154 lines
5.5 KiB
HTML
154 lines
5.5 KiB
HTML
<!doctype html>
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<html>
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<head>
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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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<title>WebRTC video.requestVideoFrameCallback() test</title>
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<script src="/webrtc/RTCPeerConnection-helper.js"></script>
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</head>
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<body>
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<div id="log"></div>
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<div>
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<video id="local-view" muted autoplay="autoplay"></video>
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<video id="remote-view" muted autoplay="autoplay"></video>
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</div>
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<!-- These files are in place when executing on W3C. -->
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<script src="/resources/testharness.js"></script>
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<script src="/resources/testharnessreport.js"></script>
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<script type="text/javascript">
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var test = async_test('Test video.requestVideoFrameCallback() parameters for WebRTC applications.');
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//
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// This test is based on /webrtc/simplecall.https.html, but it calls to
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// video.requestVideoFrameCallback() before ending, to verify WebRTC required
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// and optional parameters.
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//
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var gFirstConnection = null;
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var gSecondConnection = null;
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var verify_params = (now, metadata) => {
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assert_greater_than(now, 0);
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// Verify all required fields
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assert_greater_than(metadata.presentationTime, 0);
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assert_greater_than(metadata.expectedDisplayTime, 0);
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assert_greater_than(metadata.presentedFrames, 0);
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assert_greater_than(metadata.width, 0);
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assert_greater_than(metadata.height, 0);
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assert_true("mediaTime" in metadata, "mediaTime should be present");
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// Verify WebRTC only fields.
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assert_true("rtpTimestamp" in metadata, "rtpTimestamp should be present");
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assert_true("receiveTime" in metadata, "receiveTime should be present");
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// Verifying that captureTime is too unreliable to be included in tests.
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}
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var verify_local_metadata = (now, metadata) => {
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assert_greater_than(metadata.expectedDisplayTime, 0);
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assert_greater_than(metadata.presentedFrames, 0);
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assert_greater_than(metadata.width, 0);
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assert_greater_than(metadata.height, 0);
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assert_true("captureTime" in metadata, "captureTime should always be present for local sources.");
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assert_greater_than(metadata.captureTime, 0);
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}
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// If the remote video gets video data that implies the negotiation
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// as well as the ICE and DTLS connection are up.
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document.getElementById('remote-view')
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.addEventListener('loadedmetadata', function() {
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document.getElementById('remote-view').requestVideoFrameCallback(test.step_func_done(verify_params));
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});
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document.getElementById('local-view')
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.addEventListener('loadmetadata', function() {
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document.getElementById('local-view').requestVideoFrameCallback(test.step_func_done(verify_local_metadata));
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});
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function getNoiseStreamOkCallback(localStream) {
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gFirstConnection = new RTCPeerConnection(null);
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test.add_cleanup(() => gFirstConnection.close());
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gFirstConnection.onicecandidate = onIceCandidateToFirst;
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gSecondConnection = new RTCPeerConnection(null);
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test.add_cleanup(() => gSecondConnection.close());
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gSecondConnection.onicecandidate = onIceCandidateToSecond;
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gSecondConnection.ontrack = onRemoteTrack;
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localStream.getTracks().forEach(function(track) {
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// Bidirectional streams are needed in order for captureTime to be
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// populated. Use the same source in both directions.
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gFirstConnection.addTrack(track, localStream);
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gSecondConnection.addTrack(track, localStream);
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});
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gFirstConnection.createOffer().then(onOfferCreated, failed('createOffer'));
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var videoTag = document.getElementById('local-view');
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videoTag.srcObject = localStream;
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};
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var onOfferCreated = test.step_func(function(offer) {
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gFirstConnection.setLocalDescription(offer);
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// This would normally go across the application's signaling solution.
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// In our case, the "signaling" is to call this function.
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receiveCall(offer.sdp);
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});
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function receiveCall(offerSdp) {
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var parsedOffer = new RTCSessionDescription({ type: 'offer',
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sdp: offerSdp });
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gSecondConnection.setRemoteDescription(parsedOffer);
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gSecondConnection.createAnswer().then(onAnswerCreated,
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failed('createAnswer'));
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};
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var onAnswerCreated = test.step_func(function(answer) {
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gSecondConnection.setLocalDescription(answer);
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// Similarly, this would go over the application's signaling solution.
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handleAnswer(answer.sdp);
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});
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function handleAnswer(answerSdp) {
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var parsedAnswer = new RTCSessionDescription({ type: 'answer',
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sdp: answerSdp });
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gFirstConnection.setRemoteDescription(parsedAnswer);
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};
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var onIceCandidateToFirst = test.step_func(function(event) {
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// If event.candidate is null = no more candidates.
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if (event.candidate) {
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gSecondConnection.addIceCandidate(event.candidate);
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}
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});
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var onIceCandidateToSecond = test.step_func(function(event) {
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if (event.candidate) {
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gFirstConnection.addIceCandidate(event.candidate);
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}
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});
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var onRemoteTrack = test.step_func(function(event) {
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var videoTag = document.getElementById('remote-view');
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if (!videoTag.srcObject) {
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videoTag.srcObject = event.streams[0];
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}
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});
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// Returns a suitable error callback.
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function failed(function_name) {
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return test.unreached_func('WebRTC called error callback for ' + function_name);
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}
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// This function starts the test.
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test.step(function() {
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getNoiseStream({ video: true, audio: true })
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.then(test.step_func(getNoiseStreamOkCallback), failed('getNoiseStream'));
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});
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</script>
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</body>
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</html>
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