7 +96,7 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
 
@@ -113,7 +113,7 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
 
@@ -137,7 +137,7 @@
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
 
@@ -171,7 +171,9 @@
     assert_equals(transceiver.sender.track, null);
     assert_equals(transceiver.direction, 'recvonly');
 
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const sender = pc.addTrack(track);
 
     assert_equals(sender, transceiver.sender);
@@ -188,7 +190,9 @@
     assert_equals(transceiver.sender.track, null);
     assert_equals(transceiver.direction, 'sendrecv');
 
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const sender = pc.addTrack(track);
 
     assert_equals(sender.track, track);
@@ -201,8 +205,10 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
 
-    const transceiver =
-        caller.addTransceiver(generateMediaStreamTrack('audio'));
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
+    const transceiver = caller.addTransceiver(track);
     {
       const offer = await caller.createOffer();
       await caller.setLocalDescription(offer);
@@ -227,7 +233,7 @@
 
     // |transceiver.sender| is currently not used for sending, but it should not
     // be reused because it has been used for sending before.
-    const sender = caller.addTrack(generateMediaStreamTrack('audio'));
+    const sender = caller.addTrack(track);
     assert_true(sender != null);
     assert_not_equals(sender, transceiver.sender);
   }, 'addTrack with existing sender that has been used to send should create new sender');
@@ -240,7 +246,9 @@
     assert_equals(transceiver.sender.track, null);
     assert_equals(transceiver.direction, 'recvonly');
 
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const sender = pc.addTrack(track);
 
     assert_equals(sender.track, track);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-createOffer.html

@@ -15,7 +15,7 @@
   //   generateAnswer()
   //   countAudioLine()
   //   countVideoLine()
-  //   assert_session_desc_equals()
+  //   assert_session_desc_similar()
 
   /*
    *  4.3.2.  createOffer()
@@ -54,8 +54,8 @@
       pc.setLocalDescription(offer)
       .then(() => {
         assert_equals(pc.signalingState, 'have-local-offer');
-        assert_session_desc_equals(pc.localDescription, offer);
-        assert_session_desc_equals(pc.pendingLocalDescription, offer);
+        assert_session_desc_similar(pc.localDescription, offer);
+        assert_session_desc_similar(pc.pendingLocalDescription, offer);
         assert_equals(pc.currentLocalDescription, null);
 
         assert_array_equals(states, ['have-local-offer']);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-getStats.https.html

@@ -78,10 +78,13 @@
     });
   }, 'getStats() with track added via addTrack should succeed');
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
-    const track = generateMediaStreamTrack('audio');
+
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     pc.addTransceiver(track);
 
     return pc.getStats(track);
@@ -238,7 +241,7 @@
 
     const dataChannel = pc1.createDataChannel('test-channel');
 
-    return navigator.mediaDevices.getUserMedia({
+    return getNoiseStream({
       audio: true,
       video: true
     })

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-helper.js

@@ -93,12 +93,12 @@ function isSimilarSessionDescription(sessionDesc1, sessionDesc2) {
   }
 }
 
-function assert_session_desc_equals(sessionDesc1, sessionDesc2) {
+function assert_session_desc_similar(sessionDesc1, sessionDesc2) {
   assert_true(isSimilarSessionDescription(sessionDesc1, sessionDesc2),
     'Expect both session descriptions to have the same count of media lines');
 }
 
-function assert_session_desc_not_equals(sessionDesc1, sessionDesc2) {
+function assert_session_desc_not_similar(sessionDesc1, sessionDesc2) {
   assert_false(isSimilarSessionDescription(sessionDesc1, sessionDesc2),
     'Expect both session descriptions to have different count of media lines');
 }
@@ -350,12 +350,110 @@ function generateMediaStreamTrack(kind) {
   return track;
 }
 
-// Obtain a MediaStreamTrack of kind using getUserMedia.
+// These media tracks will be continually updated with deterministic "noise" in
+// order to ensure UAs do not cease transmission in response to apparent
+// silence.
+//
+// > Many codecs and systems are capable of detecting "silence" and changing
+// > their behavior in this case by doing things such as not transmitting any
+// > media.
+//
+// Source: https://w3c.github.io/webrtc-pc/#offer-answer-options
+const trackFactories = {
+  // Share a single context between tests to avoid exceeding resource limits
+  // without requiring explicit destruction.
+  audioContext: null,
+
+  /**
+   * Given a set of requested media types, determine if the user agent is
+   * capable of procedurally generating a suitable media stream.
+   *
+   * @param {object} requested
+   * @param {boolean} [requested.audio] - flag indicating whether the desired
+   *                                      stream should include an audio track
+   * @param {boolean} [requested.video] - flag indicating whether the desired
+   *                                      stream should include a video track
+   *
+   * @returns {boolean}
+   */
+  canCreate(requested) {
+    const supported = {
+      audio: !!window.MediaStreamAudioDestinationNode,
+      video: !!HTMLCanvasElement.prototype.captureStream
+    };
+
+    return (!requested.audio || supported.audio) &&
+      (!requested.video || supported.video);
+  },
+
+  audio() {
+    const ctx = trackFactories.audioContext = trackFactories.audioContext ||
+      new AudioContext();
+    const oscillator = ctx.createOscillator();
+    const dst = oscillator.connect(ctx.createMediaStreamDestination());
+    oscillator.start();
+    return dst.stream.getAudioTracks()[0];
+  },
+
+  video({width = 640, height = 480} = {}) {
+    const canvas = Object.assign(
+      document.createElement("canvas"), {width, height}
+    );
+    const ctx = canvas.getContext('2d');
+    const stream = canvas.captureStream();
+
+    let count = 0;
+    setInterval(() => {
+      ctx.fillStyle = `rgb(${count%255}, ${count*count%255}, ${count%255})`;
+      count += 1;
+
+      ctx.fillRect(0, 0, width, height);
+    }, 100);
+
+    if (document.body) {
+      document.body.appendChild(canvas);
+    } else {
+      document.addEventListener('DOMContentLoaded', () => {
+        document.body.appendChild(canvas);
+      });
+    }
+
+    return stream.getVideoTracks()[0];
+  }
+};
+
+// Generate a MediaStream bearing the specified tracks.
+//
+// @param {object} [caps]
+// @param {boolean} [caps.audio] - flag indicating whether the generated stream
+//                                 should include an audio track
+// @param {boolean} [caps.video] - flag indicating whether the generated stream
+//                                 should include a video track
+async function getNoiseStream(caps = {}) {
+  if (!trackFactories.canCreate(caps)) {
+    return navigator.mediaDevices.getUserMedia(caps);
+  }
+  const tracks = [];
+
+  if (caps.audio) {
+    tracks.push(trackFactories.audio());
+  }
+
+  if (caps.video) {
+    tracks.push(trackFactories.video());
+  }
+
+  return new MediaStream(tracks);
+}
+
+// Obtain a MediaStreamTrack of kind using procedurally-generated streams (and
+// falling back to `getUserMedia` when the user agent cannot generate the
+// requested streams).
 // Return Promise of pair of track and associated mediaStream.
 // Assumes that there is at least one available device
 // to generate the track.
 function getTrackFromUserMedia(kind) {
-  return navigator.mediaDevices.getUserMedia({ [kind]: true })
+  return getNoiseStream({ [kind]: true })
   .then(mediaStream => {
     const [track] = mediaStream.getTracks();
     return [track, mediaStream];

