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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-05-15 03:35:49 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-05-15 03:35:49 +0000
commitd8bbc7858622b6d9c278469aab701ca0b609cddf (patch)
treeeff41dc61d9f714852212739e6b3738b82a2af87 /testing/web-platform/tests/webrtc-encoded-transform
parentReleasing progress-linux version 125.0.3-1~progress7.99u1. (diff)
downloadfirefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.tar.xz
firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.zip
Merging upstream version 126.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'testing/web-platform/tests/webrtc-encoded-transform')
-rw-r--r--testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk-worker.js13
-rw-r--r--testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk.https.html16
-rw-r--r--testing/web-platform/tests/webrtc-encoded-transform/tentative/RTCEncodedAudioFrame-metadata.https.html129
3 files changed, 158 insertions, 0 deletions
diff --git a/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk-worker.js b/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk-worker.js
new file mode 100644
index 0000000000..cc9dee42ac
--- /dev/null
+++ b/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk-worker.js
@@ -0,0 +1,13 @@
+onrtctransform = async (event) => {
+ const { port } = event.transformer.options;
+ port.postMessage("started");
+
+ const reader = event.transformer.readable.getReader();
+ const writer = event.transformer.writable.getWriter();
+
+ const { done, value } = await reader.read();
+
+ writer.write(null).catch(err => port.postMessage([err.name, 'null']));
+ writer.write(value).catch(err => port.postMessage([err.name, value.constructor.name]));
+};
+self.postMessage('registered');
diff --git a/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk.https.html b/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk.https.html
new file mode 100644
index 0000000000..a837f627f2
--- /dev/null
+++ b/testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk.https.html
@@ -0,0 +1,16 @@
+<!DOCTYPE html>
+<meta charset="utf-8">
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script src ="routines.js"></script>
+
+<video id="video1" autoplay></video>
+
+<script>
+promise_test(async (test) => {
+ const {sender, receiver, senderPc, receiverPc} = await createConnectionWithTransform(test, 'RTCRtpScriptTransform-bad-chunk-worker.js', {audio: true});
+
+ assert_array_equals(await getNextMessage(sender.transform.port), ["TypeError", "null"]);
+ assert_array_equals(await getNextMessage(sender.transform.port), ["TypeError", "RTCEncodedAudioFrame"]);
+}, "Writing bad chunks should error the stream");
+</script>
diff --git a/testing/web-platform/tests/webrtc-encoded-transform/tentative/RTCEncodedAudioFrame-metadata.https.html b/testing/web-platform/tests/webrtc-encoded-transform/tentative/RTCEncodedAudioFrame-metadata.https.html
new file mode 100644
index 0000000000..1e420e6f72
--- /dev/null
+++ b/testing/web-platform/tests/webrtc-encoded-transform/tentative/RTCEncodedAudioFrame-metadata.https.html
@@ -0,0 +1,129 @@
+<!DOCTYPE html>
+<meta charset="utf-8">
+<title>RTCEncodedAudioFrame can be cloned and distributed</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script src=/resources/testdriver.js></script>
+<script src=/resources/testdriver-vendor.js></script>
+<script src='../../mediacapture-streams/permission-helper.js'></script>
+<script src="../../webrtc/RTCPeerConnection-helper.js"></script>
+<script src="../../service-workers/service-worker/resources/test-helpers.sub.js"></script>
+
+<script>
+"use strict";
+promise_test(async t => {
+ const caller1 = new RTCPeerConnection();
+ t.add_cleanup(() => caller1.close());
+ const callee1 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => callee1.close());
+ await setMediaPermission("granted", ["microphone"]);
+ const inputStream = await navigator.mediaDevices.getUserMedia({audio:true});
+ const inputTrack = inputStream.getAudioTracks()[0];
+ t.add_cleanup(() => inputTrack.stop());
+ caller1.addTrack(inputTrack)
+ exchangeIceCandidates(caller1, callee1);
+
+ const caller2 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => caller2.close());
+ const sender2 = caller2.addTransceiver("audio").sender;
+ const writer2 = sender2.createEncodedStreams().writable.getWriter();
