content/test/data/media/peerconnection-call.html

   1 <html>
   2 <head>
   3   <script type="text/javascript" src="webrtc_test_utilities.js"></script>
   4   <script type="text/javascript" src="webrtc_test_audio.js"></script>
   5   <script type="text/javascript">
   6   $ = function(id) {
   7     return document.getElementById(id);
   8   };
   9
  10   window.onerror = function(errorMsg, url, lineNumber, column, errorObj) {
  11     failTest('Error: ' + errorMsg + '\nScript: ' + url +
  12              '\nLine: ' + lineNumber + '\nColumn: ' + column +
  13              '\nStackTrace: ' + errorObj);
  14   }
  15
  16   var gFirstConnection = null;
  17   var gSecondConnection = null;
  18   var gTestWithoutMsid = false;
  19   var gLocalStream = null;
  20   var gSentTones = '';
  21
  22   var gRemoteStreams = {};
  23
  24   // Default transform functions, overridden by some test cases.
  25   var transformSdp = function(sdp) { return sdp; };
  26   var transformRemoteSdp = function(sdp) { return sdp; };
  27   var onLocalDescriptionError = function(error) { failTest(error); };
  28   var onRemoteDescriptionError = function(error) { failTest(error); };
  29
  30   // When using external SDES, the crypto key is chosen by javascript.
  31   var EXTERNAL_SDES_LINES = {
  32     'audio': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
  33         'inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR',
  34     'video': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
  35         'inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj',
  36     'data': 'a=crypto:1 AES_CM_128_HMAC_SHA1_80 ' +
  37         'inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj'
  38   };
  39
  40   setAllEventsOccuredHandler(reportTestSuccess);
  41
  42   // Test that we can setup a call with an audio and video track (must request
  43   // video in this call since we expect video to be playing).
  44   function call(constraints) {
  45     createConnections(null);
  46     navigator.webkitGetUserMedia(constraints,
  47         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
  48     waitForVideo('remote-view-1');
  49     waitForVideo('remote-view-2');
  50   }
  51
  52   // Test that we can setup a call with a video track and that the remote peer
  53   // receives black frames if the local video track is disabled.
  54   function callAndDisableLocalVideo(constraints) {
  55     createConnections(null);
  56     navigator.webkitGetUserMedia(constraints,
  57         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
  58     detectVideoPlaying('remote-view-1',
  59         function () {
  60           assertEquals(gLocalStream.getVideoTracks().length, 1);
  61           gLocalStream.getVideoTracks()[0].enabled = false;
  62           waitForBlackVideo('remote-view-1');
  63         });
  64   }
  65
  66   // Test that we can setup call with an audio and video track and check that
  67   // the video resolution is as expected.
  68   function callAndExpectResolution(constraints,
  69                                    expected_width,
  70                                    expected_height) {
  71     createConnections(null);
  72     navigator.webkitGetUserMedia(constraints,
  73         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
  74     waitForVideoWithResolution('remote-view-1',
  75                                expected_width,
  76                                expected_height);
  77     waitForVideoWithResolution('remote-view-2',
  78                                expected_width,
  79                                expected_height);
  80   }
  81
  82
  83   // First calls without streams on any connections, and then adds a stream
  84   // to peer connection 1 which gets sent to peer connection 2. We must wait
  85   // for the first negotiation to complete before starting the second one, which
  86   // is why we wait until the connection is stable before re-negotiating.
  87   function callEmptyThenAddOneStreamAndRenegotiate(constraints) {
  88     createConnections(null);
  89     negotiate();
  90     waitForConnectionToStabilize(gFirstConnection, function() {
  91       navigator.webkitGetUserMedia(constraints,
  92           addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
  93       // Only the first connection is sending here.
  94       waitForVideo('remote-view-2');
  95     });
  96   }
  97
  98   // The second set of constraints should request video (e.g. video:true) since
  99   // we expect video to be playing after the second renegotiation.
 100   function callAndRenegotiateToVideo(constraints, renegotiationConstraints) {
 101     createConnections(null);
 102     navigator.webkitGetUserMedia(constraints,
 103         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 104
 105     waitForConnectionToStabilize(gFirstConnection, function() {
 106       gFirstConnection.removeStream(gLocalStream);
 107       gSecondConnection.removeStream(gLocalStream);
 108
 109       navigator.webkitGetUserMedia(renegotiationConstraints,
 110           addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 111       waitForVideo('remote-view-1');
 112       waitForVideo('remote-view-2');
 113     });
 114   }
 115
 116   // The second set of constraints should request audio (e.g. audio:true) since
 117   // we expect audio to be playing after the second renegotiation.