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-removeTrack.https.html

@@ -12,7 +12,6 @@
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
   // generateAnswer
-  // generateMediaStreamTrack
 
   /*
     5.1.  RTCPeerConnection Interface Extensions
@@ -35,22 +34,23 @@
     5.1.  removeTrack
       3.  If connection's [[isClosed]] slot is true, throw an InvalidStateError.
    */
-  test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track);
     const { sender } = transceiver;
 
     pc.close();
     assert_throws('InvalidStateError', () => pc.removeTrack(sender));
-
   }, 'addTransceiver - Calling removeTrack when connection is closed should throw InvalidStateError');
 
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     const sender = pc.addTrack(track, stream);
@@ -59,23 +59,24 @@
     assert_throws('InvalidStateError', () => pc.removeTrack(sender));
   }, 'addTrack - Calling removeTrack when connection is closed should throw InvalidStateError');
 
-  test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track);
     const { sender } = transceiver;
 
     const pc2 = new RTCPeerConnection();
     pc2.close();
     assert_throws('InvalidStateError', () => pc2.removeTrack(sender));
-
   }, 'addTransceiver - Calling removeTrack on different connection that is closed should throw InvalidStateError');
 
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     const sender = pc.addTrack(track, stream);
@@ -89,22 +90,23 @@
     5.1.  removeTrack
       4.  If sender was not created by connection, throw an InvalidAccessError.
    */
-  test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track);
     const { sender } = transceiver;
 
     const pc2 = new RTCPeerConnection();
     assert_throws('InvalidAccessError', () => pc2.removeTrack(sender));
-
   }, 'addTransceiver - Calling removeTrack on different connection should throw InvalidAccessError');
 
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     const sender = pc.addTrack(track, stream);
@@ -117,9 +119,11 @@
     5.1.  removeTrack
       7.  Set sender.track to null.
    */
-  test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track);
     const { sender } = transceiver;
 
@@ -130,14 +134,13 @@
     pc.removeTrack(sender);
     assert_equals(sender.track, null);
     assert_equals(transceiver.direction, 'recvonly');
-
   }, 'addTransceiver - Calling removeTrack with valid sender should set sender.track to null');
 
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
 
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+    const stream = await getNoiseStream({ audio: true });
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     const sender = pc.addTrack(track, stream);
@@ -159,7 +162,9 @@
     t.add_cleanup(() => caller.close());
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = caller.addTransceiver(track);
     const { sender } = transceiver;
 
@@ -192,7 +197,9 @@
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track, { direction: 'sendonly' });
     const { sender } = transceiver;
 
@@ -224,7 +231,9 @@
     t.add_cleanup(() => caller.close());
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = caller.addTransceiver(track, { direction: 'recvonly' });
     const { sender } = transceiver;
 
@@ -256,7 +265,9 @@
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
-    const track = generateMediaStreamTrack('audio');
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
+    const [track] = stream.getTracks();
     const transceiver = pc.addTransceiver(track, { direction: 'inactive' });
     const { sender } = transceiver;
 

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setLocalDescription-answer.html

@@ -13,7 +13,7 @@
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
   //   generateAnswer
-  //   assert_session_desc_equals
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition
@@ -73,11 +73,11 @@
         pc.setLocalDescription(answer)
         .then(() => {
           assert_equals(pc.signalingState, 'stable');
-          assert_session_desc_equals(pc.localDescription, answer);
-          assert_session_desc_equals(pc.remoteDescription, offer);
+          assert_session_desc_similar(pc.localDescription, answer);
+          assert_session_desc_similar(pc.remoteDescription, offer);
 
