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|
<html>
<head>
<script type="text/javascript" src="webrtc_test_utilities.js"></script>
<script type="text/javascript" src="webrtc_test_audio.js"></script>
<script type="text/javascript">
$ = function(id) {
return document.getElementById(id);
};
window.onerror = function(errorMsg, url, lineNumber, column, errorObj) {
failTest('Error: ' + errorMsg + '\nScript: ' + url +
'\nLine: ' + lineNumber + '\nColumn: ' + column +
'\nStackTrace: ' + errorObj);
}
var gFirstConnection = null;
var gSecondConnection = null;
var gTestWithoutMsid = false;
var gLocalStream = null;
var gSentTones = '';
var gRemoteStreams = {};
// Default transform functions, overridden by some test cases.
var transformSdp = function(sdp) { return sdp; };
var transformRemoteSdp = function(sdp) { return sdp; };
var onLocalDescriptionError = function(error) { failTest(error); };
var onRemoteDescriptionError = function(error) { failTest(error); };
// When using external SDES, the crypto key is chosen by javascript.
var EXTERNAL_SDES_LINES = {
'audio': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
'inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR',
'video': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
'inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj',
'data': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
'inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj'
};
setAllEventsOccuredHandler(reportTestSuccess);
// Test that we can setup a call with an audio and video track (must request
// video in this call since we expect video to be playing).
function call(constraints) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test that we can setup a call with a video track and that the remote peer
// receives black frames if the local video track is disabled.
function callAndDisableLocalVideo(constraints) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
detectVideoPlaying('remote-view-1',
function () {
assertEquals(gLocalStream.getVideoTracks().length, 1);
gLocalStream.getVideoTracks()[0].enabled = false;
waitForBlackVideo('remote-view-1');
});
}
// Test that we can setup call with an audio and video track and check that
// the video resolution is as expected.
function callAndExpectResolution(constraints,
expected_width,
expected_height) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideoWithResolution('remote-view-1',
expected_width,
expected_height);
waitForVideoWithResolution('remote-view-2',
expected_width,
expected_height);
}
// First calls without streams on any connections, and then adds a stream
// to peer connection 1 which gets sent to peer connection 2. We must wait
// for the first negotiation to complete before starting the second one, which
// is why we wait until the connection is stable before re-negotiating.
function callEmptyThenAddOneStreamAndRenegotiate(constraints) {
createConnections(null);
negotiate();
waitForConnectionToStabilize(gFirstConnection, function() {
navigator.webkitGetUserMedia(constraints,
addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
// Only the first connection is sending here.
waitForVideo('remote-view-2');
});
}
// The second set of constraints should request video (e.g. video:true) since
// we expect video to be playing after the second renegotiation.
function callAndRenegotiateToVideo(constraints, renegotiationConstraints) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForConnectionToStabilize(gFirstConnection, function() {
gFirstConnection.removeStream(gLocalStream);
gSecondConnection.removeStream(gLocalStream);
navigator.webkitGetUserMedia(renegotiationConstraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
});
}
// The second set of constraints should request audio (e.g. audio:true) since
// we expect audio to be playing after the second renegotiation.
function callAndRenegotiateToAudio(constraints, renegotiationConstraints) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForConnectionToStabilize(gFirstConnection, function() {
gFirstConnection.removeStream(gLocalStream);
gSecondConnection.removeStream(gLocalStream);
navigator.webkitGetUserMedia(renegotiationConstraints,
addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
var onCallEstablished = function() {
ensureAudioPlaying(gSecondConnection);
};
waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
});
}
// First makes a call between pc1 and pc2 where a stream is sent from pc1 to
// pc2. The stream sent from pc1 to pc2 is cloned from the stream received on
// pc2 to test that cloning of remote video tracks works as intended and is
// sent back to pc1.
