WebPlatformForEmbedded · cadubentzen · Sep 3, 2022 · May 14, 2024 · Jun 5, 2024 · Apr 29, 2024
diff --git a/...3c/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-rollback-expected.txt b/...3c/web-platform-tests/webrtc/RTCPeerConnection-setRemoteDescription-rollback-expected.txt
@@ -17,7 +17,7 @@ PASS rollback of a local offer to negotiated stable state should enable applying
 PASS rollback a local offer with audio direction change to negotiated stable state and then add video receiver
 PASS two transceivers with same mids
 PASS onremovetrack fires during remote rollback
-FAIL rollback of a remote offer with stream changes assert_equals: expected 2 but got 1
+PASS rollback of a remote offer with stream changes
 PASS removeTrack() with a sender being rolled back does not crash or throw
 PASS Implicit rollback with only a datachannel works
 
diff --git a/...utTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setStreams.https-expected.txt b/...utTests/imported/w3c/web-platform-tests/webrtc/RTCRtpSender-setStreams.https-expected.txt
@@ -1,9 +1,7 @@
 
-Harness Error (TIMEOUT), message = null
-
 PASS setStreams causes streams to be reported via ontrack on callee
 PASS setStreams can be used to reconstruct a stream with a track on the remote side
 PASS Adding streams and changing direction causes new streams to be reported via ontrack on callee
-TIMEOUT Adding streams to an active transceiver causes new streams to be reported via ontrack on callee Test timed out
+PASS Adding streams to an active transceiver causes new streams to be reported via ontrack on callee
 PASS setStreams() fires InvalidStateError on a closed peer connection.
 
diff --git a/LayoutTests/platform/glib/TestExpectations b/LayoutTests/platform/glib/TestExpectations
@@ -1474,6 +1474,8 @@ webrtc/video-av1.html [ Skip ]
 
 # GStreamer's DTLS agent currently generates RSA certificates only. DTLS 1.2 is not supported yet (AFAIK).
 webrtc/datachannel/dtls10.html [ Failure ]
+# FIXME: Remove Timeout expectation once bug #275685 is fixed.
+webkit.org/b/269285 webrtc/h265.html [ Failure Timeout ]
 
 # Too slow with filtering implemented in WebKit. Should be done directly by GstWebRTC.
 webrtc/datachannel/filter-ice-candidate.html [ Skip ]

diff --git a/LayoutTests/webrtc/msid-setCodecPreferences-expected.txt b/LayoutTests/webrtc/msid-setCodecPreferences-expected.txt
@@ -0,0 +1,3 @@
+
+PASS msid present in offer SDP after setting codec preferences
+
diff --git a/LayoutTests/webrtc/msid-setCodecPreferences.html b/LayoutTests/webrtc/msid-setCodecPreferences.html
@@ -0,0 +1,27 @@
+<!doctype html>
+<html>
+    <head>
+        <meta charset="utf-8">
+        <title>Testing basic video exchange from offerer to receiver</title>
+        <script src="../resources/testharness.js"></script>
+        <script src="../resources/testharnessreport.js"></script>
+    </head>
+    <body>
+        <script src ="routines.js"></script>
+        <script>
+var track, firstConnection, secondConnection;
+promise_test(async (test) => {
+    if (window.testRunner)
+        testRunner.setUserMediaPermission(true);
+
+    let pc = new RTCPeerConnection();
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true });
+	let track = stream.getVideoTracks()[0];
+    pc.addTrack(track, stream);
+    pc.getTransceivers()[0].setCodecPreferences([{mimeType: "video/VP8", clockRate: 90000}]);
+    let offer = await pc.createOffer();
+    assert_true(offer.sdp.includes(`a=msid:${stream.id} ${track.id}`), 'offer SDP includes a=msid line');
+}, "msid present in offer SDP after setting codec preferences");
+        </script>
+    </body>
+</html>
diff --git a/LayoutTests/webrtc/setConfiguration-after-createDataChannel-or-addTransceiver-expected.txt b/LayoutTests/webrtc/setConfiguration-after-createDataChannel-or-addTransceiver-expected.txt
@@ -0,0 +1,5 @@
+
+PASS setConfiguration after data channel is created
+PASS setConfiguration after video transceiver is added
+PASS setConfiguration after audio transceiver is added
+
diff --git a/LayoutTests/webrtc/setConfiguration-after-createDataChannel-or-addTransceiver.html b/LayoutTests/webrtc/setConfiguration-after-createDataChannel-or-addTransceiver.html
@@ -0,0 +1,39 @@
+<!doctype html>
+<html>
+    <head>
+        <meta charset="utf-8">
+        <title>Testing media fields in SDP when setConfiguration comes after createDataChannel/addTransceiver</title>
+        <script src="../resources/testharness.js"></script>
+        <script src="../resources/testharnessreport.js"></script>
+    </head>
+    <body>
+        <script src ="routines.js"></script>
+        <script>
+function testMediaInSDP(addTransceiverOrDataChannel, regex) {
+    return async (test) => {
+        const pc = new RTCPeerConnection();
+        addTransceiverOrDataChannel(pc);
+        pc.setConfiguration({});
+        await pc.setLocalDescription();
+        const sdp = pc.localDescription.sdp;
+        assert_true(regex.test(sdp));
+    }
+}
+
+promise_test(testMediaInSDP(
+        pc => pc.createDataChannel("data-channel"),
+        /\r\nm=application.*webrtc-datachannel\r\n/),
+    'setConfiguration after data channel is created');
+
+promise_test(testMediaInSDP(
+        pc => pc.addTransceiver("video"),
+        /\r\nm=video.*\r\n/),
+    'setConfiguration after video transceiver is added');
+
+promise_test(testMediaInSDP(
+        pc => pc.addTransceiver("audio"),
+        /\r\nm=audio.*\r\n/),
+    'setConfiguration after audio transceiver is added');
+        </script>
+    </body>
+</html>
diff --git a/Source/WebCore/Modules/mediastream/MediaStream.h b/Source/WebCore/Modules/mediastream/MediaStream.h
@@ -78,6 +78,9 @@ class MediaStream final
 