 118   function callAndRenegotiateToAudio(constraints, renegotiationConstraints) {
 119     createConnections(null);
 120     navigator.webkitGetUserMedia(constraints,
 121         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 122
 123     waitForConnectionToStabilize(gFirstConnection, function() {
 124       gFirstConnection.removeStream(gLocalStream);
 125       gSecondConnection.removeStream(gLocalStream);
 126
 127       navigator.webkitGetUserMedia(renegotiationConstraints,
 128           addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
 129
 130       var onCallEstablished = function() {
 131         ensureAudioPlaying(gSecondConnection);
 132       };
 133
 134       waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
 135     });
 136   }
 137
 138   // First makes a call between pc1 and pc2 where a stream is sent from pc1 to
 139   // pc2. The stream sent from pc1 to pc2 is cloned from the stream received on
 140   // pc2  to test that cloning of remote video tracks works as intended and is
 141   // sent back to pc1.
 142   function callAndForwardRemoteStream(constraints) {
 143     createConnections(null);
 144     navigator.webkitGetUserMedia(constraints,
 145                                  addStreamToTheFirstConnectionAndNegotiate,
 146                                  printGetUserMediaError);
 147     var onRemoteStream2 = function() {
 148       // Video has been detected to be playing in pc2. Clone the received
 149       // stream and send it back to pc1.
 150       gSecondConnection.addStream(gRemoteStreams['remote-view-2'].clone());
 151       negotiate();
 152     }
 153
 154     // Wait for remove video to be playing in pc2. Once video is playing,
 155     // forward the remove stream from pc2 to pc1.
 156     detectVideoPlaying('remote-view-2', onRemoteStream2);
 157
 158     // Wait for video to be forwarded back to connection 1.
 159     waitForVideo('remote-view-1');
 160   }
 161
 162   // First makes a call between pc1 and pc2, and then construct a new media
 163   // stream using the remote audio and video tracks, connect the new media
 164   // stream to a video element. These operations should not crash Chrome.
 165   function ConnectChromiumSinkToRemoteAudioTrack() {
 166     createConnections(null);
 167     navigator.webkitGetUserMedia({audio: true, video: true},
 168                                  addStreamToBothConnectionsAndNegotiate,
 169                                  printGetUserMediaError);
 170
 171     detectVideoPlaying('remote-view-2', function() {
 172       // Construct a new media stream with remote tracks.
 173       var newStream = new webkitMediaStream();
 174       newStream.addTrack(
 175           gSecondConnection.getRemoteStreams()[0].getAudioTracks()[0]);
 176       newStream.addTrack(
 177           gSecondConnection.getRemoteStreams()[0].getVideoTracks()[0]);
 178       var videoElement = document.createElement('video');
 179
 180       // No crash for this operation.
 181       videoElement.src = URL.createObjectURL(newStream);
 182       waitForVideo('remote-view-2');
 183     });
 184   }
 185
 186   // Test that we can setup call with an audio and video track and
 187   // simulate that the remote peer don't support MSID.
 188   function callWithoutMsidAndBundle() {
 189     createConnections(null);
 190     transformSdp = removeBundle;
 191     transformRemoteSdp = removeMsid;
 192     gTestWithoutMsid = true;
 193     navigator.webkitGetUserMedia({audio: true, video: true},
 194         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 195     waitForVideo('remote-view-1');
 196     waitForVideo('remote-view-2');
 197   }
 198
 199   // Test that we can't setup a call with an unsupported video codec
 200   function negotiateUnsupportedVideoCodec() {
 201     createConnections(null);
 202     transformSdp = removeVideoCodec;
 203
 204     onLocalDescriptionError = function(error) {
 205       var expectedMsg = 'Failed to set local offer sdp:' +
 206           ' Session error code: ERROR_CONTENT. Session error description:' +
 207           ' Failed to set video receive codecs..';
 208       assertEquals(expectedMsg, error);
 209       reportTestSuccess();
 210     };
 211     navigator.webkitGetUserMedia({audio: true, video: true},
 212         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 213   }
 214
 215   // Test that we can't setup a call if one peer does not support encryption
 216   function negotiateNonCryptoCall() {
 217     createConnections(null);
 218     transformSdp = removeCrypto;
 219     onLocalDescriptionError = function(error) {
 220       var expectedMsg = 'Failed to set local offer sdp:' +
 221           ' Called with SDP without DTLS fingerprint.';
 222
 223       assertEquals(expectedMsg, error);
 224       reportTestSuccess();
 225     };
 226     navigator.webkitGetUserMedia({audio: true, video: true},
 227         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 228   }
 229
 230   // Test that we can negotiate a call with an SDP offer that includes a
 231   // b=AS:XX line to control audio and video bandwidth
 232   function negotiateOfferWithBLine() {
 233     createConnections(null);
 234     transformSdp = addBandwithControl;
 235     navigator.webkitGetUserMedia({audio: true, video: true},
 236         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 237     waitForVideo('remote-view-1');
 238     waitForVideo('remote-view-2');
 239   }
 240
 241   // Test that we can setup call with legacy settings.
