Add m-esr52 at 52.6.0

author: Matt A. Tobin <mattatobin@localhost.localdomain> 2018-02-02 04:16:08 -0500
committer: Matt A. Tobin <mattatobin@localhost.localdomain> 2018-02-02 04:16:08 -0500
commit: 5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree: 10027f336435511475e392454359edea8e25895d /testing/web-platform/tests/webrtc
parent: 49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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6 files changed, 597 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/OWNERS b/testing/web-platform/tests/webrtc/OWNERS
new file mode 100644
index 000000000..1deb98d49
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/OWNERS
@@ -0,0 +1,4 @@
+@dontcallmedom
+@tidoust
+@alvestrand
+@phoglund
diff --git a/testing/web-platform/tests/webrtc/datachannel-emptystring.html b/testing/web-platform/tests/webrtc/datachannel-emptystring.html
new file mode 100644
index 000000000..6af436a1d
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/datachannel-emptystring.html
@@ -0,0 +1,108 @@
+<!doctype html>
+<!--
+This test creates a data channel between two local PeerConnection instances
+and ensures that an empty string sent by one is received by the second.
+-->
+
+<html>
+<head>
+  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+  <title>RTCPeerConnection Data Channel Empty String Test</title>
+</head>
+<body>
+  <div id="log"></div>
+  <h2>Messages exchanged</h2>
+  <div id="msg"></div>
+
+  <!-- These files are in place when executing on W3C. -->
+  <script src="/resources/testharness.js"></script>
+  <script src="/resources/testharnessreport.js"></script>
+  <script type="text/javascript">
+  var test = async_test('Can send empty strings across a WebRTC data channel.');
+
+  var gFirstConnection = null;
+  var gSecondConnection = null;
+  var sendChannel = null;
+  var receiveChannel = null;
+
+  var onReceiveChannel = function (event) {
+    receiveChannel = event.channel;
+    receiveChannel.onmessage = onReceiveMessage;
+  };
+
+
+  // When the data channel is open, send an empty string message
+  // followed by a message that contains the string "done".
+  var onSendChannelOpen = test.step_func(function (event) {
+    var msgEl = document.getElementById('msg');
+    sendChannel.send('');
+    msgEl.innerHTML += 'Sent: [empty string]<br>';
+    sendChannel.send('done');
+    msgEl.innerHTML += 'Sent: "done"<br>';
+  });
+
+  // Check the messages received on the other side.
+  // There should be an empty string message followed by a message that
+  // contains the string "done".
+  // Pass/Fail the test according to the messages received
+  var emptyMessageReceived = false;
+  var onReceiveMessage = test.step_func(function (event) {
+    var msgEl = document.getElementById('msg');
+    msgEl.innerHTML += 'Received: ' +
+        (event.data ? '"' + event.data + '"' : '[empty string]') + '<br>';
+    if (emptyMessageReceived) {
+      assert_equals(event.data, 'done', 'The "done" message was not received');
+      test.done();
+    }
+    else {
+      assert_equals(event.data, '', 'Empty message not received');
+      emptyMessageReceived = true;
+    }
+  });
+
+  function exchangeIce(pc) {
+     return function(e) {
+       if (e.candidate) {
+          pc.addIceCandidate(e.candidate);
+       }
+     };
+  }
+
+  function exchangeDescription(pc1, pc2) {
+     return function() {
+       return pc1.setRemoteDescription(pc2.localDescription);
+     };
+  }
+
+  test.step(function() {
+    gFirstConnection = new RTCPeerConnection(null);
+
+    gSecondConnection = new RTCPeerConnection(null);
+
+    gFirstConnection.onicecandidate = exchangeIce(gSecondConnection);
+    gSecondConnection.onicecandidate = exchangeIce(gFirstConnection);
+
+    gSecondConnection.ondatachannel = onReceiveChannel;
+
+    // Note the data channel will preserve the order of messages
+    // exchanged over it by default.