function callAndForwardRemoteStream(constraints) {
createConnections(null);
navigator.webkitGetUserMedia(constraints,
addStreamToTheFirstConnectionAndNegotiate,
printGetUserMediaError);
var onRemoteStream2 = function() {
// Video has been detected to be playing in pc2. Clone the received
// stream and send it back to pc1.
gSecondConnection.addStream(gRemoteStreams['remote-view-2'].clone());
negotiate();
}
// Wait for remove video to be playing in pc2. Once video is playing,
// forward the remove stream from pc2 to pc1.
detectVideoPlaying('remote-view-2', onRemoteStream2);
// Wait for video to be forwarded back to connection 1.
waitForVideo('remote-view-1');
}
// First makes a call between pc1 and pc2, and then construct a new media
// stream using the remote audio and video tracks, connect the new media
// stream to a video element. These operations should not crash Chrome.
function ConnectChromiumSinkToRemoteAudioTrack() {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate,
printGetUserMediaError);
detectVideoPlaying('remote-view-2', function() {
// Construct a new media stream with remote tracks.
var newStream = new webkitMediaStream();
newStream.addTrack(
gSecondConnection.getRemoteStreams()[0].getAudioTracks()[0]);
newStream.addTrack(
gSecondConnection.getRemoteStreams()[0].getVideoTracks()[0]);
var videoElement = document.createElement('video');
// No crash for this operation.
videoElement.src = URL.createObjectURL(newStream);
waitForVideo('remote-view-2');
});
}
// Test that we can setup call with an audio and video track and
// simulate that the remote peer don't support MSID.
function callWithoutMsidAndBundle() {
createConnections(null);
transformSdp = removeBundle;
transformRemoteSdp = removeMsid;
gTestWithoutMsid = true;
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test that we can't setup a call with an unsupported video codec
function negotiateUnsupportedVideoCodec() {
createConnections(null);
transformSdp = removeVideoCodec;
onLocalDescriptionError = function(error) {
var expectedMsg = 'Failed to set local offer sdp:' +
' Session error code: ERROR_CONTENT. Session error description:' +
' Failed to set local video description recv parameters..';
assertEquals(expectedMsg, error);
reportTestSuccess();
};
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
}
// Test that we can't setup a call if one peer does not support encryption
function negotiateNonCryptoCall() {
createConnections(null);
transformSdp = removeCrypto;
onLocalDescriptionError = function(error) {
var expectedMsg = 'Failed to set local offer sdp:' +
' Called with SDP without DTLS fingerprint.';
assertEquals(expectedMsg, error);
reportTestSuccess();
};
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
}
// Test that we can negotiate a call with an SDP offer that includes a
// b=AS:XX line to control audio and video bandwidth
function negotiateOfferWithBLine() {
createConnections(null);
transformSdp = addBandwithControl;
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test that we can setup call with legacy settings.
function callWithLegacySdp() {
transformSdp = function(sdp) {
return removeBundle(useGice(useExternalSdes(sdp)));
};
createConnections({
'mandatory': {'RtpDataChannels': true, 'DtlsSrtpKeyAgreement': false}
});
setupDataChannel({reliable: false});
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test only a data channel.
function callWithDataOnly() {
createConnections({optional:[{RtpDataChannels: true}]});
setupDataChannel({reliable: false});
negotiate();
}
function callWithSctpDataOnly() {
createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
setupSctpDataChannel({reliable: true});
negotiate();
}
// Test call with audio, video and a data channel.
function callWithDataAndMedia() {
createConnections({optional:[{RtpDataChannels: true}]});
setupDataChannel({reliable: false});
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate,
printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
function callWithSctpDataAndMedia() {
createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
setupSctpDataChannel({reliable: true});
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate,
printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test call with a data channel and later add audio and video.
function callWithDataAndLaterAddMedia() {
createConnections({optional:[{RtpDataChannels: true}]});
setupDataChannel({reliable: false});
negotiate();
// Set an event handler for when the data channel has been closed.
setAllEventsOccuredHandler(function() {
// When the video is flowing the test is done.
