From def92d1b8e9d373e2f6f27c366d578d97d8960c6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Wed, 15 May 2024 05:34:50 +0200
Subject: Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../RTCRtpScriptTransform-bad-chunk-worker.js      |  13 +++
 .../RTCRtpScriptTransform-bad-chunk.https.html     |  16 +++
 .../RTCEncodedAudioFrame-metadata.https.html       | 129 +++++++++++++++++++++
 3 files changed, 158 insertions(+)
 create mode 100644 testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk-worker.js
 create mode 100644 testing/web-platform/tests/webrtc-encoded-transform/RTCRtpScriptTransform-bad-chunk.https.html
 create mode 100644 testing/web-platform/tests/webrtc-encoded-transform/tentative/RTCEncodedAudioFrame-metadata.https.html

(limited to 'testing/web-platform/tests/webrtc-encoded-transform')
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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