Bug 1478285 [wpt PR 12179] - webrtc wpt: remove generateMediaStreamTrack, add cleanup after getNoiseStream, a=testonly

Automatic update from web-platform-testswebrtc wpt: remove generateMediaStreamTrack, add cleanup after getNoiseStream

BUG=836871

Change-Id: I278b34cb305a13a029e84eaaeec57144e2858678
Reviewed-on: https://chromium-review.googlesource.com/1149364
Reviewed-by: Henrik Boström <hbos@chromium.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#578611}

--

wpt-commits: 36c9fdd4ef36c289f416a16707074157d818ea7c
wpt-pr: 12179

testing/web-platform/meta/MANIFEST.json

@@ -642526,7 +642526,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-addTransceiver.https.html": [
-   "347603cef6e5eaed70077ebc4bbb8abcff7b78d4",
+   "c0c5d782c2fa365052d08d63eb14954a756c8bc0",
    "testharness"
   ],
   "webrtc/RTCPeerConnection-canTrickleIceCandidates.html": [
@@ -642578,7 +642578,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-helper.js": [
-   "597170f8c91ca0790b68c32def31a48e529ded6f",
+   "27ff1873790cfc4f71a07e1735ec4389ee330a18",
    "support"
   ],
   "webrtc/RTCPeerConnection-iceConnectionState.html": [
@@ -642654,7 +642654,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-setRemoteDescription-tracks.https.html": [
-   "91828f962aaf7b6076398da9a0447aef3586e46d",
+   "7179f1ef78ccd5b5b956df6241d0b2b47c7425a4",
    "testharness"
   ],
   "webrtc/RTCPeerConnection-setRemoteDescription.html": [
@@ -642802,7 +642802,7 @@
    "support"
   ],
   "webrtc/interfaces.https.html": [
-   "0eefc90c210c9c1587978fbc4ae9df7f5a4da080",
+   "a834f895abcc13d31da6a529b0c30d72e01c753c",
    "testharness"
   ],
   "webrtc/no-media-call.html": [

testing/web-platform/tests/webrtc/RTCPeerConnection-addTransceiver.https.html

@@ -3,16 +3,12 @@
 <title>RTCPeerConnection.prototype.addTransceiver</title>
 <script src="/resources/testharness.js"></script>
 <script src="/resources/testharnessreport.js"></script>
-<script src="RTCPeerConnection-helper.js"></script>
 <script>
   'use strict';
 
   // Test is based on the following editor draft:
   // https://rawgit.com/w3c/webrtc-pc/cc8d80f455b86c8041d63bceb8b457f45c72aa89/webrtc.html
 
-  // The following helper functions are called from RTCPeerConnection-helper.js:
-  //   generateMediaStreamTrack()
-
   /*
     5.1.  RTCPeerConnection Interface Extensions
 
@@ -244,11 +240,13 @@
       5.  If the first argument is a MediaStreamTrack , let it be track and let
           kind be track.kind.
    */
-  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();
     const transceiver = pc.addTransceiver(track);
     const { sender, receiver } = transceiver;
 
@@ -282,11 +280,13 @@
 
   }, 'addTransceiver(track) should have result with sender.track be given track');
 
-  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();
     const transceiver1 = pc.addTransceiver(track);
     const transceiver2 = pc.addTransceiver(track);
 

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

@@ -330,30 +330,6 @@ function assert_equals_array_buffer(buffer1, buffer2) {
   }
 }
 
-// Generate a MediaStreamTrack for testing use.
-// We generate it by creating an anonymous RTCPeerConnection,
-// call addTransceiver(), and use the remote track
-// from RTCRtpReceiver. This track is supposed to
-// receive media from a remote peer and be played locally.
-// We use this approach instead of getUserMedia()
-// to bypass the permission dialog and fake media devices,
-// as well as being able to generate many unique tracks.
-function generateMediaStreamTrack(kind) {
-  const pc = new RTCPeerConnection();
-
-  assert_idl_attribute(pc, 'addTransceiver',
-    'Expect pc to have addTransceiver() method');
-
-  const transceiver = pc.addTransceiver(kind);
-  const { receiver } = transceiver;
-  const { track } = receiver;
-
-  assert_true(track instanceof MediaStreamTrack,
-    'Expect receiver track to be instance of MediaStreamTrack');
-
-  return track;
-}
-
 // These media tracks will be continually updated with deterministic "noise" in
 // order to ensure UAs do not cease transmission in response to apparent
 // silence.

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

@@ -85,6 +85,8 @@
       getNoiseStream({audio: true}),
       getNoiseStream({audio: true}),
     ]);
+    t.add_cleanup(() => localStreams.forEach((stream) =>
+      stream.getTracks().forEach((track) => track.stop())));
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     caller.addTrack(localStreams[1].getTracks()[0], localStreams[0]);
     let ontrackEventsFired = 0;
@@ -129,6 +131,8 @@
       getNoiseStream({audio: true}),
       getNoiseStream({audio: true}),
     ]);
+    t.add_cleanup(() => localStreams.forEach((stream) =>
+      stream.getTracks().forEach((track) => track.stop())));
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     const remoteStreams = [];
     callee.ontrack = e => {
@@ -161,6 +165,8 @@
       getNoiseStream({audio: true}),
       getNoiseStream({audio: true}),
     ]);
+    t.add_cleanup(() => localStreams.forEach((stream) =>
+      stream.getTracks().forEach((track) => track.stop())));
     caller.addTrack(localStreams[0].getTracks()[0], localStreams[0]);
     const remoteStreams = [];
     callee.ontrack = e => {
@@ -190,6 +196,8 @@
       getNoiseStream({audio: true}),
       getNoiseStream({audio: true}),
     ]);
+    t.add_cleanup(() => localStreams.forEach((stream) =>
+      stream.getTracks().forEach((track) => track.stop())));
     caller.addTrack(localStreams[0].getTracks()[0],
                     localStreams[0], localStreams[1]);
     const ontrackPromise = addEventListenerPromise(t, callee, 'track', e => {

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

@@ -11,7 +11,7 @@
 
   // The following helper functions are called from RTCPeerConnection-helper.js:
   //   generateAnswer()
-  //   generateMediaStreamTrack()
+  //   getNoiseStream()
 
   // Put the global IDL test objects under a parent object.
   // This allows easier search for the test cases when
@@ -155,10 +155,7 @@
 
       'RTCIceTransport': ['idlTestObjects.iceTransport'],
 
-      // Test on both MediaStreamTrack from getUserMedia and transceiver
-      'MediaStreamTrack': [
-        `idlTestObjects.mediaStreamTrack`,
-        `generateMediaStreamTrack('audio')`]
+      'MediaStreamTrack': ['idlTestObjects.mediaStreamTrack']
     });
 
     idlArray.test();