setAllEventsOccuredHandler(reportTestSuccess);
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
});
}
// Test that we can setup call and send DTMF.
function callAndSendDtmf(tones) {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
var onCallEstablished = function() {
// Send DTMF tones. Allocate the sender in the window to keep it from
// being garbage collected. https://crbug.com/486654.
var localAudioTrack = gLocalStream.getAudioTracks()[0];
window.dtmfSender = gFirstConnection.createDTMFSender(localAudioTrack);
window.dtmfSender.ontonechange = onToneChange;
window.dtmfSender.insertDTMF(tones);
// Wait for the DTMF tones callback.
addExpectedEvent();
var waitDtmf = setInterval(function() {
if (gSentTones == tones) {
clearInterval(waitDtmf);
eventOccured();
}
}, 100);
}
// Do the DTMF test after we have received video.
detectVideoPlaying('remote-view-2', onCallEstablished);
}
function testCreateOfferOptions() {
createConnections(null);
var offerOptions = {
'offerToReceiveAudio': false,
'offerToReceiveVideo': true
};
gFirstConnection.createOffer(
function(offer) {
assertEquals(-1, offer.sdp.search('m=audio'));
assertNotEquals(-1, offer.sdp.search('m=video'));
reportTestSuccess();
},
function(error) { failTest(error); },
offerOptions);
}
function callAndEnsureAudioIsPlaying(constraints) {
createConnections(null);
// Add the local stream to gFirstConnection to play one-way audio.
navigator.webkitGetUserMedia(constraints,
addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
var onCallEstablished = function() {
ensureAudioPlaying(gSecondConnection);
};
waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
}
function callWithIsac16KAndEnsureAudioIsPlaying(constraints) {
transformSdp = function(sdp) {
sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
'm=audio $1 RTP/SAVPF 103 126\r\n');
sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
if (sdp.search('a=rtpmap:103 ISAC/16000') == -1)
failTest('Missing iSAC 16K codec on Android; cannot force codec.');
return sdp;
}
callAndEnsureAudioIsPlaying(constraints);
}
function enableRemoteVideo(peerConnection, enabled) {
remoteStream = peerConnection.getRemoteStreams()[0];
remoteStream.getVideoTracks()[0].enabled = enabled;
}
function enableRemoteAudio(peerConnection, enabled) {
remoteStream = peerConnection.getRemoteStreams()[0];
remoteStream.getAudioTracks()[0].enabled = enabled;
}
function enableLocalVideo(peerConnection, enabled) {
localStream = peerConnection.getLocalStreams()[0];
localStream.getVideoTracks()[0].enabled = enabled;
}
function enableLocalAudio(peerConnection, enabled) {
localStream = peerConnection.getLocalStreams()[0];
localStream.getAudioTracks()[0].enabled = enabled;
}
function callAndEnsureRemoteAudioTrackMutingWorks() {
callAndEnsureAudioIsPlaying({audio: true, video: true});
setAllEventsOccuredHandler(function() {
setAllEventsOccuredHandler(reportTestSuccess);
// Call is up, now mute the remote track and check we stop playing out
// audio (after a small delay, we don't expect it to happen instantly).
enableRemoteAudio(gSecondConnection, false);
ensureSilence(gSecondConnection);
});
}
function callAndEnsureLocalAudioTrackMutingWorks() {
callAndEnsureAudioIsPlaying({audio: true, video: true});
setAllEventsOccuredHandler(function() {
setAllEventsOccuredHandler(reportTestSuccess);
// Call is up, now mute the local track of the sending side and ensure
// the receiving side stops receiving audio.
enableLocalAudio(gFirstConnection, false);
ensureSilence(gSecondConnection);
});
}
function callAndEnsureAudioTrackUnmutingWorks() {
callAndEnsureAudioIsPlaying({audio: true, video: true});
setAllEventsOccuredHandler(function() {
setAllEventsOccuredHandler(reportTestSuccess);
// Mute, wait a while, unmute, verify audio gets back up.
