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webrtc wpt: add legacy callback based variant of simplecall
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copied from webrtc/simplecall.https.html
without cleaning up the style. See
  #18236

BUG=None

Change-Id: Icbfb02768f455203f22b6bb357f448d31e323037
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/5217871
Reviewed-by: Harald Alvestrand <hta@chromium.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#1249892}
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fippo authored and chromium-wpt-export-bot committed Jan 21, 2024
1 parent a10b8c0 commit 5d54018
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108 changes: 108 additions & 0 deletions webrtc/legacy/simplecall_callbacks.https.html
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<!doctype html>
<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" 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('Can set up a basic WebRTC call.');

var gFirstConnection = null;
var gSecondConnection = null;

// 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() {
// Call negotiated: done.
test.done();
});

function getNoiseStreamOkCallback(localStream) {
gFirstConnection = new RTCPeerConnection(null);
test.add_cleanup(() => gFirstConnection.close());
gFirstConnection.onicecandidate = onIceCandidateToFirst;
localStream.getTracks().forEach(function(track) {
gFirstConnection.addTrack(track, 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);
test.add_cleanup(() => gSecondConnection.close());
gSecondConnection.onicecandidate = onIceCandidateToSecond;
gSecondConnection.ontrack = onRemoteTrack;

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);
};

var onIceCandidateToFirst = test.step_func(function(event) {
gSecondConnection.addIceCandidate(event.candidate);
});

var onIceCandidateToSecond = test.step_func(function(event) {
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>

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