 242   function callWithLegacySdp() {
 243     transformSdp = function(sdp) {
 244       return removeBundle(useGice(useExternalSdes(sdp)));
 245     };
 246     createConnections({
 247       'mandatory': {'RtpDataChannels': true, 'DtlsSrtpKeyAgreement': false}
 248     });
 249     setupDataChannel({reliable: false});
 250     navigator.webkitGetUserMedia({audio: true, video: true},
 251         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 252     waitForVideo('remote-view-1');
 253     waitForVideo('remote-view-2');
 254   }
 255
 256   // Test only a data channel.
 257   function callWithDataOnly() {
 258     createConnections({optional:[{RtpDataChannels: true}]});
 259     setupDataChannel({reliable: false});
 260     negotiate();
 261   }
 262
 263   function callWithSctpDataOnly() {
 264     createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
 265     setupSctpDataChannel({reliable: true});
 266     negotiate();
 267   }
 268
 269   // Test call with audio, video and a data channel.
 270   function callWithDataAndMedia() {
 271     createConnections({optional:[{RtpDataChannels: true}]});
 272     setupDataChannel({reliable: false});
 273     navigator.webkitGetUserMedia({audio: true, video: true},
 274       addStreamToBothConnectionsAndNegotiate,
 275       printGetUserMediaError);
 276     waitForVideo('remote-view-1');
 277     waitForVideo('remote-view-2');
 278   }
 279
 280   function callWithSctpDataAndMedia() {
 281     createConnections({optional: [{DtlsSrtpKeyAgreement: true}]});
 282     setupSctpDataChannel({reliable: true});
 283     navigator.webkitGetUserMedia({audio: true, video: true},
 284       addStreamToBothConnectionsAndNegotiate,
 285       printGetUserMediaError);
 286     waitForVideo('remote-view-1');
 287     waitForVideo('remote-view-2');
 288   }
 289
 290   // Test call with a data channel and later add audio and video.
 291   function callWithDataAndLaterAddMedia() {
 292     createConnections({optional:[{RtpDataChannels: true}]});
 293     setupDataChannel({reliable: false});
 294     negotiate();
 295
 296     // Set an event handler for when the data channel has been closed.
 297     setAllEventsOccuredHandler(function() {
 298       // When the video is flowing the test is done.
 299       setAllEventsOccuredHandler(reportTestSuccess);
 300       navigator.webkitGetUserMedia({audio: true, video: true},
 301         addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 302       waitForVideo('remote-view-1');
 303       waitForVideo('remote-view-2');
 304     });
 305   }
 306
 307   // Test that we can setup call and send DTMF.
 308   function callAndSendDtmf(tones) {
 309     createConnections(null);
 310     navigator.webkitGetUserMedia({audio: true, video: true},
 311       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 312     var onCallEstablished = function() {
 313       // Send DTMF tones. Allocate the sender in the window to keep it from
 314       // being garbage collected. https://crbug.com/486654.
 315       var localAudioTrack = gLocalStream.getAudioTracks()[0];
 316       window.dtmfSender = gFirstConnection.createDTMFSender(localAudioTrack);
 317       window.dtmfSender.ontonechange = onToneChange;
 318       window.dtmfSender.insertDTMF(tones);
 319       // Wait for the DTMF tones callback.
 320       addExpectedEvent();
 321       var waitDtmf = setInterval(function() {
 322         if (gSentTones == tones) {
 323           clearInterval(waitDtmf);
 324           eventOccured();
 325         }
 326       }, 100);
 327     }
 328
 329     // Do the DTMF test after we have received video.
 330     detectVideoPlaying('remote-view-2', onCallEstablished);
 331   }
 332
 333   function testCreateOfferOptions() {
 334     createConnections(null);
 335     var offerOptions = {
 336       'offerToReceiveAudio': false,
 337       'offerToReceiveVideo': true
 338     };
 339
 340     gFirstConnection.createOffer(
 341         function(offer) {
 342           assertEquals(-1, offer.sdp.search('m=audio'));
 343           assertNotEquals(-1, offer.sdp.search('m=video'));
 344
 345           reportTestSuccess();
 346         },
 347         function(error) { failTest(error); },
 348         offerOptions);
 349   }
 350
 351   function callAndEnsureAudioIsPlaying(constraints) {
 352     createConnections(null);
 353
 354     // Add the local stream to gFirstConnection to play one-way audio.
