Bug 1471691 [wpt PR 11694] - webrtc wpt: remove generateMediaStreamTrack usage, a=testonly

Automatic update from web-platform-testswebrtc wpt: remove generateMediaStreamTrack usage

removes most usage of generateMediaStreamTrack in favor of
await-ing tracks from getUserMedia and stopping them)

BUG=836871

Change-Id: Id84f6cb6b3537faf2bb83332c5d7993550246581
Reviewed-on: https://chromium-review.googlesource.com/1117069
Reviewed-by: Henrik Boström <hbos@chromium.org>
Commit-Queue: Henrik Boström <hbos@chromium.org>
Cr-Commit-Position: refs/heads/master@{#574855}

--

wpt-commits: 948babd1549e445e83260bcc715202a7cbda98ec
wpt-pr: 11694

testing/web-platform/meta/MANIFEST.json

@@ -625201,7 +625201,7 @@
    "support"
   ],
   "webrtc/RTCDTMFSender-insertDTMF.https.html": [
-   "068c96875c7b7a0a19cfdaa6d7af5b94a57ff71c",
+   "f215e71673d05719f7e293c1078b139c75a5ac9f",
    "testharness"
   ],
   "webrtc/RTCDTMFSender-ontonechange-long.https.html": [
@@ -625249,7 +625249,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-addTrack.https.html": [
-   "c924fe69d43bdef8ff1bd9e645c5cecc6b4fe34f",
+   "74096ec62288029bf7aea81008e63e43a5f40549",
    "testharness"
   ],
   "webrtc/RTCPeerConnection-addTransceiver.html": [
@@ -625297,7 +625297,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-getStats.https.html": [
-   "dd972db6f5b3ef771ff817fbeb18fb65de01710a",
+   "fbb26c647a8759d1b9da637f7167b1a805f647c4",
    "testharness"
   ],
   "webrtc/RTCPeerConnection-getTransceivers.html": [
@@ -625333,7 +625333,7 @@
    "testharness"
   ],
   "webrtc/RTCPeerConnection-removeTrack.https.html": [
-   "a5c919e396c7f8a916a9e8fe6c766f9dc263a6f8",
+   "cfe015c4ad648d0bd5009d0ccd9629fefdc3e5ae",
    "testharness"
   ],
   "webrtc/RTCPeerConnection-setDescription-transceiver.html": [

testing/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);

testing/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
@@ -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);

testing/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);

testing/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
@@ -35,15 +34,16 @@
     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 => {
@@ -59,16 +59,17 @@
     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 => {
@@ -89,15 +90,16 @@
     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 => {
@@ -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;
 
@@ -130,7 +134,6 @@
     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 => {
@@ -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;