+ sender2.replaceTrack(new MediaStreamTrackGenerator({ kind: 'audio' }));
+
+ const framesReceivedCorrectly = new Promise((resolve, reject) => {
+ callee1.ontrack = async e => {
+ const receiverStreams = e.receiver.createEncodedStreams();
+ const receiverReader = receiverStreams.readable.getReader();
+ const result = await receiverReader.read();
+ const original = result.value;
+ let newFrame = new RTCEncodedAudioFrame(original);
+ assert_equals(original.getMetadata().rtpTimestamp, newFrame.getMetadata().rtpTimestamp);
+ assert_equals(original.getMetadata().absCaptureTime, newFrame.getMetadata().absCaptureTime);
+ assert_array_equals(Array.from(original.data), Array.from(newFrame.data));
+ await writer2.write(newFrame);
+ resolve();
+ }
+ });
+
+ await exchangeOfferAnswer(caller1, callee1);
+
+ return framesReceivedCorrectly;
+}, "Constructing audio frame before sending works");
+
+promise_test(async t => {
+ const caller1 = new RTCPeerConnection();
+
+ t.add_cleanup(() => caller1.close());
+ const callee1 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => callee1.close());
+ await setMediaPermission("granted", ["microphone"]);
+ const inputStream = await navigator.mediaDevices.getUserMedia({audio:true});
+ const inputTrack = inputStream.getAudioTracks()[0];
+ t.add_cleanup(() => inputTrack.stop());
+ caller1.addTrack(inputTrack)
+ exchangeIceCandidates(caller1, callee1);
+
+ const caller2 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => caller2.close());
+ const sender2 = caller2.addTransceiver("audio").sender;
+ const writer2 = sender2.createEncodedStreams().writable.getWriter();
+ sender2.replaceTrack(new MediaStreamTrackGenerator({ kind: 'audio' }));
+
+ const framesReceivedCorrectly = new Promise((resolve, reject) => {
+ callee1.ontrack = async e => {
+ const receiverStreams = e.receiver.createEncodedStreams();
+ const receiverReader = receiverStreams.readable.getReader();
+ const result = await receiverReader.read();
+ const original = result.value;
+ let newMetadata = original.getMetadata();
+ newMetadata.rtpTimestamp = newMetadata.rtpTimestamp + 1;
+ let newFrame = new RTCEncodedAudioFrame(original, newMetadata);
+ assert_not_equals(original.getMetadata().rtpTimestamp, newFrame.getMetadata().rtpTimestamp);
+ assert_equals(newMetadata.rtpTimestamp, newFrame.getMetadata().rtpTimestamp);
+ assert_equals(original.getMetadata().absCaptureTime, newFrame.getMetadata().absCaptureTime);
+ assert_array_equals(Array.from(original.data), Array.from(newFrame.data));
+ await writer2.write(newFrame);
+ resolve();
+ }
+ });
+
+ await exchangeOfferAnswer(caller1, callee1);
+
+ return framesReceivedCorrectly;
+}, "Constructing audio frame with metadata argument before sending works");
+
+promise_test(async t => {
+ const caller1 = new RTCPeerConnection();
+ t.add_cleanup(() => caller1.close());
+ const callee1 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => callee1.close());
+ await setMediaPermission("granted", ["microphone"]);
+ const inputStream = await navigator.mediaDevices.getUserMedia({audio:true});
+ const inputTrack = inputStream.getAudioTracks()[0];
+ t.add_cleanup(() => inputTrack.stop());
+ caller1.addTrack(inputTrack)
+ exchangeIceCandidates(caller1, callee1);
+
+ const caller2 = new RTCPeerConnection({encodedInsertableStreams:true});
+ t.add_cleanup(() => caller2.close());
+ const sender2 = caller2.addTransceiver("audio").sender;
+ const writer2 = sender2.createEncodedStreams().writable.getWriter();
+ sender2.replaceTrack(new MediaStreamTrackGenerator({ kind: 'audio' }));
+
+ const framesReceivedCorrectly = new Promise((resolve, reject) => {
+ callee1.ontrack = async e => {
+ const receiverStreams = e.receiver.createEncodedStreams();
+ const receiverReader = receiverStreams.readable.getReader();
+ const result = await receiverReader.read();
+ const original = result.value;
+ let newMetadata = original.getMetadata();
+ newMetadata.synchronizationSource = newMetadata.synchronizationSource + 1;
+ assert_throws_dom("InvalidModificationError", () => new RTCEncodedAudioFrame(original, newMetadata));
+ resolve();
+ }
+ });
+
+ await exchangeOfferAnswer(caller1, callee1);
+
+ return framesReceivedCorrectly;
+}, "Constructing audio frame with bad metadata argument before sending does not work");
+</script>
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