// (Also, ensure video muting doesn't affect audio).
enableRemoteAudio(gSecondConnection, false);
enableRemoteVideo(gSecondConnection, false);
setTimeout(function() {
enableRemoteAudio(gSecondConnection, true);
}, 500);
setTimeout(function() {
ensureAudioPlaying(gSecondConnection);
}, 1500);
});
}
function callAndEnsureLocalVideoMutingDoesntMuteAudio() {
callAndEnsureAudioIsPlaying({audio: true, video: true});
setAllEventsOccuredHandler(function() {
setAllEventsOccuredHandler(reportTestSuccess);
enableLocalVideo(gFirstConnection, false);
ensureAudioPlaying(gSecondConnection);
});
}
function callAndEnsureRemoteVideoMutingDoesntMuteAudio() {
callAndEnsureAudioIsPlaying({audio: true, video: true});
setAllEventsOccuredHandler(function() {
setAllEventsOccuredHandler(reportTestSuccess);
enableRemoteVideo(gSecondConnection, false);
ensureAudioPlaying(gSecondConnection);
});
}
function callAndEnsureVideoTrackMutingWorks() {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
addExpectedEvent();
detectVideoPlaying('remote-view-2', function() {
// Disable the receiver's remote media stream. Video should stop.
// (Also, ensure muting audio doesn't affect video).
enableRemoteVideo(gSecondConnection, false);
enableRemoteAudio(gSecondConnection, false);
detectVideoStopped('remote-view-2', function() {
// Video has stopped: unmute and succeed if it starts playing again.
enableRemoteVideo(gSecondConnection, true);
detectVideoPlaying('remote-view-2', eventOccured);
})
});
}
// Test call with a new Video MediaStream that has been created based on a
// stream generated by getUserMedia.
function callWithNewVideoMediaStream() {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
}
// Test call with a new Video MediaStream that has been created based on a
// stream generated by getUserMedia. When Video is flowing, an audio track
// is added to the sent stream and the video track is removed. This
// is to test that adding and removing of remote tracks on an existing
// mediastream works.
function callWithNewVideoMediaStreamLaterSwitchToAudio() {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
waitForVideo('remote-view-1');
waitForVideo('remote-view-2');
// Set an event handler for when video is playing.
setAllEventsOccuredHandler(function() {
// Add an audio track to the local stream and remove the video track and
// then renegotiate. But first - setup the expectations.
var localStream = gFirstConnection.getLocalStreams()[0];
var remoteStream1 = gFirstConnection.getRemoteStreams()[0];
// Add an expected event that onaddtrack will be called on the remote
// mediastream received on gFirstConnection when the audio track is
// received.
addExpectedEvent();
remoteStream1.onaddtrack = function(){
assertEquals(remoteStream1.getAudioTracks()[0].id,
localStream.getAudioTracks()[0].id);
eventOccured();
}
// Add an expectation that the received video track is removed from
// gFirstConnection.
addExpectedEvent();
remoteStream1.onremovetrack = function() {
eventOccured();
}
// Add an expected event that onaddtrack will be called on the remote
// mediastream received on gSecondConnection when the audio track is
// received.
remoteStream2 = gSecondConnection.getRemoteStreams()[0];
addExpectedEvent();
remoteStream2.onaddtrack = function() {
assertEquals(remoteStream2.getAudioTracks()[0].id,
localStream.getAudioTracks()[0].id);
eventOccured();
}
// Add an expectation that the received video track is removed from
// gSecondConnection.
addExpectedEvent();
remoteStream2.onremovetrack = function() {
eventOccured();
}
// When all the above events have occurred- the test pass.
setAllEventsOccuredHandler(reportTestSuccess);
localStream.addTrack(gLocalStream.getAudioTracks()[0]);
localStream.removeTrack(localStream.getVideoTracks()[0]);
negotiate();
});
}
// Loads this page inside itself using an iframe, and ensures we can make a
// successful getUserMedia + peerconnection call inside the iframe.