-          assert_session_desc_equals(pc.currentLocalDescription, answer);
-          assert_session_desc_equals(pc.currentRemoteDescription, offer);
+          assert_session_desc_similar(pc.currentLocalDescription, answer);
+          assert_session_desc_similar(pc.currentRemoteDescription, offer);
 
           assert_equals(pc.pendingLocalDescription, null);
           assert_equals(pc.pendingRemoteDescription, null);
@@ -105,11 +105,11 @@
         pc.setLocalDescription({ type: 'answer' })
         .then(() => {
           assert_equals(pc.signalingState, 'stable');
-          assert_session_desc_equals(pc.localDescription, answer);
-          assert_session_desc_equals(pc.remoteDescription, offer);
+          assert_session_desc_similar(pc.localDescription, answer);
+          assert_session_desc_similar(pc.remoteDescription, offer);
 
-          assert_session_desc_equals(pc.currentLocalDescription, answer);
-          assert_session_desc_equals(pc.currentRemoteDescription, offer);
+          assert_session_desc_similar(pc.currentLocalDescription, answer);
+          assert_session_desc_similar(pc.currentRemoteDescription, offer);
 
           assert_equals(pc.pendingLocalDescription, null);
           assert_equals(pc.pendingRemoteDescription, null);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setLocalDescription-offer.html

@@ -12,8 +12,8 @@
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
-  //   assert_session_desc_not_equals
-  //   assert_session_desc_equals
+  //   assert_session_desc_not_similar
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition
@@ -67,8 +67,8 @@
       pc.setLocalDescription(offer)
       .then(() => {
         assert_equals(pc.signalingState, 'have-local-offer');
-        assert_session_desc_equals(pc.localDescription, offer);
-        assert_session_desc_equals(pc.pendingLocalDescription, offer);
+        assert_session_desc_similar(pc.localDescription, offer);
+        assert_session_desc_similar(pc.pendingLocalDescription, offer);
         assert_equals(pc.currentLocalDescription, null);
 
         assert_array_equals(states, ['have-local-offer']);
@@ -89,8 +89,8 @@
       pc.setLocalDescription({ type: 'offer' })
       .then(() => {
         assert_equals(pc.signalingState, 'have-local-offer');
-        assert_session_desc_equals(pc.localDescription, offer);
-        assert_session_desc_equals(pc.pendingLocalDescription, offer);
+        assert_session_desc_similar(pc.localDescription, offer);
+        assert_session_desc_similar(pc.pendingLocalDescription, offer);
         assert_equals(pc.currentLocalDescription, null);
       }));
   }, 'setLocalDescription with type offer and null sdp should use lastOffer generated from createOffer');
@@ -126,7 +126,7 @@
     .then(offer1 =>
       pc.createOffer({ offerToReceiveVideo: true })
       .then(offer2 => {
-        assert_session_desc_not_equals(offer1, offer2);
+        assert_session_desc_not_similar(offer1, offer2);
         return promise_rejects(t, 'InvalidModificationError',
           pc.setLocalDescription(offer1));
       }));
@@ -147,10 +147,10 @@
         .then(offer2 =>
           pc.setLocalDescription(offer2)
           .then(offer2 => {
-            assert_session_desc_not_equals(offer1, offer2);
+            assert_session_desc_not_similar(offer1, offer2);
             assert_equals(pc.signalingState, 'have-local-offer');
-            assert_session_desc_equals(pc.localDescription, offer2);
-            assert_session_desc_equals(pc.pendingLocalDescription, offer2);
+            assert_session_desc_similar(pc.localDescription, offer2);
+            assert_session_desc_similar(pc.pendingLocalDescription, offer2);
             assert_equals(pc.currentLocalDescription, null);
 
             assert_array_equals(states, ['have-local-offer']);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setLocalDescription-pranswer.html

@@ -13,7 +13,7 @@
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
   //   generateAnswer
-  //   assert_session_desc_equals
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition
@@ -97,12 +97,12 @@
         .then(() => {
           assert_equals(pc.signalingState, 'have-local-pranswer');
 
-          assert_session_desc_equals(pc.remoteDescription, offer);
-          assert_session_desc_equals(pc.pendingRemoteDescription, offer);
+          assert_session_desc_similar(pc.remoteDescription, offer);
+          assert_session_desc_similar(pc.pendingRemoteDescription, offer);
           assert_equals(pc.currentRemoteDescription, null);
 
-          assert_session_desc_equals(pc.localDescription, pranswer);
-          assert_session_desc_equals(pc.pendingLocalDescription, pranswer);
+          assert_session_desc_similar(pc.localDescription, pranswer);
+          assert_session_desc_similar(pc.pendingLocalDescription, pranswer);
           assert_equals(pc.currentLocalDescription, null);
 
           assert_equals(pc.pendingRemoteDescription, null);
@@ -152,11 +152,11 @@
         .then(() => pc.setLocalDescription(answer))
         .then(() => {
           assert_equals(pc.signalingState, 'stable');
-          assert_session_desc_equals(pc.localDescription, answer);
-          assert_session_desc_equals(pc.remoteDescription, offer);
+          assert_session_desc_similar(pc.localDescription, answer);
+          assert_session_desc_similar(pc.remoteDescription, offer);
 
-          assert_session_desc_equals(pc.currentLocalDescription, answer);
-          assert_session_desc_equals(pc.currentRemoteDescription, offer);
+          assert_session_desc_similar(pc.currentLocalDescription, answer);
+          assert_session_desc_similar(pc.currentRemoteDescription, offer);
 
           assert_equals(pc.pendingLocalDescription, null);
           assert_equals(pc.pendingRemoteDescription, null);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setLocalDescription-rollback.html

@@ -11,7 +11,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
-  //   assert_session_desc_equals
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setLocalDescription.html