     RefPtr<MediaStream> clone();
 
+    using WeakValueType = MediaStreamPrivate::Observer::WeakValueType;
+    using MediaStreamPrivate::Observer::weakPtrFactory;
+
     bool active() const { return m_isActive; }
     bool muted() const { return m_private->muted(); }
 

diff --git a/Source/WebCore/Modules/mediastream/MediaStreamTrack.cpp b/Source/WebCore/Modules/mediastream/MediaStreamTrack.cpp
@@ -610,12 +610,21 @@ void MediaStreamTrack::trackMutedChanged(MediaStreamTrackPrivate&)
     if (document->activeDOMObjectsAreStopped() || m_ended)
         return;
 
-    queueTaskKeepingObjectAlive(*this, TaskSource::Networking, [this, muted = m_private->muted()] {
+    Function<void()> updateMuted = [this, muted = m_private->muted()] {
         if (!scriptExecutionContext() || scriptExecutionContext()->activeDOMObjectsAreStopped())
             return;
+
+        if (m_muted == muted)
+            return;
+
         m_muted = muted;
         dispatchEvent(Event::create(muted ? eventNames().muteEvent : eventNames().unmuteEvent, Event::CanBubble::No, Event::IsCancelable::No));
-    });
+    };
+    if (m_shouldFireMuteEventImmediately)
+        updateMuted();
+    else
+        queueTaskKeepingObjectAlive(*this, TaskSource::Networking, WTFMove(updateMuted));
+
     configureTrackRendering();
 
     bool wasInterrupted = m_isInterrupted;

diff --git a/Source/WebCore/Modules/mediastream/MediaStreamTrack.h b/Source/WebCore/Modules/mediastream/MediaStreamTrack.h
@@ -163,6 +163,8 @@ class MediaStreamTrack
     const void* logIdentifier() const final { return m_private->logIdentifier(); }
 #endif
 
+    void setShouldFireMuteEventImmediately(bool value) { m_shouldFireMuteEventImmediately = value; }
+
 protected:
     MediaStreamTrack(ScriptExecutionContext&, Ref<MediaStreamTrackPrivate>&&);
 
@@ -212,6 +214,7 @@ class MediaStreamTrack
     bool m_ended { false };
     const bool m_isCaptureTrack { false };
     bool m_isInterrupted { false };
+    bool m_shouldFireMuteEventImmediately { false };
 };
 
 typedef Vector<Ref<MediaStreamTrack>> MediaStreamTrackVector;