function callInsideIframe(constraints) {
runInsideIframe(function(iframe) {
// Run a regular webrtc call inside the iframe.
iframe.contentWindow.call(constraints);
});
}
// Func should accept an iframe as its first argument.
function runInsideIframe(func) {
var iframe = document.createElement('iframe');
document.body.appendChild(iframe);
iframe.onload = onIframeLoaded;
iframe.src = window.location;
function onIframeLoaded() {
var iframe = window.document.querySelector('iframe');
// Propagate test success out of the iframe.
iframe.contentWindow.setAllEventsOccuredHandler(
window.parent.reportTestSuccess);
func(iframe);
}
}
// This function is used for setting up a test that:
// 1. Creates a data channel on |gFirstConnection| and sends data to
// |gSecondConnection|.
// 2. When data is received on |gSecondConnection| a message
// is sent to |gFirstConnection|.
// 3. When data is received on |gFirstConnection|, the data
// channel is closed. The test passes when the state transition completes.
function setupDataChannel(params) {
var sendDataString = "send some text on a data channel."
firstDataChannel = gFirstConnection.createDataChannel(
"sendDataChannel", params);
assertEquals('connecting', firstDataChannel.readyState);
// When |firstDataChannel| transition to open state, send a text string.
firstDataChannel.onopen = function() {
assertEquals('open', firstDataChannel.readyState);
firstDataChannel.send(sendDataString);
}
// When |firstDataChannel| receive a message, close the channel and
// initiate a new offer/answer exchange to complete the closure.
firstDataChannel.onmessage = function(event) {
assertEquals(event.data, sendDataString);
firstDataChannel.close();
negotiate();
}
// When |firstDataChannel| transition to closed state, the test pass.
addExpectedEvent();
firstDataChannel.onclose = function() {
assertEquals('closed', firstDataChannel.readyState);
eventOccured();
}
// Event handler for when |gSecondConnection| receive a new dataChannel.
gSecondConnection.ondatachannel = function (event) {
// Make secondDataChannel global to make sure it's not gc'd.
secondDataChannel = event.channel;
// When |secondDataChannel| receive a message, send a message back.
secondDataChannel.onmessage = function(event) {
assertEquals(event.data, sendDataString);
console.log("gSecondConnection received data");
assertEquals('open', secondDataChannel.readyState);
secondDataChannel.send(sendDataString);
}
}
}
// SCTP data channel setup is slightly different then RTP based
// channels. Due to a bug in libjingle, we can't send data immediately
// after channel becomes open. So for that reason in SCTP,
// we are sending data from second channel, when ondatachannel event is
// received. So data flow happens 2 -> 1 -> 2.
function setupSctpDataChannel(params) {
var sendDataString = "send some text on a data channel."
firstDataChannel = gFirstConnection.createDataChannel(
"sendDataChannel", params);
assertEquals('connecting', firstDataChannel.readyState);
// When |firstDataChannel| transition to open state, send a text string.
firstDataChannel.onopen = function() {
assertEquals('open', firstDataChannel.readyState);
}
// When |firstDataChannel| receive a message, send message back.
// initiate a new offer/answer exchange to complete the closure.
firstDataChannel.onmessage = function(event) {
assertEquals('open', firstDataChannel.readyState);
assertEquals(event.data, sendDataString);
firstDataChannel.send(sendDataString);
}
// Event handler for when |gSecondConnection| receive a new dataChannel.
gSecondConnection.ondatachannel = function (event) {
// Make secondDataChannel global to make sure it's not gc'd.
secondDataChannel = event.channel;
secondDataChannel.onopen = function() {
secondDataChannel.send(sendDataString);
}
// When |secondDataChannel| receive a message, close the channel and
// initiate a new offer/answer exchange to complete the closure.
secondDataChannel.onmessage = function(event) {
assertEquals(event.data, sendDataString);
assertEquals('open', secondDataChannel.readyState);
secondDataChannel.close();
negotiate();
}
// When |secondDataChannel| transition to closed state, the test pass.
