| <!doctype html> |
| <meta charset=utf-8> |
| <!-- This file contains a test that waits for 2 seconds. --> |
| <meta name="timeout" content="long"> |
| <title>senderCaptureTimeOffset attribute in RTCRtpSynchronizationSource</title> |
| <div><video id="remote" width="124" height="124" autoplay></video></div> |
| <script src="/resources/testharness.js"></script> |
| <script src="/resources/testharnessreport.js"></script> |
| <script src="/webrtc/RTCPeerConnection-helper.js"></script> |
| <script src="/webrtc/RTCStats-helper.js"></script> |
| <script src="/webrtc-extensions/RTCRtpSynchronizationSource-helper.js"></script> |
| <script> |
| 'use strict'; |
| |
| function listenForSenderCaptureTimeOffset(t, receiver) { |
| return new Promise((resolve) => { |
| function listen() { |
| const ssrcs = receiver.getSynchronizationSources(); |
| assert_true(ssrcs != undefined); |
| if (ssrcs.length > 0) { |
| assert_equals(ssrcs.length, 1); |
| if (ssrcs[0].captureTimestamp != undefined) { |
| resolve(ssrcs[0].senderCaptureTimeOffset); |
| return true; |
| } |
| } |
| return false; |
| }; |
| t.step_wait(listen, 'No abs-capture-time capture time header extension.'); |
| }); |
| } |
| |
| // Passes if `getSynchronizationSources()` contains `senderCaptureTimeOffset` if |
| // and only if expected. |
| for (const kind of ['audio', 'video']) { |
| promise_test(async t => { |
| const [caller, callee] = await initiateSingleTrackCall( |
| t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */false, |
| /* absCaptureTimeAnswered= */false); |
| const receiver = callee.getReceivers()[0]; |
| |
| for (const ssrc of await listenForSSRCs(t, receiver)) { |
| assert_equals(typeof ssrc.senderCaptureTimeOffset, 'undefined'); |
| } |
| }, '[' + kind + '] getSynchronizationSources() should not contain ' + |
| 'senderCaptureTimeOffset if absolute capture time RTP header extension ' + |
| 'is not offered'); |
| |
| promise_test(async t => { |
| const [caller, callee] = await initiateSingleTrackCall( |
| t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */false, |
| /* absCaptureTimeAnswered= */false); |
| const receiver = callee.getReceivers()[0]; |
| |
| for (const ssrc of await listenForSSRCs(t, receiver)) { |
| assert_equals(typeof ssrc.senderCaptureTimeOffset, 'undefined'); |
| } |
| }, '[' + kind + '] getSynchronizationSources() should not contain ' + |
| 'senderCaptureTimeOffset if absolute capture time RTP header extension ' + |
| 'is offered, but not answered'); |
| |
| promise_test(async t => { |
| const [caller, callee] = await initiateSingleTrackCall( |
| t, /* caps= */{[kind]: true}, /* absCaptureTimeOffered= */true, |
| /* absCaptureTimeAnswered= */true); |
| const receiver = callee.getReceivers()[0]; |
| let senderCaptureTimeOffset = await listenForSenderCaptureTimeOffset( |
| t, receiver); |
| assert_true(senderCaptureTimeOffset != undefined); |
| }, '[' + kind + '] getSynchronizationSources() should contain ' + |
| 'senderCaptureTimeOffset if absolute capture time RTP header extension ' + |
| 'is negotiated'); |
| } |
| |
| // Passes if `senderCaptureTimeOffset` is zero, which is expected since the test |
| // creates a local peer connection between `caller` and `callee`. |
| promise_test(async t => { |
| const [caller, callee] = await initiateSingleTrackCall( |
| t, /* caps= */{audio: true, video: true}, |
| /* absCaptureTimeOffered= */true, /* absCaptureTimeAnswered= */true); |
| const receivers = callee.getReceivers(); |
| assert_equals(receivers.length, 2); |
| |
| for (let i = 0; i < 2; ++i) { |
| let senderCaptureTimeOffset = await listenForSenderCaptureTimeOffset( |
| t, receivers[i]); |
| assert_equals(senderCaptureTimeOffset, 0); |
| } |
| }, 'Audio and video RTCRtpSynchronizationSource.senderCaptureTimeOffset must ' + |
| 'be zero'); |
| |
| </script> |
