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| author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
|---|---|---|
| committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
| commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
| tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html | |
| parent | Initial commit. (diff) | |
| download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip | |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html')
| -rw-r--r-- | testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html | 92 |
1 files changed, 92 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html b/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html new file mode 100644 index 0000000000..63ad9bf888 --- /dev/null +++ b/testing/web-platform/tests/webrtc-extensions/RTCRtpSynchronizationSource-senderCaptureTimeOffset.html @@ -0,0 +1,92 @@ +<!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> |