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Side by Side Diff: chrome/test/data/extensions/api_test/tab_capture/end_to_end.js

Issue 2446173002: Use RTCPeerConnection instead of webkitRTCPeerConnection (Closed)
Patch Set: rebase Created 4 years, 1 month ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 // The tests here cover the end-to-end functionality of tab capturing and 5 // The tests here cover the end-to-end functionality of tab capturing and
6 // playback as video. The page generates video test patterns that rotate 6 // playback as video. The page generates video test patterns that rotate
7 // cyclicly, and the rendering output of the tab is captured into a 7 // cyclicly, and the rendering output of the tab is captured into a
8 // local MediaStream. This stream is then piped into a video element for 8 // local MediaStream. This stream is then piped into a video element for
9 // playback, and a canvas is used to examine the frames of the video for 9 // playback, and a canvas is used to examine the frames of the video for
10 // expected content. The stream may be plumbed one of two ways, depending on 10 // expected content. The stream may be plumbed one of two ways, depending on
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200 maxFrameRate: frameRate, 200 maxFrameRate: frameRate,
201 } 201 }
202 } 202 }
203 }, 203 },
204 function remoteTheStream(captureStream) { 204 function remoteTheStream(captureStream) {
205 chrome.test.assertTrue(!!captureStream); 205 chrome.test.assertTrue(!!captureStream);
206 if (transportMethod == 'local') { 206 if (transportMethod == 'local') {
207 receiveStream = captureStream; 207 receiveStream = captureStream;
208 waitForExpectedColors(colorDeviation); 208 waitForExpectedColors(colorDeviation);
209 } else if (transportMethod == 'webrtc') { 209 } else if (transportMethod == 'webrtc') {
210 var sender = new webkitRTCPeerConnection(null); 210 var sender = new RTCPeerConnection();
211 var receiver = new webkitRTCPeerConnection(null); 211 var receiver = new RTCPeerConnection();
212 sender.onicecandidate = function (event) { 212 sender.onicecandidate = function (event) {
213 if (event.candidate) { 213 if (event.candidate) {
214 receiver.addIceCandidate(new RTCIceCandidate(event.candidate)); 214 receiver.addIceCandidate(new RTCIceCandidate(event.candidate));
215 } 215 }
216 }; 216 };
217 receiver.onicecandidate = function (event) { 217 receiver.onicecandidate = function (event) {
218 if (event.candidate) { 218 if (event.candidate) {
219 sender.addIceCandidate(new RTCIceCandidate(event.candidate)); 219 sender.addIceCandidate(new RTCIceCandidate(event.candidate));
220 } 220 }
221 }; 221 };
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237 } else { 237 } else {
238 chrome.test.fail("Unknown transport method: " + transportMethod); 238 chrome.test.fail("Unknown transport method: " + transportMethod);
239 } 239 }
240 }); 240 });
241 } 241 }
242 ]); 242 ]);
243 243
244 // TODO(miu): Once the WebAudio API is finalized, we should add a test to emit a 244 // TODO(miu): Once the WebAudio API is finalized, we should add a test to emit a
245 // tone from the sender page, and have the receiver page check for the audio 245 // tone from the sender page, and have the receiver page check for the audio
246 // tone. 246 // tone.
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