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| 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/simplecall.html | |
| parent | 49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff) | |
| download | UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.gz UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.lz UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.xz UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.zip | |
Add m-esr52 at 52.6.0
Diffstat (limited to 'testing/web-platform/tests/webrtc/simplecall.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc/simplecall.html | 118 |
1 files changed, 118 insertions, 0 deletions
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> |