@@ -12,8 +12,8 @@
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
-  //   assert_session_desc_not_equals
-  //   assert_session_desc_equals
+  //   assert_session_desc_not_similar
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition
@@ -63,10 +63,10 @@
         pc.setLocalDescription(offer2)
         .then(() => {
           assert_equals(pc.signalingState, 'have-local-offer');
-          assert_session_desc_not_equals(offer1, offer2);
-          assert_session_desc_equals(pc.localDescription, offer2);
-          assert_session_desc_equals(pc.currentLocalDescription, offer1);
-          assert_session_desc_equals(pc.pendingLocalDescription, offer2);
+          assert_session_desc_not_similar(offer1, offer2);
+          assert_session_desc_similar(pc.localDescription, offer2);
+          assert_session_desc_similar(pc.currentLocalDescription, offer1);
+          assert_session_desc_similar(pc.pendingLocalDescription, offer2);
 
           assert_array_equals(states, ['have-local-offer', 'stable', 'have-local-offer']);
         })));
@@ -89,9 +89,9 @@
         pc.setLocalDescription(offer)
         .then(() => {
           assert_equals(pc.signalingState, 'have-local-offer');
-          assert_session_desc_equals(pc.localDescription, offer);
-          assert_session_desc_equals(pc.currentLocalDescription, answer);
-          assert_session_desc_equals(pc.pendingLocalDescription, offer);
+          assert_session_desc_similar(pc.localDescription, offer);
+          assert_session_desc_similar(pc.currentLocalDescription, answer);
+          assert_session_desc_similar(pc.pendingLocalDescription, offer);
 
           assert_array_equals(states, ['have-remote-offer', 'stable', 'have-local-offer']);
         })));

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-answer.html

@@ -13,7 +13,7 @@
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer()
   //   generateAnswer()
-  //   assert_session_desc_equals()
+  //   assert_session_desc_similar()
 
   /*
     4.3.2.  Interface Definition
@@ -73,11 +73,11 @@
         .then(() => {
           assert_equals(pc.signalingState, 'stable');
 
-          assert_session_desc_equals(pc.localDescription, offer);
-          assert_session_desc_equals(pc.remoteDescription, answer);
+          assert_session_desc_similar(pc.localDescription, offer);
+          assert_session_desc_similar(pc.remoteDescription, answer);
 
-          assert_session_desc_equals(pc.currentLocalDescription, offer);
-          assert_session_desc_equals(pc.currentRemoteDescription, answer);
+          assert_session_desc_similar(pc.currentLocalDescription, offer);
+          assert_session_desc_similar(pc.currentRemoteDescription, answer);
 
           assert_equals(pc.pendingLocalDescription, null);
           assert_equals(pc.pendingRemoteDescription, null);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-offer.html

@@ -11,7 +11,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
-  //   assert_session_desc_equals()
+  //   assert_session_desc_similar()
 
   /*
     4.3.2.  Interface Definition
@@ -65,8 +65,8 @@
       return pc2.setRemoteDescription(offer)
       .then(() => {
         assert_equals(pc2.signalingState, 'have-remote-offer');
-        assert_session_desc_equals(pc2.remoteDescription, offer);
-        assert_session_desc_equals(pc2.pendingRemoteDescription, offer);
+        assert_session_desc_similar(pc2.remoteDescription, offer);
+        assert_session_desc_similar(pc2.pendingRemoteDescription, offer);
         assert_equals(pc2.currentRemoteDescription, null);
 
         assert_array_equals(states, ['have-remote-offer']);
@@ -91,8 +91,8 @@
       .then(() => pc2.setRemoteDescription(offer))
       .then(() => {
         assert_equals(pc2.signalingState, 'have-remote-offer');
-        assert_session_desc_equals(pc2.remoteDescription, offer);
-        assert_session_desc_equals(pc2.pendingRemoteDescription, offer);
+        assert_session_desc_similar(pc2.remoteDescription, offer);
+        assert_session_desc_similar(pc2.pendingRemoteDescription, offer);
         assert_equals(pc2.currentRemoteDescription, null);
 
         assert_array_equals(states, ['have-remote-offer']);
@@ -117,14 +117,14 @@
        .then(()=> {
         return pc1.createOffer({ offerToReceiveAudio: true })
         .then(offer2 => {
-          assert_session_desc_not_equals(offer1, offer2);
+          assert_session_desc_not_similar(offer1, offer2);
 
           return pc2.setRemoteDescription(offer1)
           .then(() => pc2.setRemoteDescription(offer2))
           .then(() => {
             assert_equals(pc2.signalingState, 'have-remote-offer');
-            assert_session_desc_equals(pc2.remoteDescription, offer2);
-            assert_session_desc_equals(pc2.pendingRemoteDescription, offer2);
+            assert_session_desc_similar(pc2.remoteDescription, offer2);
+            assert_session_desc_similar(pc2.pendingRemoteDescription, offer2);
             assert_equals(pc2.currentRemoteDescription, null);
 
             assert_array_equals(states, ['have-remote-offer']);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-pranswer.html

@@ -13,7 +13,7 @@
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
   //   generateAnswer
-  //   assert_session_desc_equals
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition
@@ -97,12 +97,12 @@
         .then(() => {
           assert_equals(pc.signalingState, 'have-remote-pranswer');
 
-          assert_session_desc_equals(pc.localDescription, offer);
-          assert_session_desc_equals(pc.pendingLocalDescription, offer);
+          assert_session_desc_similar(pc.localDescription, offer);
+          assert_session_desc_similar(pc.pendingLocalDescription, offer);
           assert_equals(pc.currentLocalDescription, null);
 