addExpectedEvent();
secondDataChannel.onclose = function() {
assertEquals('closed', secondDataChannel.readyState);
eventOccured();
}
}
}
// Test call with a stream that has been created by getUserMedia, clone
// the stream to a cloned stream, send them via the same peer connection.
function addTwoMediaStreamsToOneConnection() {
createConnections(null);
navigator.webkitGetUserMedia({audio: true, video: true},
cloneStreamAndAddTwoStreamsToOneConnection, printGetUserMediaError);
}
function onToneChange(tone) {
gSentTones += tone.tone;
}
function createConnections(constraints) {
gFirstConnection = createConnection(constraints, 'remote-view-1');
assertEquals('stable', gFirstConnection.signalingState);
gSecondConnection = createConnection(constraints, 'remote-view-2');
assertEquals('stable', gSecondConnection.signalingState);
}
function createConnection(constraints, remoteView) {
var pc = new webkitRTCPeerConnection(null, constraints);
pc.onaddstream = function(event) {
onRemoteStream(event, remoteView);
}
return pc;
}
function displayAndRemember(localStream) {
var localStreamUrl = URL.createObjectURL(localStream);
$('local-view').src = localStreamUrl;
gLocalStream = localStream;
}
// Called if getUserMedia fails.
function printGetUserMediaError(error) {
var message = 'getUserMedia request unexpectedly failed:';
if (error.constraintName)
message += ' could not satisfy constraint ' + error.constraintName;
else
message += ' devices not working/user denied access.';
failTest(message);
}
// Called if getUserMedia succeeds and we want to send from both connections.
function addStreamToBothConnectionsAndNegotiate(localStream) {
displayAndRemember(localStream);
gFirstConnection.addStream(localStream);
gSecondConnection.addStream(localStream);
negotiate();
}
// Called if getUserMedia succeeds when we want to send from one connection.
function addStreamToTheFirstConnectionAndNegotiate(localStream) {
displayAndRemember(localStream);
gFirstConnection.addStream(localStream);
negotiate();
}
function verifyHasOneAudioAndVideoTrack(stream) {
assertEquals(1, stream.getAudioTracks().length);
assertEquals(1, stream.getVideoTracks().length);
}
// Called if getUserMedia succeeds, then clone the stream, send two streams
// from one peer connection.
function cloneStreamAndAddTwoStreamsToOneConnection(localStream) {
displayAndRemember(localStream);
var clonedStream = null;
if (typeof localStream.clone === "function") {
clonedStream = localStream.clone();
} else {
clonedStream = new webkitMediaStream(localStream);
}
gFirstConnection.addStream(localStream);
gFirstConnection.addStream(clonedStream);
// Verify the local streams are correct.
assertEquals(2, gFirstConnection.getLocalStreams().length);
verifyHasOneAudioAndVideoTrack(gFirstConnection.getLocalStreams()[0]);
verifyHasOneAudioAndVideoTrack(gFirstConnection.getLocalStreams()[1]);
// The remote side should receive two streams. After that, verify the
// remote side has the correct number of streams and tracks.
addExpectedEvent();
addExpectedEvent();
gSecondConnection.onaddstream = function(event) {
eventOccured();
}
setAllEventsOccuredHandler(function() {
// Negotiation complete, verify remote streams on the receiving side.
assertEquals(2, gSecondConnection.getRemoteStreams().length);
verifyHasOneAudioAndVideoTrack(gSecondConnection.getRemoteStreams()[0]);
verifyHasOneAudioAndVideoTrack(gSecondConnection.getRemoteStreams()[1]);
reportTestSuccess();
});
negotiate();
}
// A new MediaStream is created with video track from |localStream| and is
// added to both peer connections.