 355     navigator.webkitGetUserMedia(constraints,
 356       addStreamToTheFirstConnectionAndNegotiate, printGetUserMediaError);
 357
 358     var onCallEstablished = function() {
 359       ensureAudioPlaying(gSecondConnection);
 360     };
 361
 362     waitForConnectionToStabilize(gFirstConnection, onCallEstablished);
 363   }
 364
 365   function callWithIsac16KAndEnsureAudioIsPlaying(constraints) {
 366     transformSdp = function(sdp) {
 367       sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g,
 368                         'm=audio $1 RTP/SAVPF 103 126\r\n');
 369       sdp = sdp.replace('a=fmtp:111 minptime=10', 'a=fmtp:103 minptime=10');
 370       if (sdp.search('a=rtpmap:103 ISAC/16000') == -1)
 371         failTest('Missing iSAC 16K codec on Android; cannot force codec.');
 372
 373       return sdp;
 374     }
 375     callAndEnsureAudioIsPlaying(constraints);
 376   }
 377
 378   function enableRemoteVideo(peerConnection, enabled) {
 379     remoteStream = peerConnection.getRemoteStreams()[0];
 380     remoteStream.getVideoTracks()[0].enabled = enabled;
 381   }
 382
 383   function enableRemoteAudio(peerConnection, enabled) {
 384     remoteStream = peerConnection.getRemoteStreams()[0];
 385     remoteStream.getAudioTracks()[0].enabled = enabled;
 386   }
 387
 388   function enableLocalVideo(peerConnection, enabled) {
 389     localStream = peerConnection.getLocalStreams()[0];
 390     localStream.getVideoTracks()[0].enabled = enabled;
 391   }
 392
 393   function enableLocalAudio(peerConnection, enabled) {
 394     localStream = peerConnection.getLocalStreams()[0];
 395     localStream.getAudioTracks()[0].enabled = enabled;
 396   }
 397
 398   function callAndEnsureRemoteAudioTrackMutingWorks() {
 399     callAndEnsureAudioIsPlaying({audio: true, video: true});
 400     setAllEventsOccuredHandler(function() {
 401       setAllEventsOccuredHandler(reportTestSuccess);
 402
 403       // Call is up, now mute the remote track and check we stop playing out
 404       // audio (after a small delay, we don't expect it to happen instantly).
 405       enableRemoteAudio(gSecondConnection, false);
 406       ensureSilence(gSecondConnection);
 407     });
 408   }
 409
 410   function callAndEnsureLocalAudioTrackMutingWorks() {
 411     callAndEnsureAudioIsPlaying({audio: true, video: true});
 412     setAllEventsOccuredHandler(function() {
 413       setAllEventsOccuredHandler(reportTestSuccess);
 414
 415       // Call is up, now mute the local track of the sending side and ensure
 416       // the receiving side stops receiving audio.
 417       enableLocalAudio(gFirstConnection, false);
 418       ensureSilence(gSecondConnection);
 419     });
 420   }
 421
 422   function callAndEnsureAudioTrackUnmutingWorks() {
 423     callAndEnsureAudioIsPlaying({audio: true, video: true});
 424     setAllEventsOccuredHandler(function() {
 425       setAllEventsOccuredHandler(reportTestSuccess);
 426
 427       // Mute, wait a while, unmute, verify audio gets back up.
 428       // (Also, ensure video muting doesn't affect audio).
 429       enableRemoteAudio(gSecondConnection, false);
 430       enableRemoteVideo(gSecondConnection, false);
 431
 432       setTimeout(function() {
 433         enableRemoteAudio(gSecondConnection, true);
 434       }, 500);
 435
 436       setTimeout(function() {
 437         ensureAudioPlaying(gSecondConnection);
 438       }, 1500);
 439     });
 440   }
 441
 442   function callAndEnsureLocalVideoMutingDoesntMuteAudio() {
 443     callAndEnsureAudioIsPlaying({audio: true, video: true});
 444     setAllEventsOccuredHandler(function() {
 445       setAllEventsOccuredHandler(reportTestSuccess);
 446       enableLocalVideo(gFirstConnection, false);
 447       ensureAudioPlaying(gSecondConnection);
 448     });
 449   }
 450
 451   function callAndEnsureRemoteVideoMutingDoesntMuteAudio() {
 452     callAndEnsureAudioIsPlaying({audio: true, video: true});
 453     setAllEventsOccuredHandler(function() {
 454       setAllEventsOccuredHandler(reportTestSuccess);
 455       enableRemoteVideo(gSecondConnection, false);
 456       ensureAudioPlaying(gSecondConnection);
 457     });
 458   }
 459
 460   function callAndEnsureVideoTrackMutingWorks() {
 461     createConnections(null);
 462     navigator.webkitGetUserMedia({audio: true, video: true},
 463       addStreamToBothConnectionsAndNegotiate, printGetUserMediaError);
 464
 465     addExpectedEvent();
 466     detectVideoPlaying('remote-view-2', function() {
 467       // Disable the receiver's remote media stream. Video should stop.
 468       // (Also, ensure muting audio doesn't affect video).
 469       enableRemoteVideo(gSecondConnection, false);
 470       enableRemoteAudio(gSecondConnection, false);
 471
 472       detectVideoStopped('remote-view-2', function() {
 473         // Video has stopped: unmute and succeed if it starts playing again.
 474         enableRemoteVideo(gSecondConnection, true);
 475         detectVideoPlaying('remote-view-2', eventOccured);
 476       })
 477     });
 478   }
 479
 480   // Test call with a new Video MediaStream that has been created based on a
 481   // stream generated by getUserMedia.