-          assert_session_desc_equals(pc.remoteDescription, pranswer);
-          assert_session_desc_equals(pc.pendingRemoteDescription, pranswer);
+          assert_session_desc_similar(pc.remoteDescription, pranswer);
+          assert_session_desc_similar(pc.pendingRemoteDescription, pranswer);
           assert_equals(pc.currentRemoteDescription, null);
 
           assert_array_equals(states, ['have-local-offer', 'have-remote-pranswer']);
@@ -151,12 +151,12 @@
         .then(() => {
           assert_equals(pc.signalingState, 'stable');
 
-          assert_session_desc_equals(pc.localDescription, offer);
-          assert_session_desc_equals(pc.currentLocalDescription, offer);
+          assert_session_desc_similar(pc.localDescription, offer);
+          assert_session_desc_similar(pc.currentLocalDescription, offer);
           assert_equals(pc.pendingLocalDescription, null);
 
-          assert_session_desc_equals(pc.remoteDescription, answer);
-          assert_session_desc_equals(pc.currentRemoteDescription, answer);
+          assert_session_desc_similar(pc.remoteDescription, answer);
+          assert_session_desc_similar(pc.currentRemoteDescription, answer);
           assert_equals(pc.pendingRemoteDescription, null);
 
           assert_array_equals(states, ['have-local-offer', 'have-remote-pranswer', 'stable']);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-rollback.html

@@ -12,7 +12,7 @@
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateOffer
-  //   assert_session_desc_equals
+  //   assert_session_desc_similar
 
   /*
     4.3.2.  Interface Definition

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html

@@ -23,7 +23,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     caller.addTrack(localStream.getTracks()[0]);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -41,7 +41,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -64,7 +64,7 @@
     t.add_cleanup(() => callee.close());
     let eventSequence = '';
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -82,8 +82,8 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStreams = await Promise.all([
-      navigator.mediaDevices.getUserMedia({audio: true}),
-      navigator.mediaDevices.getUserMedia({audio: true}),
+      getNoiseStream({audio: true}),
+      getNoiseStream({audio: true}),
     ]);
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     caller.addTrack(localStreams[1].getTracks()[0], localStreams[0]);
@@ -126,8 +126,8 @@
     t.add_cleanup(() => callee.close());
     let eventSequence = '';
     const localStreams = await Promise.all([
-      navigator.mediaDevices.getUserMedia({audio: true}),
-      navigator.mediaDevices.getUserMedia({audio: true}),
+      getNoiseStream({audio: true}),
+      getNoiseStream({audio: true}),
     ]);
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     const remoteStreams = [];
@@ -158,8 +158,8 @@
     t.add_cleanup(() => callee.close());
     let eventSequence = '';
     const localStreams = await Promise.all([
-      navigator.mediaDevices.getUserMedia({audio: true}),
-      navigator.mediaDevices.getUserMedia({audio: true}),
+      getNoiseStream({audio: true}),
+      getNoiseStream({audio: true}),
     ]);
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     const remoteStreams = [];
@@ -187,8 +187,8 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStreams = await Promise.all([
-      navigator.mediaDevices.getUserMedia({audio: true}),
-      navigator.mediaDevices.getUserMedia({audio: true}),
+      getNoiseStream({audio: true}),
+      getNoiseStream({audio: true}),
     ]);
     caller.addTrack(localStreams[0].getTracks()[0],
                     localStreams[0], localStreams[1]);
@@ -215,7 +215,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     caller.addTrack(localStream.getTracks()[0]);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -232,7 +232,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     const sender = caller.addTrack(localStream.getTracks()[0]);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track');
@@ -250,7 +250,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     const [track] = localStream.getTracks();
     const sender = caller.addTrack(track, localStream);
@@ -279,7 +279,7 @@
     t.add_cleanup(() => callee.close());
     let eventSequence = '';
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     const sender = caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -306,7 +306,7 @@
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     const sender = caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -332,7 +332,7 @@
     t.add_cleanup(() => callee.close());
     let eventSequence = '';
     const localStream =
-        await navigator.mediaDevices.getUserMedia({audio: true});
+        await getNoiseStream({audio: true});
     t.add_cleanup(() => localStream.getTracks().forEach(track => track.stop()));
     const sender = caller.addTrack(localStream.getTracks()[0], localStream);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {
@@ -356,7 +356,7 @@
   promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
-    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    const stream = await getNoiseStream({audio: true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const sender = pc.addTrack(stream.getTracks()[0]);
     pc.removeTrack(sender);

tests/wpt/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription.html

@@ -11,8 +11,8 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
-  //   assert_session_desc_not_equals()
-  //   assert_session_desc_equals()
+  //   assert_session_desc_not_similar()
+  //   assert_session_desc_similar()
 
   /*
     4.3.2.  Interface Definition
@@ -116,10 +116,10 @@
     await pc.setLocalDescription(await pc.createAnswer());
     const offer2 = await pc2.createOffer({ offerToReceiveVideo: true });
     await pc.setRemoteDescription(offer2);
-    assert_session_desc_not_equals(offer1, offer2);
-    assert_session_desc_equals(pc.remoteDescription, offer2);
-    assert_session_desc_equals(pc.currentRemoteDescription, offer1);
-    assert_session_desc_equals(pc.pendingRemoteDescription, offer2);
+    assert_session_desc_not_similar(offer1, offer2);
+    assert_session_desc_similar(pc.remoteDescription, offer2);
+    assert_session_desc_similar(pc.currentRemoteDescription, offer1);
+    assert_session_desc_similar(pc.pendingRemoteDescription, offer2);
   }, 'Calling setRemoteDescription() again after one round of remote-offer/local-answer should succeed');
 
   promise_test(async t => {
@@ -136,8 +136,8 @@
     await pc.setRemoteDescription(answer);
     await pc.setRemoteDescription(await pc2.createOffer());
     assert_equals(pc.remoteDescription.sdp, pc.pendingRemoteDescription.sdp);
-    assert_session_desc_equals(pc.remoteDescription, offer);
-    assert_session_desc_equals(pc.currentRemoteDescription, answer);
+    assert_session_desc_similar(pc.remoteDescription, offer);
+    assert_session_desc_similar(pc.currentRemoteDescription, answer);
   }, 'Switching role from offerer to answerer after going back to stable state should succeed');
 
