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| 1 <html> | 1 <html> |
| 2 <head> | 2 <head> |
| 3 <script type="text/javascript" src="webrtc_test_utilities.js"></script> | 3 <script type="text/javascript" src="webrtc_test_utilities.js"></script> |
| 4 <script type="text/javascript" src="webrtc_test_audio.js"></script> | 4 <script type="text/javascript" src="webrtc_test_audio.js"></script> |
| 5 <script type="text/javascript"> | 5 <script type="text/javascript"> |
| 6 $ = function(id) { | 6 $ = function(id) { |
| 7 return document.getElementById(id); | 7 return document.getElementById(id); |
| 8 }; | 8 }; |
| 9 | 9 |
| 10 window.onerror = function(errorMsg, url, lineNumber, column, errorObj) { | 10 window.onerror = function(errorMsg, url, lineNumber, column, errorObj) { |
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| 150 | 150 |
| 151 var onCallEstablished = function() { | 151 var onCallEstablished = function() { |
| 152 ensureAudioPlaying(gSecondConnection, beLenient); | 152 ensureAudioPlaying(gSecondConnection, beLenient); |
| 153 }; | 153 }; |
| 154 | 154 |
| 155 waitForConnectionToStabilize(gFirstConnection, onCallEstablished); | 155 waitForConnectionToStabilize(gFirstConnection, onCallEstablished); |
| 156 }); | 156 }); |
| 157 } | 157 } |
| 158 | 158 |
| 159 // First makes a call between pc1 and pc2 where a stream is sent from pc1 to | 159 // First makes a call between pc1 and pc2 where a stream is sent from pc1 to |
| 160 // pc2. The stream sent from pc1 to pc2 is cloned from the stream received on | 160 // pc2. The stream sent from pc1 to pc2 is cloned from the stream received on |
| 161 // pc2 to test that cloning of remote video tracks works as intended and is | 161 // pc2 to test that cloning of remote video tracks works as intended and is |
| 162 // sent back to pc1. | 162 // sent back to pc1. |
| 163 function callAndForwardRemoteStream(constraints) { | 163 function callAndForwardRemoteStream(constraints) { |
| 164 createConnections(null); | 164 createConnections(null); |
| 165 navigator.webkitGetUserMedia(constraints, | 165 navigator.webkitGetUserMedia(constraints, |
| 166 addStreamToTheFirstConnectionAndNegotiate, | 166 addStreamToTheFirstConnectionAndNegotiate, |
| 167 printGetUserMediaError); | 167 printGetUserMediaError); |
| 168 var onRemoteStream2 = function() { | 168 var onRemoteStream2 = function() { |
| 169 // Video has been detected to be playing in pc2. Clone the received | 169 // Video has been detected to be playing in pc2. Clone the received |
| 170 // stream and send it back to pc1. | 170 // stream and send it back to pc1. |
| 171 gSecondConnection.addStream(gRemoteStreams['remote-view-2'].clone()); | 171 gSecondConnection.addStream(gRemoteStreams['remote-view-2'].clone()); |
| 172 negotiate(); | 172 negotiate(); |
| 173 } | 173 } |
| 174 | 174 |
| 175 // Wait for remove video to be playing in pc2. Once video is playing, | 175 // Wait for remove video to be playing in pc2. Once video is playing, |
| 176 // forward the remove stream from pc2 to pc1. | 176 // forward the remove stream from pc2 to pc1. |
| 177 detectVideoPlaying('remote-view-2', onRemoteStream2); | 177 detectVideoPlaying('remote-view-2', onRemoteStream2); |
| 178 | 178 |
| 179 // Wait for video to be forwarded back to connection 1. | 179 // Wait for video to be forwarded back to connection 1. |
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| 324 waitForVideo('remote-view-2'); | 324 waitForVideo('remote-view-2'); |
| 325 }); | 325 }); |
| 326 } | 326 } |
| 327 | 327 |
| 328 // Test that we can setup call and send DTMF. | 328 // Test that we can setup call and send DTMF. |
| 329 function callAndSendDtmf(tones) { | 329 function callAndSendDtmf(tones) { |
| 330 createConnections(null); | 330 createConnections(null); |
| 331 navigator.webkitGetUserMedia({audio: true, video: true}, | 331 navigator.webkitGetUserMedia({audio: true, video: true}, |
| 332 addStreamToBothConnectionsAndNegotiate, printGetUserMediaError); | 332 addStreamToBothConnectionsAndNegotiate, printGetUserMediaError); |
| 333 var onCallEstablished = function() { | 333 var onCallEstablished = function() { |
| 334 // Send DTMF tones. | 334 // Send DTMF tones. Allocate the sender in the window to keep it from |
| 335 // being garbage collected. https://crbug.com/486654. |
| 335 var localAudioTrack = gLocalStream.getAudioTracks()[0]; | 336 var localAudioTrack = gLocalStream.getAudioTracks()[0]; |
| 336 var dtmfSender = gFirstConnection.createDTMFSender(localAudioTrack); | 337 window.dtmfSender = gFirstConnection.createDTMFSender(localAudioTrack); |
| 337 dtmfSender.ontonechange = onToneChange; | 338 window.dtmfSender.ontonechange = onToneChange; |
| 338 dtmfSender.insertDTMF(tones); | 339 window.dtmfSender.insertDTMF(tones); |
| 339 // Wait for the DTMF tones callback. | 340 // Wait for the DTMF tones callback. |
| 340 addExpectedEvent(); | 341 addExpectedEvent(); |
| 341 var waitDtmf = setInterval(function() { | 342 var waitDtmf = setInterval(function() { |
| 342 if (gSentTones == tones) { | 343 if (gSentTones == tones) { |
| 343 clearInterval(waitDtmf); | 344 clearInterval(waitDtmf); |
| 344 eventOccured(); | 345 eventOccured(); |
| 345 } | 346 } |
| 346 }, 100); | 347 }, 100); |
| 347 } | 348 } |
| 348 | 349 |
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| 933 <td><canvas width="320" height="240" id="remote-view-2-canvas" | 934 <td><canvas width="320" height="240" id="remote-view-2-canvas" |
| 934 style="display:none"></canvas></td> | 935 style="display:none"></canvas></td> |
| 935 <td><canvas width="320" height="240" id="remote-view-3-canvas" | 936 <td><canvas width="320" height="240" id="remote-view-3-canvas" |
| 936 style="display:none"></canvas></td> | 937 style="display:none"></canvas></td> |
| 937 <td><canvas width="320" height="240" id="remote-view-4-canvas" | 938 <td><canvas width="320" height="240" id="remote-view-4-canvas" |
| 938 style="display:none"></canvas></td> | 939 style="display:none"></canvas></td> |
| 939 </tr> | 940 </tr> |
| 940 </table> | 941 </table> |
| 941 </body> | 942 </body> |
| 942 </html> | 943 </html> |
| OLD | NEW |
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Chromium Code Reviews
Issue 1138073003:
Endowing WebRTC DTMF test with better resistance to gc(). (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master