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<!doctype html>
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
<title>WebRTC video.requestVideoFrameCallback() test</title>
<script src="/webrtc/RTCPeerConnection-helper.js"></script>
</head>
<body>
<div id="log"></div>
<div>
<video id="local-view" muted autoplay="autoplay"></video>
<video id="remote-view" muted 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 type="text/javascript">
var test = async_test('Test video.requestVideoFrameCallback() parameters for WebRTC applications.');
//
// This test is based on /webrtc/simplecall.https.html, but it calls to
// video.requestVideoFrameCallback() before ending, to verify WebRTC required
// and optional parameters.
//
var gFirstConnection = null;
var gSecondConnection = null;
var gCallbackCounter = 0;
var verify_params = (now, metadata) => {
gCallbackCounter = gCallbackCounter + 1;
assert_greater_than(now, 0);
// Verify all required fields
assert_greater_than(metadata.presentationTime, 0);
assert_greater_than(metadata.expectedDisplayTime, 0);
assert_greater_than(metadata.presentedFrames, 0);
assert_greater_than(metadata.width, 0);
assert_greater_than(metadata.height, 0);
assert_true("mediaTime" in metadata, "mediaTime should be present");
// Verify WebRTC only fields.
assert_true("rtpTimestamp" in metadata, "rtpTimestamp should be present");
assert_true("receiveTime" in metadata, "receiveTime should be present");
// captureTime is not available until roundtrip time estimation is done.
if (gCallbackCounter > 60 || "captureTime" in metadata) {
assert_true("captureTime" in metadata, "captureTime should be present");
test.done();
}
else {
// Keep requesting callbacks.
document.getElementById('remote-view').requestVideoFrameCallback(test.step_func(verify_params));
}
}
var verify_local_metadata = (now, metadata) => {
assert_greater_than(metadata.expectedDisplayTime, 0);
assert_greater_than(metadata.presentedFrames, 0);
assert_greater_than(metadata.width, 0);
assert_greater_than(metadata.height, 0);
assert_true("captureTime" in metadata, "captureTime should always be present for local sources.");
assert_greater_than(metadata.captureTime, 0);
}
// If the remote video gets video data that implies the negotiation
// as well as the ICE and DTLS connection are up.
document.getElementById('remote-view')
.addEventListener('loadedmetadata', function() {
document.getElementById('remote-view').requestVideoFrameCallback(test.step_func(verify_params));
});
document.getElementById('local-view')
.addEventListener('loadmetadata', function() {
document.getElementById('local-view').requestVideoFrameCallback(test.step_func_done(verify_local_metadata));
});
function getNoiseStreamOkCallback(localStream) {
gFirstConnection = new RTCPeerConnection(null);
test.add_cleanup(() => gFirstConnection.close());
gFirstConnection.onicecandidate = onIceCandidateToFirst;
gSecondConnection = new RTCPeerConnection(null);
test.add_cleanup(() => gSecondConnection.close());
gSecondConnection.onicecandidate = onIceCandidateToSecond;
gSecondConnection.ontrack = onRemoteTrack;
localStream.getTracks().forEach(function(track) {
// Bidirectional streams are needed in order for captureTime to be
// populated. Use the same source in both directions.
gFirstConnection.addTrack(track, localStream);
gSecondConnection.addTrack(track, localStream);
});
gFirstConnection.createOffer().then(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) {
var parsedOffer = new RTCSessionDescription({ type: 'offer',
sdp: offerSdp });
gSecondConnection.setRemoteDescription(parsedOffer);
gSecondConnection.createAnswer().then(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);
};
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 onRemoteTrack = test.step_func(function(event) {
var videoTag = document.getElementById('remote-view');
if (!videoTag.srcObject) {
videoTag.srcObject = event.streams[0];
}
});
// Returns a suitable error callback.
function failed(function_name) {
return test.unreached_func('WebRTC called error callback for ' + function_name);
}
// This function starts the test.
test.step(function() {
getNoiseStream({ video: true, audio: true })
.then(test.step_func(getNoiseStreamOkCallback), failed('getNoiseStream'));
});
</script>
</body>
</html>
|