   /*

tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-codecs.html

@@ -12,6 +12,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCRtpParameters-helper.js:
+  //   doOfferAnswerExchange
   //   validateSenderRtpParameters
 
   /*
@@ -80,10 +81,13 @@
           with a newly created InvalidModificationError. Note that this also applies
           to transactionId.
    */
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
+
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
+
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -99,10 +103,11 @@
       sender.setParameters(param));
   }, 'setParameters() with codec.payloadType modified should reject with InvalidModificationError');
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -118,10 +123,11 @@
       sender.setParameters(param));
   }, 'setParameters() with codec.mimeType modified should reject with InvalidModificationError');
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -137,10 +143,11 @@
       sender.setParameters(param));
   }, 'setParameters() with codec.clockRate modified should reject with InvalidModificationError');
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -156,11 +163,11 @@
       sender.setParameters(param));
   }, 'setParameters() with codec.channels modified should reject with InvalidModificationError');
 
-
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -176,11 +183,11 @@
       sender.setParameters(param));
   }, 'setParameters() with codec.sdpFmtpLine modified should reject with InvalidModificationError');
 
-
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -197,37 +204,4 @@
       sender.setParameters(param));
   }, 'setParameters() with new codecs inserted should reject with InvalidModificationError');
 
-  promise_test(t => {
-    const pc = new RTCPeerConnection();
-    t.add_cleanup(() => pc.close());
-    const { sender } = pc.addTransceiver('audio');
-    const param = sender.getParameters();
-    validateSenderRtpParameters(param);
-
-    const { codecs } = param;
-
-    // skip and pass test if there is less than 2 codecs
-    if(codecs.length >= 2) {
-      const tmp = codecs[0];
-      codecs[0] = codecs[1];
-      codecs[1] = tmp;
-    }
-
-    return sender.setParameters(param);
-  }, 'setParameters with reordered codecs should succeed');
-
-  promise_test(t => {
-    const pc = new RTCPeerConnection();
-    t.add_cleanup(() => pc.close());
-    const { sender } = pc.addTransceiver('audio');
-    const param = sender.getParameters();
-    validateSenderRtpParameters(param);
-
-    const { codecs } = param;
-
-    param.codecs = codecs.slice(1);
-
-    return sender.setParameters(param);
-  }, 'setParameters with dropped codec should succeed');
-
 </script>

tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-encodings.html

@@ -12,6 +12,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCRtpParameters-helper.js:
+  //   doOfferAnswerExchange
   //   validateSenderRtpParameters
 
   /*
@@ -114,9 +115,12 @@
           Otherwise, set it to a list containing a single RTCRtpEncodingParameters
           with active set to true.
    */
-  test(() => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
+    t.add_cleanup(() => pc.close());
     const transceiver = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
+
     const param = transceiver.sender.getParameters();
     validateSenderRtpParameters(param);
     const { encodings } = param;
@@ -125,9 +129,11 @@
     assert_equals(encoding.active, true);
   }, 'addTransceiver() with undefined sendEncodings should have default encoding parameter with active set to true');
 
-  test(() => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
+    t.add_cleanup(() => pc.close());
     const transceiver = pc.addTransceiver('audio', { sendEncodings: [] });
+    await doOfferAnswerExchange(t, pc);
 
     const param = transceiver.sender.getParameters();
     validateSenderRtpParameters(param);
@@ -151,8 +157,9 @@
     5.2.  getParameters
       - encodings is set to the value of the [[send encodings]] internal slot.
    */
-  test(() => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
+    t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio', {
       sendEncodings: [{
         dtx: 'enabled',
@@ -164,6 +171,7 @@
         rid: 'foo'
       }]
     });
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -190,10 +198,12 @@
           with a newly created InvalidModificationError. Note that this also applies
           to transactionId.
    */
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
+
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -208,10 +218,12 @@
       sender.setParameters(param));
   }, `sender.setParameters() with mismatch number of encodings should reject with InvalidModificationError`);
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
+
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
@@ -221,32 +233,11 @@
       sender.setParameters(param));
   }, `sender.setParameters() with encodings unset should reject with InvalidModificationError`);
 
-  promise_test(t => {
-    const pc = new RTCPeerConnection();
-    t.add_cleanup(() => pc.close());
-    const { sender } = pc.addTransceiver('audio');
-
-    const param = sender.getParameters();
-    validateSenderRtpParameters(param);
-    const encoding = getFirstEncoding(param);
-    const { ssrc } = encoding;
-
-    // ssrc may not be set since it is optional
-    if(ssrc === undefined) {
-      encoding.ssrc = 2;
-    } else {
-      // If it is set, increase the number by 1 to make it different from original
-      encoding.ssrc = ssrc + 1;
-    }
-
-    return promise_rejects(t, 'InvalidModificationError',
-      sender.setParameters(param));
-  }, `setParameters() with modified encoding.ssrc field should reject with InvalidModificationError`);
-
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -265,34 +256,13 @@
       sender.setParameters(param));
   }, `setParameters() with modified encoding.rtx field should reject with InvalidModificationError`);
 