+    sendChannel = gFirstConnection.createDataChannel('sendDataChannel');
+    sendChannel.onopen = onSendChannelOpen;
+
+    gFirstConnection.createOffer()
+    .then(gFirstConnection.setLocalDescription.bind(gFirstConnection))
+    .then(exchangeDescription(gSecondConnection, gFirstConnection))
+    .then(function() {
+      return gSecondConnection.createAnswer();
+    })
+    .then(gSecondConnection.setLocalDescription.bind(gSecondConnection))
+    .then(exchangeDescription(gFirstConnection, gSecondConnection))
+    .catch(test.step_func(function(e) {
+      assert_unreached('Error ' + e.name + ': ' + e.message);
+     }));
+  });
+</script>
+
+</body>
+</html>
+
diff --git a/testing/web-platform/tests/webrtc/no-media-call.html b/testing/web-platform/tests/webrtc/no-media-call.html
new file mode 100644
index 000000000..5059ba6db
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/no-media-call.html
@@ -0,0 +1,139 @@
+<!doctype html>
+<!--
+This test uses the legacy callback API with no media, and thus does not require fake media devices.
+-->
+
+<html>
+<head>
+  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+  <title>RTCPeerConnection No-Media Connection Test</title>
+</head>
+<body>
+  <div id="log"></div>
+  <h2>iceConnectionState info</h2>
+  <div id="stateinfo">
+  </div>
+
+  <!-- These files are in place when executing on W3C. -->
+  <script src="/resources/testharness.js"></script>
+  <script src="/resources/testharnessreport.js"></script>
+  <script type="text/javascript">
+  var test = async_test('Can set up a basic WebRTC call with no data.');
+
+  var gFirstConnection = null;
+  var gSecondConnection = null;
+
+  var onOfferCreated = test.step_func(function(offer) {
+    gFirstConnection.setLocalDescription(offer, ignoreSuccess,
+                                         failed('setLocalDescription first'));
+
+    // This would normally go across the application's signaling solution.
+    // In our case, the "signaling" is to call this function.
+    receiveCall(offer.sdp);
+  });
+
+  function receiveCall(offerSdp) {
+
+    var parsedOffer = new RTCSessionDescription({ type: 'offer',
+                                                  sdp: offerSdp });
+    // These functions use the legacy interface extensions to RTCPeerConnection.
+    gSecondConnection.setRemoteDescription(parsedOffer,
+      function() {
+        gSecondConnection.createAnswer(onAnswerCreated,
+                                       failed('createAnswer'));
+      },
+      failed('setRemoteDescription second'));
+  };
+
+  var onAnswerCreated = test.step_func(function(answer) {
+    gSecondConnection.setLocalDescription(answer, ignoreSuccess,
+                                          failed('setLocalDescription second'));
+
+    // Similarly, this would go over the application's signaling solution.
+    handleAnswer(answer.sdp);
+  });
+
+  function handleAnswer(answerSdp) {
+    var parsedAnswer = new RTCSessionDescription({ type: 'answer',
+                                                   sdp: answerSdp });
+    gFirstConnection.setRemoteDescription(parsedAnswer, ignoreSuccess,
+                                          failed('setRemoteDescription first'));
+  };
+
+  var onIceCandidateToFirst = test.step_func(function(event) {
+    // If event.candidate is null = no more candidates.
+    if (event.candidate) {
+      gSecondConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onIceCandidateToSecond = test.step_func(function(event) {
+    if (event.candidate) {
+      gFirstConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onRemoteStream = test.step_func(function(event) {
+    assert_unreached('WebRTC received a stream when there was none');
+  });
+
+  var onIceConnectionStateChange = test.step_func(function(event) {
+    assert_equals(event.type, 'iceconnectionstatechange');
+    assert_not_equals(gFirstConnection.iceConnectionState, "failed", "iceConnectionState of first connection");
+    assert_not_equals(gSecondConnection.iceConnectionState, "failed", "iceConnectionState of second connection");
+    var stateinfo = document.getElementById('stateinfo');
+    stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
+                        + '<br>Second: ' + gSecondConnection.iceConnectionState;
+    // Note: All these combinations are legal states indicating that the
+    // call has connected. All browsers should end up in completed/completed,
+    // but as of this moment, we've chosen to terminate the test early.