 482   function callWithNewVideoMediaStream() {
 483     createConnections(null);
 484     navigator.webkitGetUserMedia({audio: true, video: true},
 485         createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
 486     waitForVideo('remote-view-1');
 487     waitForVideo('remote-view-2');
 488   }
 489
 490   // Test call with a new Video MediaStream that has been created based on a
 491   // stream generated by getUserMedia. When Video is flowing, an audio track
 492   // is added to the sent stream and the video track is removed. This
 493   // is to test that adding and removing of remote tracks on an existing
 494   // mediastream works.
 495   function callWithNewVideoMediaStreamLaterSwitchToAudio() {
 496     createConnections(null);
 497     navigator.webkitGetUserMedia({audio: true, video: true},
 498         createNewVideoStreamAndAddToBothConnections, printGetUserMediaError);
 499
 500     waitForVideo('remote-view-1');
 501     waitForVideo('remote-view-2');
 502
 503     // Set an event handler for when video is playing.
 504     setAllEventsOccuredHandler(function() {
 505       // Add an audio track to the local stream and remove the video track and
 506       // then renegotiate. But first - setup the expectations.
 507       var localStream = gFirstConnection.getLocalStreams()[0];
 508       var remoteStream1 = gFirstConnection.getRemoteStreams()[0];
 509
 510       // Add an expected event that onaddtrack will be called on the remote
 511       // mediastream received on gFirstConnection when the audio track is
 512       // received.
 513       addExpectedEvent();
 514       remoteStream1.onaddtrack = function(){
 515         assertEquals(remoteStream1.getAudioTracks()[0].id,
 516                      localStream.getAudioTracks()[0].id);
 517         eventOccured();
 518       }
 519
 520       // Add an expectation that the received video track is removed from
 521       // gFirstConnection.
 522       addExpectedEvent();
 523       remoteStream1.onremovetrack = function() {
 524         eventOccured();
 525       }
 526
 527       // Add an expected event that onaddtrack will be called on the remote
 528       // mediastream received on gSecondConnection when the audio track is
 529       // received.
 530       remoteStream2 = gSecondConnection.getRemoteStreams()[0];
 531       addExpectedEvent();
 532       remoteStream2.onaddtrack = function() {
 533         assertEquals(remoteStream2.getAudioTracks()[0].id,
 534                      localStream.getAudioTracks()[0].id);
 535         eventOccured();
 536       }
 537
 538       // Add an expectation that the received video track is removed from
 539       // gSecondConnection.
 540       addExpectedEvent();
 541       remoteStream2.onremovetrack = function() {
 542         eventOccured();
 543       }
 544       // When all the above events have occurred- the test pass.
 545       setAllEventsOccuredHandler(reportTestSuccess);
 546
 547       localStream.addTrack(gLocalStream.getAudioTracks()[0]);
 548       localStream.removeTrack(localStream.getVideoTracks()[0]);
 549       negotiate();
 550     });
 551   }
 552
 553   // Loads this page inside itself using an iframe, and ensures we can make a
 554   // successful getUserMedia + peerconnection call inside the iframe.
 555   function callInsideIframe(constraints) {
 556     runInsideIframe(function(iframe) {
 557       // Run a regular webrtc call inside the iframe.
 558       iframe.contentWindow.call(constraints);
 559     });
 560   }
 561
 562   // Func should accept an iframe as its first argument.
 563   function runInsideIframe(func) {
 564     var iframe = document.createElement('iframe');
 565     document.body.appendChild(iframe);
 566     iframe.onload = onIframeLoaded;
 567     iframe.src = window.location;
 568
 569     function onIframeLoaded() {
 570       var iframe = window.document.querySelector('iframe');
 571
 572       // Propagate test success out of the iframe.
 573       iframe.contentWindow.setAllEventsOccuredHandler(
 574           window.parent.reportTestSuccess);
 575
 576       func(iframe);
 577     }
 578   }
 579
 580   // This function is used for setting up a test that:
 581   // 1. Creates a data channel on |gFirstConnection| and sends data to
 582   //    |gSecondConnection|.
 583   // 2. When data is received on |gSecondConnection| a message
 584   //    is sent to |gFirstConnection|.
 585   // 3. When data is received on |gFirstConnection|, the data
 586   //    channel is closed. The test passes when the state transition completes.
 587   function setupDataChannel(params) {
 588     var sendDataString = "send some text on a data channel."
 589     firstDataChannel = gFirstConnection.createDataChannel(
 590         "sendDataChannel", params);
 591     assertEquals('connecting', firstDataChannel.readyState);
 592
 593     // When |firstDataChannel| transition to open state, send a text string.
 594     firstDataChannel.onopen = function() {
 595       assertEquals('open', firstDataChannel.readyState);
 596       firstDataChannel.send(sendDataString);
 597     }
 598
 599     // When |firstDataChannel| receive a message, close the channel and
 600     // initiate a new offer/answer exchange to complete the closure.
 601     firstDataChannel.onmessage = function(event) {
 602       assertEquals(event.data, sendDataString);
 603       firstDataChannel.close();
 604       negotiate();
 605     }
 606
 607     // When |firstDataChannel| transition to closed state, the test pass.