-  promise_test(t => {
-    const pc = new RTCPeerConnection();
-    t.add_cleanup(() => pc.close());
-    const { sender } = pc.addTransceiver('audio');
-
-    const param = sender.getParameters();
-    validateSenderRtpParameters(param);
-    const encoding = getFirstEncoding(param);
-    const { fec } = encoding;
-
-    if(fec === undefined) {
-      encoding.fec = { ssrc: 2 };
-    } else if(fec.ssrc === undefined) {
-      fec.ssrc = 2;
-    } else {
-      fec.ssrc += 1;
-    }
-
-    return promise_rejects(t, 'InvalidModificationError',
-      sender.setParameters(param));
-  }, `setParameters() with modified encoding.fec field should reject with InvalidModificationError`);
-
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio', {
       sendEncodings: [{ rid: 'foo' }],
     });
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -311,10 +281,11 @@
           value less than 1.0, abort these steps and return a promise rejected with
           a newly created RangeError.
    */
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -325,10 +296,11 @@
       sender.setParameters(param));
   }, `setParameters() with encoding.scaleResolutionDownBy field set to less than 1.0 should reject with RangeError`);
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -346,52 +318,76 @@
   }, `setParameters() with encoding.scaleResolutionDownBy field set to greater than 1.0 should succeed`);
 
   // Helper function to test that modifying an encoding field should succeed
-  function test_modified_encoding(field, value1, value2, desc) {
-    promise_test(t => {
+  function test_modified_encoding(kind, field, value1, value2, desc) {
+    promise_test(async t => {
       const pc = new RTCPeerConnection();
       t.add_cleanup(() => pc.close());
-      const { sender } = pc.addTransceiver('audio', {
+      const { sender } = pc.addTransceiver(kind, {
         sendEncodings: [{
           [field]: value1
         }]
       });
+      await doOfferAnswerExchange(t, pc);
 
-      const param = sender.getParameters();
-      validateSenderRtpParameters(param);
-      const encoding = getFirstEncoding(param);
+      const param1 = sender.getParameters();
+      validateSenderRtpParameters(param1);
+      const encoding1 = getFirstEncoding(param1);
 
-      assert_equals(encoding[field], value1);
-      encoding[field] = value2;
+      assert_equals(encoding1[field], value1);
+      encoding1[field] = value2;
 
-      return sender.setParameters(param)
-      .then(() => {
-        const param = sender.getParameters();
-        validateSenderRtpParameters(param);
-        const encoding = getFirstEncoding(param);
-        assert_equals(encoding[field], value2);
-      });
-    }, desc);
+      await sender.setParameters(param1);
+      const param2 = sender.getParameters();
+      validateSenderRtpParameters(param2);
+      const encoding2 = getFirstEncoding(param2);
+      assert_equals(encoding2[field], value2);
+    }, desc + ' with RTCRtpTransceiverInit');
+
+    promise_test(async t => {
+      const pc = new RTCPeerConnection();
+      t.add_cleanup(() => pc.close());
+      const { sender } = pc.addTransceiver(kind);
+      await doOfferAnswerExchange(t, pc);
+
+      const initParam = sender.getParameters();
+      validateSenderRtpParameters(initParam);
+      initParam.encodings[0][field] = value1;
+      await sender.setParameters(initParam);
+
+      const param1 = sender.getParameters();
+      validateSenderRtpParameters(param1);
+      const encoding1 = getFirstEncoding(param1);
+
+      assert_equals(encoding1[field], value1);
+      encoding1[field] = value2;
+
+      await sender.setParameters(param1);
+      const param2 = sender.getParameters();
+      validateSenderRtpParameters(param2);
+      const encoding2 = getFirstEncoding(param2);
+      assert_equals(encoding2[field], value2);
+    }, desc + ' without RTCRtpTransceiverInit');
   }
 
-  test_modified_encoding('dtx', 'enabled', 'disabled',
+  test_modified_encoding('audio', 'dtx', 'enabled', 'disabled',
     'setParameters() with modified encoding.dtx should succeed');
 
-  test_modified_encoding('dtx', 'enabled', undefined,
+  test_modified_encoding('audio', 'dtx', 'enabled', undefined,
     'setParameters() with unset encoding.dtx should succeed');
 
-  test_modified_encoding('active', true, false,
+  test_modified_encoding('audio', 'active', false, true,
     'setParameters() with modified encoding.active should succeed');
 
-  test_modified_encoding('priority', 'very-low', 'high',
+  test_modified_encoding('audio', 'priority', 'very-low', 'high',
     'setParameters() with modified encoding.priority should succeed');
 
-  test_modified_encoding('ptime', 2, 4,
+  test_modified_encoding('audio', 'ptime', 2, 4,
     'setParameters() with modified encoding.ptime should succeed');
 
-  test_modified_encoding('maxBitrate', 2, 4,
+  test_modified_encoding('audio', 'maxBitrate', 10000, 20000,
     'setParameters() with modified encoding.maxBitrate should succeed');
 
-  test_modified_encoding('maxBitrate', 24, 16,
+  test_modified_encoding('video', 'maxFramerate', 24, 16,
     'setParameters() with modified encoding.maxFramerate should succeed');
 
 </script>

tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-helper.js

@@ -8,6 +8,19 @@
 // This file depends on dictionary-helper.js which should
 // be loaded from the main HTML file.
 