+    // TODO: Revise test to ensure completed/completed is reached.
+    if (gFirstConnection.iceConnectionState == 'connected' &&
+        gSecondConnection.iceConnectionState == 'connected') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'connected' &&
+        gSecondConnection.iceConnectionState == 'completed') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'completed' &&
+        gSecondConnection.iceConnectionState == 'connected') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'completed' &&
+        gSecondConnection.iceConnectionState == 'completed') {
+      test.done()
+    }
+  });
+
+  // Returns a suitable error callback.
+  function failed(function_name) {
+    return test.step_func(function() {
+      assert_unreached('WebRTC called error callback for ' + function_name);
+    });
+  }
+
+  // Returns a suitable do-nothing.
+  function ignoreSuccess(function_name) {
+  }
+
+  // This function starts the test.
+  test.step(function() {
+    gFirstConnection = new RTCPeerConnection(null);
+    gFirstConnection.onicecandidate = onIceCandidateToFirst;
+    gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;
+
+    gSecondConnection = new RTCPeerConnection(null);
+    gSecondConnection.onicecandidate = onIceCandidateToSecond;
+    gSecondConnection.onaddstream = onRemoteStream;
+    gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;
+
+    // The offerToReceiveVideo is necessary and sufficient to make
+    // an actual connection.
+    gFirstConnection.createOffer(onOfferCreated, failed('createOffer'),
+        {offerToReceiveVideo: true});
+  });
+</script>
+
+</body>
+</html>
diff --git a/testing/web-platform/tests/webrtc/promises-call.html b/testing/web-platform/tests/webrtc/promises-call.html
new file mode 100644
index 000000000..d83508a38
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/promises-call.html
@@ -0,0 +1,121 @@
+<!doctype html>
+<!--
+This test uses data only, and thus does not require fake media devices.
+-->
+
+<html>
+<head>
+  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+  <title>RTCPeerConnection Data-Only Connection Test with Promises</title>
+</head>
+<body>
+  <div id="log"></div>
+  <h2>iceConnectionState info</h2>
+  <div id="stateinfo">
+  </div>
+
+  <!-- These files are in place when executing on W3C. -->
+  <script src="/resources/testharness.js"></script>
+  <script src="/resources/testharnessreport.js"></script>
+  <script type="text/javascript">
+  var test = async_test('Can set up a basic WebRTC call with only data using promises.');
+
+  var gFirstConnection = null;
+  var gSecondConnection = null;
+
+  var onIceCandidateToFirst = test.step_func(function(event) {
+    // If event.candidate is null = no more candidates.
+    if (event.candidate) {
+      gSecondConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onIceCandidateToSecond = test.step_func(function(event) {
+    if (event.candidate) {
+      gFirstConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onRemoteStream = test.step_func(function(event) {
+    assert_unreached('WebRTC received a stream when there was none');
+  });
+
+  var onIceConnectionStateChange = test.step_func(function(event) {
+    assert_equals(event.type, 'iceconnectionstatechange');
+    var stateinfo = document.getElementById('stateinfo');
+    stateinfo.innerHTML = 'First: ' + gFirstConnection.iceConnectionState
+                        + '<br>Second: ' + gSecondConnection.iceConnectionState;
+    // Note: All these combinations are legal states indicating that the
+    // call has connected. All browsers should end up in completed/completed,
+    // but as of this moment, we've chosen to terminate the test early.
+    // TODO: Revise test to ensure completed/completed is reached.
+    if (gFirstConnection.iceConnectionState == 'connected' &&
+        gSecondConnection.iceConnectionState == 'connected') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'connected' &&
+        gSecondConnection.iceConnectionState == 'completed') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'completed' &&
+        gSecondConnection.iceConnectionState == 'connected') {
+      test.done()
+    }
+    if (gFirstConnection.iceConnectionState == 'completed' &&
+        gSecondConnection.iceConnectionState == 'completed') {
+      test.done()
+    }
+  });
+
+  // This function starts the test.