 608     addExpectedEvent();
 609     firstDataChannel.onclose = function() {
 610       assertEquals('closed', firstDataChannel.readyState);
 611       eventOccured();
 612     }
 613
 614     // Event handler for when |gSecondConnection| receive a new dataChannel.
 615     gSecondConnection.ondatachannel = function (event) {
 616       // Make secondDataChannel global to make sure it's not gc'd.
 617       secondDataChannel = event.channel;
 618
 619       // When |secondDataChannel| receive a message, send a message back.
 620       secondDataChannel.onmessage = function(event) {
 621         assertEquals(event.data, sendDataString);
 622         console.log("gSecondConnection received data");
 623         assertEquals('open', secondDataChannel.readyState);
 624         secondDataChannel.send(sendDataString);
 625       }
 626     }
 627   }
 628
 629   // SCTP data channel setup is slightly different then RTP based
 630   // channels. Due to a bug in libjingle, we can't send data immediately
 631   // after channel becomes open. So for that reason in SCTP,
 632   // we are sending data from second channel, when ondatachannel event is
 633   // received. So data flow happens 2 -> 1 -> 2.
 634   function setupSctpDataChannel(params) {
 635     var sendDataString = "send some text on a data channel."
 636     firstDataChannel = gFirstConnection.createDataChannel(
 637         "sendDataChannel", params);
 638     assertEquals('connecting', firstDataChannel.readyState);
 639
 640     // When |firstDataChannel| transition to open state, send a text string.
 641     firstDataChannel.onopen = function() {
 642       assertEquals('open', firstDataChannel.readyState);
 643     }
 644
 645     // When |firstDataChannel| receive a message, send message back.
 646     // initiate a new offer/answer exchange to complete the closure.
 647     firstDataChannel.onmessage = function(event) {
 648       assertEquals('open', firstDataChannel.readyState);
 649       assertEquals(event.data, sendDataString);
 650       firstDataChannel.send(sendDataString);
 651     }
 652
 653     // Event handler for when |gSecondConnection| receive a new dataChannel.
 654     gSecondConnection.ondatachannel = function (event) {
 655       // Make secondDataChannel global to make sure it's not gc'd.
 656       secondDataChannel = event.channel;
 657       secondDataChannel.onopen = function() {
 658         secondDataChannel.send(sendDataString);
 659       }
 660
 661       // When |secondDataChannel| receive a message, close the channel and
 662       // initiate a new offer/answer exchange to complete the closure.
 663       secondDataChannel.onmessage = function(event) {
 664         assertEquals(event.data, sendDataString);
 665         assertEquals('open', secondDataChannel.readyState);
 666         secondDataChannel.close();
 667         negotiate();
 668       }
 669
 670       // When |secondDataChannel| transition to closed state, the test pass.
 671       addExpectedEvent();
 672       secondDataChannel.onclose = function() {
 673         assertEquals('closed', secondDataChannel.readyState);
 674         eventOccured();
 675       }
 676     }
 677   }
 678
 679   // Test call with a stream that has been created by getUserMedia, clone
 680   // the stream to a cloned stream, send them via the same peer connection.
 681   function addTwoMediaStreamsToOneConnection() {
 682     createConnections(null);
 683     navigator.webkitGetUserMedia({audio: true, video: true},
 684         cloneStreamAndAddTwoStreamsToOneConnection, printGetUserMediaError);
 685   }
 686
 687   function onToneChange(tone) {
 688     gSentTones += tone.tone;
 689   }
 690
 691   function createConnections(constraints) {
 692     gFirstConnection = createConnection(constraints, 'remote-view-1');
 693     assertEquals('stable', gFirstConnection.signalingState);
 694
 695     gSecondConnection = createConnection(constraints, 'remote-view-2');
 696     assertEquals('stable', gSecondConnection.signalingState);
 697   }
 698
 699   function createConnection(constraints, remoteView) {
 700     var pc = new webkitRTCPeerConnection(null, constraints);
 701     pc.onaddstream = function(event) {
 702       onRemoteStream(event, remoteView);
 703     }
 704     return pc;
 705   }
 706
 707   function displayAndRemember(localStream) {
 708     var localStreamUrl = URL.createObjectURL(localStream);
 709     $('local-view').src = localStreamUrl;
 710
 711     gLocalStream = localStream;
 712   }
 713
 714   // Called if getUserMedia fails.
 715   function printGetUserMediaError(error) {
 716     var message = 'getUserMedia request unexpectedly failed:';
 717     if (error.constraintName)
 718       message += ' could not satisfy constraint ' + error.constraintName;
 719     else
 720       message += ' devices not working/user denied access.';
 721     failTest(message);
 722   }
 723
 724   // Called if getUserMedia succeeds and we want to send from both connections.
 725   function addStreamToBothConnectionsAndNegotiate(localStream) {
 726     displayAndRemember(localStream);
 727     gFirstConnection.addStream(localStream);
 728     gSecondConnection.addStream(localStream);
 729     negotiate();
 730   }
 731
 732   // Called if getUserMedia succeeds when we want to send from one connection.
