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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
| commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
| tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html | |
| parent | Initial commit. (diff) | |
| download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip | |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html | 118 |
1 files changed, 118 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html b/testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html new file mode 100644 index 0000000000..f2e2084623 --- /dev/null +++ b/testing/web-platform/tests/webrtc/simplecall-no-ssrcs.https.html @@ -0,0 +1,118 @@ +<!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 without announcing ssrcs.'); + + 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().then(onOfferCreated, failed('createOffer')); + + var videoTag = document.getElementById('local-view'); + videoTag.srcObject = localStream; + }; + + var onOfferCreated = test.step_func(function(offer) { + gFirstConnection.setLocalDescription(offer); + + // remove all a=ssrc: lines and the (obsolete) msid-semantic line. + var sdp = offer.sdp.replace(/^a=ssrc:.*$\r\n/gm, '') + .replace(/^a=msid-semantic.*$\r\n/gm, ''); + + // This would normally go across the application's signaling solution. + // In our case, the "signaling" is to call this function. + receiveCall(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().then(onAnswerCreated, + failed('createAnswer')); + }; + + var onAnswerCreated = test.step_func(function(answer) { + gSecondConnection.setLocalDescription(answer); + + // remove all a=ssrc: lines, the msid-semantic line and any a=msid:. + var sdp = answer.sdp.replace(/^a=ssrc:.*$\r\n/gm, '') + .replace(/^a=msid-semantic.*$\r\n/gm, '') + .replace(/^a=msid:.*$\r\n/gm, ''); + + // Similarly, this would go over the application's signaling solution. + handleAnswer(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> |