function createNewVideoStreamAndAddToBothConnections(localStream) {
displayAndRemember(localStream);
var newStream = new webkitMediaStream();
newStream.addTrack(localStream.getVideoTracks()[0]);
gFirstConnection.addStream(newStream);
gSecondConnection.addStream(newStream);
negotiate();
}
function negotiate() {
negotiateBetween(gFirstConnection, gSecondConnection);
}
function negotiateBetween(caller, callee) {
console.log("Negotiating call...");
// Not stable = negotiation is ongoing. The behavior of re-negotiating while
// a negotiation is ongoing is more or less undefined, so avoid this.
if (caller.signalingState != 'stable' || callee.signalingState != 'stable')
throw 'You can only negotiate when the connection is stable!';
connectOnIceCandidate(caller, callee);
caller.createOffer(
function (offer) {
onOfferCreated(offer, caller, callee);
});
}
function iceCandidateIsLoopback(candidate) {
return candidate.candidate.indexOf("127.0.0.1") > -1 ||
candidate.candidate.indexOf("::1") > -1;
}
// Helper function to invoke |callback| when gathering is completed.
function gatherIceCandidates(pc, callback) {
var candidates = [];
pc.createDataChannel("");
pc.onicecandidate = function(event) {
// null candidate indicates the gathering has completed.
if (event.candidate == null) {
callback(candidates);
} else {
candidates.push(event.candidate);
}
}
pc.createOffer(
function(offer) {
pc.setLocalDescription(offer);
},
function(error) { failTest(error); }
);
}
function callWithDevicePermissionGranted() {
var pc = new webkitRTCPeerConnection(null, null);
gatherIceCandidates(pc, function(candidates) {
assertEquals(candidates.length > 0, true);
for (i = 0; i < candidates.length; i++) {
assertEquals(iceCandidateIsLoopback(candidates[i]), false);
}
reportTestSuccess();
});
}
function callWithDevicePermissionDeniedAndEmptyIceServers() {
var pc = new webkitRTCPeerConnection({iceServers:[]}, null);
gatherIceCandidates(pc, function(candidates) {
assertEquals(candidates.length, 0);
reportTestSuccess();
});
}
function callAndExpectLoopbackCandidates() {
var pc = new webkitRTCPeerConnection(null, null);
gatherIceCandidates(pc, function(candidates) {
assertEquals(candidates.length > 0, true);
for (i = 0; i < candidates.length; i++) {
assertEquals(iceCandidateIsLoopback(candidates[i]), true);
}
reportTestSuccess();
});
}
function callWithDevicePermissionDeniedAndUndefinedIceServers() {
callAndExpectLoopbackCandidates();
}
function callWithMultipleRoutesDisabledAndUndefinedIceServers() {
callAndExpectLoopbackCandidates();
}
function onOfferCreated(offer, caller, callee) {
offer.sdp = transformSdp(offer.sdp);
console.log('Offer:\n' + offer.sdp);
caller.setLocalDescription(offer, function() {
assertEquals('have-local-offer', caller.signalingState);
receiveOffer(offer.sdp, caller, callee);
}, onLocalDescriptionError);
}
function receiveOffer(offerSdp, caller, callee) {
console.log("Receiving offer...");
offerSdp = transformRemoteSdp(offerSdp);
var parsedOffer = new RTCSessionDescription({ type: 'offer',
sdp: offerSdp });
callee.setRemoteDescription(parsedOffer,
function() {
assertEquals('have-remote-offer',
callee.signalingState);
callee.createAnswer(
function (answer) {
onAnswerCreated(answer, caller, callee);
});
},
onRemoteDescriptionError);
}
function removeMsid(offerSdp) {
offerSdp = offerSdp.replace(/a=msid-semantic.*\r\n/g, '');
offerSdp = offerSdp.replace('a=mid:audio\r\n', '');
offerSdp = offerSdp.replace('a=mid:video\r\n', '');
offerSdp = offerSdp.replace(/a=ssrc.*\r\n/g, '');
return offerSdp;
}
function removeVideoCodec(offerSdp) {
offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
'a=rtpmap:100 XVP8/90000\r\n');
return offerSdp;
}
function removeCrypto(offerSdp) {
offerSdp = offerSdp.replace(/a=crypto.*\r\n/g, 'a=Xcrypto\r\n');
offerSdp = offerSdp.replace(/a=fingerprint.*\r\n/g, '');
return offerSdp;
}
function addBandwithControl(offerSdp) {
offerSdp = offerSdp.replace('a=mid:audio\r\n', 'a=mid:audio\r\n'+
'b=AS:16\r\n');
offerSdp = offerSdp.replace('a=mid:video\r\n', 'a=mid:video\r\n'+
'b=AS:512\r\n');
return offerSdp;
}
function removeBundle(sdp) {
return sdp.replace(/a=group:BUNDLE .*\r\n/g, '');
}
function useGice(sdp) {
sdp = sdp.replace(/t=.*\r\n/g, function(subString) {
return subString + 'a=ice-options:google-ice\r\n';
});
return sdp;
}
function useExternalSdes(sdp) {
// Remove current crypto specification.