 733   function addStreamToTheFirstConnectionAndNegotiate(localStream) {
 734     displayAndRemember(localStream);
 735     gFirstConnection.addStream(localStream);
 736     negotiate();
 737   }
 738
 739   function verifyHasOneAudioAndVideoTrack(stream) {
 740     assertEquals(1, stream.getAudioTracks().length);
 741     assertEquals(1, stream.getVideoTracks().length);
 742   }
 743
 744   // Called if getUserMedia succeeds, then clone the stream, send two streams
 745   // from one peer connection.
 746   function cloneStreamAndAddTwoStreamsToOneConnection(localStream) {
 747     displayAndRemember(localStream);
 748
 749     var clonedStream = null;
 750     if (typeof localStream.clone === "function") {
 751       clonedStream = localStream.clone();
 752     } else {
 753       clonedStream = new webkitMediaStream(localStream);
 754     }
 755
 756     gFirstConnection.addStream(localStream);
 757     gFirstConnection.addStream(clonedStream);
 758
 759     // Verify the local streams are correct.
 760     assertEquals(2, gFirstConnection.getLocalStreams().length);
 761     verifyHasOneAudioAndVideoTrack(gFirstConnection.getLocalStreams()[0]);
 762     verifyHasOneAudioAndVideoTrack(gFirstConnection.getLocalStreams()[1]);
 763
 764     // The remote side should receive two streams. After that, verify the
 765     // remote side has the correct number of streams and tracks.
 766     addExpectedEvent();
 767     addExpectedEvent();
 768     gSecondConnection.onaddstream = function(event) {
 769       eventOccured();
 770     }
 771     setAllEventsOccuredHandler(function() {
 772       // Negotiation complete, verify remote streams on the receiving side.
 773       assertEquals(2, gSecondConnection.getRemoteStreams().length);
 774       verifyHasOneAudioAndVideoTrack(gSecondConnection.getRemoteStreams()[0]);
 775       verifyHasOneAudioAndVideoTrack(gSecondConnection.getRemoteStreams()[1]);
 776
 777       reportTestSuccess();
 778     });
 779
 780     negotiate();
 781   }
 782
 783   // A new MediaStream is created with video track from |localStream| and is
 784   // added to both peer connections.
 785   function createNewVideoStreamAndAddToBothConnections(localStream) {
 786     displayAndRemember(localStream);
 787     var newStream = new webkitMediaStream();
 788     newStream.addTrack(localStream.getVideoTracks()[0]);
 789     gFirstConnection.addStream(newStream);
 790     gSecondConnection.addStream(newStream);
 791     negotiate();
 792   }
 793
 794   function negotiate() {
 795     negotiateBetween(gFirstConnection, gSecondConnection);
 796   }
 797
 798   function negotiateBetween(caller, callee) {
 799     console.log("Negotiating call...");
 800     // Not stable = negotiation is ongoing. The behavior of re-negotiating while
 801     // a negotiation is ongoing is more or less undefined, so avoid this.
 802     if (caller.signalingState != 'stable' || callee.signalingState != 'stable')
 803       throw 'You can only negotiate when the connection is stable!';
 804
 805     connectOnIceCandidate(caller, callee);
 806
 807     caller.createOffer(
 808         function (offer) {
 809           onOfferCreated(offer, caller, callee);
 810         });
 811   }
 812
 813   function onOfferCreated(offer, caller, callee) {
 814     offer.sdp = transformSdp(offer.sdp);
 815     console.log('Offer:\n' + offer.sdp);
 816     caller.setLocalDescription(offer, function() {
 817       assertEquals('have-local-offer', caller.signalingState);
 818       receiveOffer(offer.sdp, caller, callee);
 819     }, onLocalDescriptionError);
 820   }
 821
 822   function receiveOffer(offerSdp, caller, callee) {
 823     console.log("Receiving offer...");
 824     offerSdp = transformRemoteSdp(offerSdp);
 825
 826     var parsedOffer = new RTCSessionDescription({ type: 'offer',
 827                                                   sdp: offerSdp });
 828     callee.setRemoteDescription(parsedOffer,
 829                                 function() {
 830                                   assertEquals('have-remote-offer',
 831                                                callee.signalingState);
 832                                   callee.createAnswer(
 833                                     function (answer) {
 834                                       onAnswerCreated(answer, caller, callee);
 835                                     });
 836                                 },
 837                                 onRemoteDescriptionError);
 838   }
 839
 840   function removeMsid(offerSdp) {
 841     offerSdp = offerSdp.replace(/a=msid-semantic.*\r\n/g, '');
 842     offerSdp = offerSdp.replace('a=mid:audio\r\n', '');
 843     offerSdp = offerSdp.replace('a=mid:video\r\n', '');
 844     offerSdp = offerSdp.replace(/a=ssrc.*\r\n/g, '');
 845     return offerSdp;
 846   }
 847
 848   function removeVideoCodec(offerSdp) {
 849     offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
 850                                 'a=rtpmap:100 XVP8/90000\r\n');
 851     return offerSdp;
 852   }
 853
 854   function removeCrypto(offerSdp) {
 855     offerSdp = offerSdp.replace(/a=crypto.*\r\n/g, 'a=Xcrypto\r\n');
 856     offerSdp = offerSdp.replace(/a=fingerprint.*\r\n/g, '');
 857     return offerSdp;
 858   }
 859
 860   function addBandwithControl(offerSdp) {
 861     offerSdp = offerSdp.replace('a=mid:audio\r\n', 'a=mid:audio\r\n'+
 862                                 'b=AS:16\r\n');
 863     offerSdp = offerSdp.replace('a=mid:video\r\n', 'a=mid:video\r\n'+
 864                                 'b=AS:512\r\n');
 865     return offerSdp;
 866   }
 867
 868   function removeBundle(sdp) {
 869     return sdp.replace(/a=group:BUNDLE .*\r\n/g, '');
 870   }
 871
 872   function useGice(sdp) {
 873     sdp = sdp.replace(/t=.*\r\n/g, function(subString) {
 874       return subString + 'a=ice-options:google-ice\r\n';
 875     });
 876     return sdp;
 877   }
 878
 879   function useExternalSdes(sdp) {
 880     // Remove current crypto specification.