+// An offer/answer exchange is necessary for getParameters() to have any
+// negotiated parameters to return.
+async function doOfferAnswerExchange(t, caller) {
+  const callee = new RTCPeerConnection();
+  t.add_cleanup(() => callee.close());
+  const offer = await caller.createOffer();
+  await caller.setLocalDescription(offer);
+  await callee.setRemoteDescription(offer);
+  const answer = await callee.createAnswer();
+  await callee.setLocalDescription(answer);
+  await caller.setRemoteDescription(answer);
+}
+
 /*
   Validates the RTCRtpParameters returned from RTCRtpSender.prototype.getParameters
 

tests/wpt/web-platform-tests/webrtc/RTCRtpParameters-transactionId.html

@@ -12,6 +12,7 @@
   // https://w3c.github.io/webrtc-pc/archives/20170605/webrtc.html
 
   // The following helper functions are called from RTCRtpParameters-helper.js:
+  //   doOfferAnswerExchange
   //   validateSenderRtpParameters
 
   /*
@@ -60,9 +61,10 @@
       - transactionId is set to a new unique identifier, used to match this
         getParameters call to a setParameters call that may occur later.
    */
-  test(() => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param1 = sender.getParameters();
     const param2 = sender.getParameters();
@@ -83,10 +85,11 @@
           with a newly created InvalidModificationError. Note that this also applies
           to transactionId.
    */
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -98,10 +101,11 @@
       sender.setParameters(param));
   }, `sender.setParameters() with transaction ID different from last getParameters() should reject with InvalidModificationError`);
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
@@ -112,24 +116,26 @@
       sender.setParameters(param));
   }, `sender.setParameters() with transaction ID unset should reject with InvalidModificationError`);
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param = sender.getParameters();
     validateSenderRtpParameters(param);
 
     return sender.setParameters(param)
     .then(() =>
-      promise_rejects(t, 'InvalidModificationError',
+      promise_rejects(t, 'InvalidStateError',
         sender.setParameters(param)));
-  }, `setParameters() twice with the same parameters should reject with InvalidModificationError`);
+  }, `setParameters() twice with the same parameters should reject with InvalidStateError`);
 
-  promise_test(t => {
+  promise_test(async t => {
     const pc = new RTCPeerConnection();
     t.add_cleanup(() => pc.close());
     const { sender } = pc.addTransceiver('audio');
+    await doOfferAnswerExchange(t, pc);
 
     const param1 = sender.getParameters();
     const param2 = sender.getParameters();

tests/wpt/web-platform-tests/webrtc/RTCRtpReceiver-getStats.https.html

@@ -62,7 +62,7 @@
     t.add_cleanup(() => caller.close());
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
-    const stream = await navigator.mediaDevices.getUserMedia({audio:true});
+    const stream = await getNoiseStream({audio:true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     caller.addTrack(track, stream);

tests/wpt/web-platform-tests/webrtc/RTCRtpSender-getStats.https.html

@@ -57,7 +57,7 @@
     t.add_cleanup(() => caller.close());
     const callee = new RTCPeerConnection();
     t.add_cleanup(() => callee.close());
-    const stream = await navigator.mediaDevices.getUserMedia({audio:true});
+    const stream = await getNoiseStream({audio:true});
     t.add_cleanup(() => stream.getTracks().forEach(track => track.stop()));
     const [track] = stream.getTracks();
     const sender = caller.addTrack(track, stream);

tests/wpt/web-platform-tests/webrtc/interfaces.https.html

@@ -79,7 +79,7 @@
 
   // Asynchoronously generate MediaStreamTrack from getUserMedia
   function asyncInitMediaStreamTrack() {
-    return navigator.mediaDevices.getUserMedia({ audio: true })
+    return getNoiseStream({ audio: true })
     .then(mediaStream => {
       idlTestObjects.mediaStreamTrack = mediaStream.getTracks()[0];
     });
@@ -109,7 +109,7 @@
     const idlArray = new IdlArray();
 
     let webrtcIdl = fetch('/interfaces/webrtc-pc.idl').then(r => r.text());
-    let mediacaptureMainIdl = fetch('/interfaces/mediacapture-main.idl').then(r => r.text());
+    let mediacaptureMainIdl = fetch('/interfaces/mediacapture-streams.idl').then(r => r.text());
 
     idlArray.add_untested_idls(mediacaptureMainIdl, { only: ['MediaStreamConstraints'] });
     idlArray.add_idls(webrtcIdl);

tests/wpt/web-platform-tests/webrtc/simplecall.https.html

@@ -1,23 +1,15 @@
 <!doctype html>
-<!--
-To run this test, you must have a webcam and a microphone or use
-fake devices by specifying
-  --use-fake-device-for-media-stream --use-fake-ui-for-media-stream
-for Chrome or by setting the
-  media.navigator.streams.fake
-property to true in Firefox.
--->
-
 <html>
 <head>
   <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
   <title>RTCPeerConnection Connection Test</title>
+  <script src="RTCPeerConnection-helper.js"></script>
 </head>
 <body>
   <div id="log"></div>
   <div>
-    <video id="local-view" autoplay="autoplay"></video>
-    <video id="remote-view" autoplay="autoplay"/>
+    <video id="local-view" muted autoplay="autoplay"></video>
+    <video id="remote-view" muted autoplay="autoplay"/>
     </video>
   </div>
 
@@ -38,7 +30,7 @@ property to true in Firefox.
     test.done();
   });
 
-  function getUserMediaOkCallback(localStream) {
+  function getNoiseStreamOkCallback(localStream) {
     gFirstConnection = new RTCPeerConnection(null);
     gFirstConnection.onicecandidate = onIceCandidateToFirst;
     localStream.getTracks().forEach(function(track) {
@@ -111,8 +103,8 @@ property to true in Firefox.
 
   // This function starts the test.
   test.step(function() {
-    navigator.mediaDevices.getUserMedia({ video: true, audio: true })
-      .then(test.step_func(getUserMediaOkCallback), failed('getUserMedia'));
+    getNoiseStream({ video: true, audio: true })
+      .then(test.step_func(getNoiseStreamOkCallback), failed('getNoiseStream'));
   });
 </script>