+  test.step(function() {
+    gFirstConnection = new RTCPeerConnection(null);
+    gFirstConnection.onicecandidate = onIceCandidateToFirst;
+    gFirstConnection.oniceconnectionstatechange = onIceConnectionStateChange;
+
+    gSecondConnection = new RTCPeerConnection(null);
+    gSecondConnection.onicecandidate = onIceCandidateToSecond;
+    gSecondConnection.onaddstream = onRemoteStream;
+    gSecondConnection.oniceconnectionstatechange = onIceConnectionStateChange;
+
+    // The createDataChannel is necessary and sufficient to make
+    // sure the ICE connection be attempted.
+    gFirstConnection.createDataChannel('channel');
+
+    var atStep = 'Create offer';
+
+    gFirstConnection.createOffer()
+    .then(function(offer) {
+      atStep = 'Set local description at first';
+      return gFirstConnection.setLocalDescription(offer);
+    })
+    .then(function() {
+      atStep = 'Set remote description at second';
+      return gSecondConnection.setRemoteDescription(
+          gFirstConnection.localDescription);
+    })
+    .then(function() {
+      atStep = 'Create answer';
+      return gSecondConnection.createAnswer();
+    })
+    .then(function(answer) {
+      atStep = 'Set local description at second';
+      return gSecondConnection.setLocalDescription(answer);
+    })
+    .then(function() {
+      atStep = 'Set remote description at first';
+      return gFirstConnection.setRemoteDescription(
+          gSecondConnection.localDescription);
+    })
+    .then(function() {
+      atStep = 'Negotiation completed';
+    })
+    .catch(test.step_func(function(e) {
+      assert_unreached('Error ' + e.name + ': ' + e.message +
+                       ' happened at step ' + atStep);
+    }));
+  });
+</script>
+
+</body>
+</html>
diff --git a/testing/web-platform/tests/webrtc/rtcpeerconnection/rtcpeerconnection-idl.html b/testing/web-platform/tests/webrtc/rtcpeerconnection/rtcpeerconnection-idl.html
new file mode 100644
index 000000000..298cfb5ca
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/rtcpeerconnection/rtcpeerconnection-idl.html
@@ -0,0 +1,107 @@
+<!doctype html>
+<html>
+<head>
+<meta charset=utf-8>
+<title>IDL check of RTCPeerConnection</title>
+<link rel="author" title="Harald Alvestrand" href="mailto:hta@google.com"/>
+<link rel="help" href="http://w3c.github.io/webrtc-pc/#rtcpeerconnection-interface">
+</head>
+<body>
+
+<h1 class="instructions">Description</h1>
+<p class="instructions">This test verifies the availability of the RTCPeerConnection interface.</p>
+<div id='log'></div>
+<script src=/resources/testharness.js></script>
+<script src=/resources/testharnessreport.js></script>
+<script src=/resources/WebIDLParser.js></script>
+<script src=/resources/idlharness.js></script>
+
+<!-- The IDL is copied from the 22 September 2015 editors' draft. -->
+<script type="text/plain">
+interface EventTarget {
+    // Only a dummy definition is needed here.