 881     sdp = sdp.replace(/a=crypto.*\r\n/g, '');
 882     sdp = sdp.replace(/a=fingerprint.*\r\n/g, '');
 883     // Add external crypto.  This is not compatible with |removeMsid|.
 884     sdp = sdp.replace(/a=mid:(\w+)\r\n/g, function(subString, group) {
 885       return subString + EXTERNAL_SDES_LINES[group] + '\r\n';
 886     });
 887     return sdp;
 888   }
 889
 890   function onAnswerCreated(answer, caller, callee) {
 891     answer.sdp = transformSdp(answer.sdp);
 892     console.log('Answer:\n' + answer.sdp);
 893     callee.setLocalDescription(answer,
 894                                function () {
 895                                  assertEquals('stable', callee.signalingState);
 896                                },
 897                                onLocalDescriptionError);
 898     receiveAnswer(answer.sdp, caller);
 899   }
 900
 901   function receiveAnswer(answerSdp, caller) {
 902     console.log("Receiving answer...");
 903     answerSdp = transformRemoteSdp(answerSdp);
 904     var parsedAnswer = new RTCSessionDescription({ type: 'answer',
 905                                                    sdp: answerSdp });
 906     caller.setRemoteDescription(parsedAnswer,
 907                                 function() {
 908                                   assertEquals('stable', caller.signalingState);
 909                                 },
 910                                 onRemoteDescriptionError);
 911   }
 912
 913   function connectOnIceCandidate(caller, callee) {
 914     caller.onicecandidate = function(event) { onIceCandidate(event, callee); }
 915     callee.onicecandidate = function(event) { onIceCandidate(event, caller); }
 916   }
 917
 918   function onIceCandidate(event, target) {
 919     if (event.candidate) {
 920       var candidate = new RTCIceCandidate(event.candidate);
 921       target.addIceCandidate(candidate);
 922     }
 923   }
 924
 925   function onRemoteStream(e, target) {
 926     console.log("Receiving remote stream...");
 927     if (gTestWithoutMsid && e.stream.id != "default") {
 928       failTest('a default remote stream was expected but instead ' +
 929           e.stream.id + ' was received.');
 930     }
 931     gRemoteStreams[target] = e.stream;
 932     var remoteStreamUrl = URL.createObjectURL(e.stream);
 933     var remoteVideo = $(target);
 934     remoteVideo.src = remoteStreamUrl;
 935   }
 936
 937   </script>
 938 </head>
 939 <body>
 940   <table border="0">
 941     <tr>
 942       <td><video width="320" height="240" id="local-view" style="display:none"
 943           autoplay muted></video></td>
 944       <td><video width="320" height="240" id="remote-view-1"
 945           style="display:none" autoplay></video></td>
 946       <td><video width="320" height="240" id="remote-view-2"
 947           style="display:none" autoplay></video></td>
 948       <td><video width="320" height="240" id="remote-view-3"
 949         style="display:none" autoplay></video></td>
 950       <td><video width="320" height="240" id="remote-view-4"
 951         style="display:none" autoplay></video></td>
 952       <!-- Canvases are named after their corresponding video elements. -->
 953       <td><canvas width="320" height="240" id="remote-view-1-canvas"
 954           style="display:none"></canvas></td>
 955       <td><canvas width="320" height="240" id="remote-view-2-canvas"
 956           style="display:none"></canvas></td>
 957       <td><canvas width="320" height="240" id="remote-view-3-canvas"
 958           style="display:none"></canvas></td>
 959       <td><canvas width="320" height="240" id="remote-view-4-canvas"
 960           style="display:none"></canvas></td>
 961     </tr>
 962   </table>
 963 </body>
 964 </html>