sdp = sdp.replace(/a=crypto.*\r\n/g, '');
sdp = sdp.replace(/a=fingerprint.*\r\n/g, '');
// Add external crypto. This is not compatible with |removeMsid|.
sdp = sdp.replace(/a=mid:(\w+)\r\n/g, function(subString, group) {
return subString + EXTERNAL_SDES_LINES[group] + '\r\n';
});
return sdp;
}
function onAnswerCreated(answer, caller, callee) {
answer.sdp = transformSdp(answer.sdp);
console.log('Answer:\n' + answer.sdp);
callee.setLocalDescription(answer,
function () {
assertEquals('stable', callee.signalingState);
},
onLocalDescriptionError);
receiveAnswer(answer.sdp, caller);
}
function receiveAnswer(answerSdp, caller) {
console.log("Receiving answer...");
answerSdp = transformRemoteSdp(answerSdp);
var parsedAnswer = new RTCSessionDescription({ type: 'answer',
sdp: answerSdp });
caller.setRemoteDescription(parsedAnswer,
function() {
assertEquals('stable', caller.signalingState);
},
onRemoteDescriptionError);
}
function connectOnIceCandidate(caller, callee) {
caller.onicecandidate = function(event) { onIceCandidate(event, callee); }
callee.onicecandidate = function(event) { onIceCandidate(event, caller); }
}
function onIceCandidate(event, target) {
if (event.candidate) {
var candidate = new RTCIceCandidate(event.candidate);
target.addIceCandidate(candidate);
}
}
function onRemoteStream(e, target) {
console.log("Receiving remote stream...");
if (gTestWithoutMsid && e.stream.id != "default") {
failTest('a default remote stream was expected but instead ' +
e.stream.id + ' was received.');
}
gRemoteStreams[target] = e.stream;
var remoteStreamUrl = URL.createObjectURL(e.stream);
var remoteVideo = $(target);
remoteVideo.src = remoteStreamUrl;
}
</script>
</head>
<body>
<table border="0">
<tr>
<td><video width="320" height="240" id="local-view" style="display:none"
autoplay muted></video></td>
<td><video width="320" height="240" id="remote-view-1"
style="display:none" autoplay></video></td>
<td><video width="320" height="240" id="remote-view-2"
style="display:none" autoplay></video></td>
<td><video width="320" height="240" id="remote-view-3"
style="display:none" autoplay></video></td>
<td><video width="320" height="240" id="remote-view-4"
style="display:none" autoplay></video></td>
<!-- Canvases are named after their corresponding video elements. -->
<td><canvas width="320" height="240" id="remote-view-1-canvas"
style="display:none"></canvas></td>
<td><canvas width="320" height="240" id="remote-view-2-canvas"
style="display:none"></canvas></td>
<td><canvas width="320" height="240" id="remote-view-3-canvas"
style="display:none"></canvas></td>
<td><canvas width="320" height="240" id="remote-view-4-canvas"
style="display:none"></canvas></td>
</tr>
</table>
</body>
</html>
|