+};
+[ Constructor (optional RTCConfiguration configuration)]
+interface RTCPeerConnection : EventTarget  {
+    Promise<RTCSessionDescription> createOffer (optional RTCOfferOptions options);
+    Promise<RTCSessionDescription> createAnswer (optional RTCAnswerOptions options);
+    Promise<void>                  setLocalDescription (RTCSessionDescription description);
+    readonly    attribute RTCSessionDescription? localDescription;
+    readonly    attribute RTCSessionDescription? currentLocalDescription;
+    readonly    attribute RTCSessionDescription? pendingLocalDescription;
+    Promise<void>                  setRemoteDescription (RTCSessionDescription description);
+    readonly    attribute RTCSessionDescription? remoteDescription;
+    readonly    attribute RTCSessionDescription? currentRemoteDescription;
+    readonly    attribute RTCSessionDescription? pendingRemoteDescription;
+    Promise<void>                  addIceCandidate (RTCIceCandidate candidate);
+    readonly    attribute RTCSignalingState      signalingState;
+    readonly    attribute RTCIceGatheringState   iceGatheringState;
+    readonly    attribute RTCIceConnectionState  iceConnectionState;
+    readonly    attribute boolean?               canTrickleIceCandidates;
+    RTCConfiguration               getConfiguration ();
+    void                           setConfiguration (RTCConfiguration configuration);
+    void                           close ();
+                attribute EventHandler           onnegotiationneeded;
+                attribute EventHandler           onicecandidate;
+                attribute EventHandler           onsignalingstatechange;
+                attribute EventHandler           oniceconnectionstatechange;
+                attribute EventHandler           onicegatheringstatechange;
+};
+
+partial interface RTCPeerConnection {
+    void createOffer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback, optional RTCOfferOptions options);
+    void setLocalDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
+    void createAnswer (RTCSessionDescriptionCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
+    void setRemoteDescription (RTCSessionDescription description, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
+    void addIceCandidate (RTCIceCandidate candidate, VoidFunction successCallback, RTCPeerConnectionErrorCallback failureCallback);
+    void getStats (MediaStreamTrack? selector, RTCStatsCallback successCallback, RTCPeerConnectionErrorCallback failureCallback);
+};
+
+partial interface RTCPeerConnection {
+    static Promise<RTCCertificate> generateCertificate (AlgorithmIdentifier keygenAlgorithm);
+};
+
+partial interface RTCPeerConnection {
+    sequence<RTCRtpSender>   getSenders ();
+    sequence<RTCRtpReceiver> getReceivers ();
+    RTCRtpSender             addTrack (MediaStreamTrack track, MediaStream... streams);
+    void                     removeTrack (RTCRtpSender sender);
+                attribute EventHandler ontrack;
+};
+
+partial interface RTCPeerConnection {
+    RTCDataChannel createDataChannel ([TreatNullAs=EmptyString] DOMString label, optional RTCDataChannelInit dataChannelDict);
+                attribute EventHandler ondatachannel;
+};
+
+partial interface RTCPeerConnection {
+    readonly    attribute RTCDTMFSender? dtmf;
+};
+
+partial interface RTCPeerConnection {
+    Promise<RTCStatsReport> getStats (optional MediaStreamTrack? selector);
+};
+
+partial interface RTCPeerConnection {
+    void               setIdentityProvider (DOMString provider, optional DOMString protocol, optional DOMString usernameHint);
+    Promise<DOMString> getIdentityAssertion ();
+    readonly    attribute Promise<RTCIdentityAssertion> peerIdentity;
+    readonly    attribute DOMString?                    idpLoginUrl;
+};
+
+</script>
+<script>
+(function() {
+  var idl_array = new IdlArray();
+  [].forEach.call(document.querySelectorAll("script[type=text\\/plain]"),
+                  function(node) {
+                    idl_array.add_idls(node.textContent);
+                  });
+  window.pc = new RTCPeerConnection(null);
+  idl_array.add_objects({"RTCPeerConnection": ["pc"]});
+  idl_array.test();
+  done();
+})();
+</script>
+</body>
+</html>
diff --git a/testing/web-platform/tests/webrtc/simplecall.html b/testing/web-platform/tests/webrtc/simplecall.html
new file mode 100644
index 000000000..698586894
--- /dev/null
+++ b/testing/web-platform/tests/webrtc/simplecall.html
@@ -0,0 +1,118 @@
+<!doctype html>
+<!--
+To quickly iterate when developing this test, use --use-fake-ui-for-media-stream
+for Chrome and set the media.navigator.permission.disabled property to true in
+Firefox. You must either have a webcam/mic available on the system or use for
+instance --use-fake-device-for-media-stream for Chrome.
+-->
+
+<html>
+<head>
+  <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
+  <title>RTCPeerConnection Connection Test</title>
+</head>
+<body>
+  <div id="log"></div>
+  <div>
+    <video id="local-view" autoplay="autoplay"></video>
+    <video id="remote-view" autoplay="autoplay"/>
+    </video>
+  </div>
+
+  <!-- These files are in place when executing on W3C. -->
+  <script src="/resources/testharness.js"></script>
+  <script src="/resources/testharnessreport.js"></script>
+  <script src="/common/vendor-prefix.js"
+          data-prefixed-objects=
+              '[{"ancestors":["navigator"], "name":"getUserMedia"}]'
+          data-prefixed-prototypes=
+               '[{"ancestors":["HTMLMediaElement"],"name":"srcObject"}]'>
+  </script>
+  <script type="text/javascript">
+  var test = async_test('Can set up a basic WebRTC call.', {timeout: 5000});
+
+  var gFirstConnection = null;
+  var gSecondConnection = null;
+
+  function getUserMediaOkCallback(localStream) {
+    gFirstConnection = new RTCPeerConnection(null);
+    gFirstConnection.onicecandidate = onIceCandidateToFirst;
+    gFirstConnection.addStream(localStream);
+    gFirstConnection.createOffer(onOfferCreated, failed('createOffer'));
+
+    var videoTag = document.getElementById('local-view');
+    videoTag.srcObject = localStream;
+  };
+
+  var onOfferCreated = test.step_func(function(offer) {
+    gFirstConnection.setLocalDescription(offer);
+
+    // This would normally go across the application's signaling solution.
+    // In our case, the "signaling" is to call this function.
+    receiveCall(offer.sdp);
+  });
+
+  function receiveCall(offerSdp) {
+    gSecondConnection = new RTCPeerConnection(null);
+    gSecondConnection.onicecandidate = onIceCandidateToSecond;
+    gSecondConnection.onaddstream = onRemoteStream;
+
+    var parsedOffer = new RTCSessionDescription({ type: 'offer',
+                                                  sdp: offerSdp });
+    gSecondConnection.setRemoteDescription(parsedOffer);
+
+    gSecondConnection.createAnswer(onAnswerCreated,
+                                   failed('createAnswer'));
+  };
+
+  var onAnswerCreated = test.step_func(function(answer) {
+    gSecondConnection.setLocalDescription(answer);
+
+    // Similarly, this would go over the application's signaling solution.
+    handleAnswer(answer.sdp);
+  });
+
+  function handleAnswer(answerSdp) {
+    var parsedAnswer = new RTCSessionDescription({ type: 'answer',
+                                                   sdp: answerSdp });
+    gFirstConnection.setRemoteDescription(parsedAnswer);
+
+    // Call negotiated: done.
+    test.done();
+  };
+
+  var onIceCandidateToFirst = test.step_func(function(event) {
+    // If event.candidate is null = no more candidates.
+    if (event.candidate) {
+      gSecondConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onIceCandidateToSecond = test.step_func(function(event) {
+    if (event.candidate) {
+      gFirstConnection.addIceCandidate(event.candidate);
+    }
+  });
+
+  var onRemoteStream = test.step_func(function(event) {
+    var videoTag = document.getElementById('remote-view');
+    videoTag.srcObject = event.stream;
+  });
+
+  // Returns a suitable error callback.
+  function failed(function_name) {
+    return test.step_func(function() {
+      assert_unreached('WebRTC called error callback for ' + function_name);
+    });
+  }
+
+  // This function starts the test.
+  test.step(function() {
+    navigator.getUserMedia({ video: true, audio: true },
+        getUserMediaOkCallback,
+        failed('getUserMedia'));
+  });
+</script>
+
+</body>
+</html>
author	Matt A. Tobin <mattatobin@localhost.localdomain>	2018-02-02 04:16:08 -0500
committer	Matt A. Tobin <mattatobin@localhost.localdomain>	2018-02-02 04:16:08 -0500
commit	5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree	10027f336435511475e392454359edea8e25895d /testing/web-platform/tests/webrtc
parent	49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
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