Diff
Modified: trunk/LayoutTests/ChangeLog (260813 => 260814)
--- trunk/LayoutTests/ChangeLog 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/LayoutTests/ChangeLog 2020-04-28 10:08:53 UTC (rev 260814)
@@ -1,3 +1,16 @@
+2020-04-28 Youenn Fablet <[email protected]>
+
+ RTCPeerConnection should not remove its created remote MediaStream objects until getting close
+ https://bugs.webkit.org/show_bug.cgi?id=211070
+
+ Reviewed by Alex Christensen.
+
+ * webrtc/direction-change-expected.txt: Added.
+ * webrtc/direction-change.html: Added.
+ * webrtc/routines.js:
+ * webrtc/video-setDirection-expected.txt:
+ * webrtc/video-setDirection.html:
+
2020-04-27 Simon Fraser <[email protected]>
Do correct clipping of composited replaced elements with border-radius
Added: trunk/LayoutTests/webrtc/direction-change-expected.txt (0 => 260814)
--- trunk/LayoutTests/webrtc/direction-change-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/direction-change-expected.txt 2020-04-28 10:08:53 UTC (rev 260814)
@@ -0,0 +1,8 @@
+
+
+PASS Setup connection with sendonly transceiver
+PASS Set transceiver as inactive
+PASS Set transceiver as sendonly
+PASS track event stream test - audio
+PASS track event stream test - video
+
Added: trunk/LayoutTests/webrtc/direction-change.html (0 => 260814)
--- trunk/LayoutTests/webrtc/direction-change.html (rev 0)
+++ trunk/LayoutTests/webrtc/direction-change.html 2020-04-28 10:08:53 UTC (rev 260814)
@@ -0,0 +1,54 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>MediaStream objects should remain the same for a remote track when changing direction</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <video id="video" autoplay=""></video>
+ <script src =""
+ <script>
+var pc1, pc2;
+var remoteStream;
+var trackEventCounter = 0;
+promise_test(async () => {
+ const localStream = await navigator.mediaDevices.getUserMedia({audio: true, video: true});
+ remoteStream = await new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ pc1 = firstConnection;
+ pc1.addTrack(localStream.getAudioTracks()[0], localStream);
+ pc1.addTrack(localStream.getVideoTracks()[0], localStream);
+ pc1.getTransceivers().forEach(transceiver => transceiver.direction = "sendonly");
+ }, (secondConnection) => {
+ pc2 = secondConnection;
+ pc2._ontrack_ = (trackEvent) => {
+ resolve(trackEvent.streams[0]);
+ };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+ video.srcObject = remoteStream;
+ await video.play();
+}, "Setup connection with sendonly transceiver");
+
+promise_test(async () => {
+ pc1.getTransceivers().forEach(transceiver => transceiver.direction = "inactive");
+ return renegotiate(pc1, pc2);
+}, "Set transceiver as inactive");
+
+promise_test(async () => {
+ // Setting direction to sendonly will trigger two new track events.
+ pc2._ontrack_ = (trackEvent) => {
+ test(() => {
+ assert_equals(remoteStream, trackEvent.streams[0]);
+ }, "track event stream test - " + trackEvent.track.kind);
+ };
+
+ pc1.getTransceivers().forEach(transceiver => transceiver.direction = "sendonly");
+ return renegotiate(pc1, pc2);
+}, "Set transceiver as sendonly");
+ </script>
+ </body>
+</html>
Modified: trunk/LayoutTests/webrtc/routines.js (260813 => 260814)
--- trunk/LayoutTests/webrtc/routines.js 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/LayoutTests/webrtc/routines.js 2020-04-28 10:08:53 UTC (rev 260814)
@@ -78,6 +78,16 @@
assert_unreached();
}
+async function renegotiate(pc1, pc2)
+{
+ let d = await pc1.createOffer();
+ await pc1.setLocalDescription(d);
+ await pc2.setRemoteDescription(d);
+ d = await pc2.createAnswer();
+ await pc1.setRemoteDescription(d);
+ await pc2.setLocalDescription(d);
+}
+
function analyseAudio(stream, duration, context)
{
return new Promise((resolve, reject) => {
Modified: trunk/LayoutTests/webrtc/video-setDirection-expected.txt (260813 => 260814)
--- trunk/LayoutTests/webrtc/video-setDirection-expected.txt 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/LayoutTests/webrtc/video-setDirection-expected.txt 2020-04-28 10:08:53 UTC (rev 260814)
@@ -1,5 +1,5 @@
PASS Going from sendrecv to inactive and back to sendrecv
-FAIL The MediaStream should remain the same assert_equals: expected object "[object MediaStream]" but got object "[object MediaStream]"
+PASS The MediaStream should remain the same
Modified: trunk/LayoutTests/webrtc/video-setDirection.html (260813 => 260814)
--- trunk/LayoutTests/webrtc/video-setDirection.html 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/LayoutTests/webrtc/video-setDirection.html 2020-04-28 10:08:53 UTC (rev 260814)
@@ -44,15 +44,6 @@
}
var pc1, pc2;
-async function renegotiate()
-{
- let d = await pc1.createOffer();
- await pc1.setLocalDescription(d);
- await pc2.setRemoteDescription(d);
- d = await pc2.createAnswer();
- await pc1.setRemoteDescription(d);
- await pc2.setLocalDescription(d);
-}
promise_test(async (t) => {
if (window.testRunner)
@@ -80,7 +71,7 @@
});
pc1.getTransceivers()[0].direction = "inactive";
- await renegotiate();
+ await renegotiate(pc1, pc2);
await promise;
promise = new Promise((resolve) => {
@@ -92,7 +83,7 @@
pc2._ontrack_ = (trackEvent) => { resolve (trackEvent.streams[0]); };
});
- await renegotiate();
+ await renegotiate(pc1, pc2);
video.srcObject = await streamPromise;
await promise;
Modified: trunk/Source/WebCore/ChangeLog (260813 => 260814)
--- trunk/Source/WebCore/ChangeLog 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/Source/WebCore/ChangeLog 2020-04-28 10:08:53 UTC (rev 260814)
@@ -1,5 +1,20 @@
2020-04-28 Youenn Fablet <[email protected]>
+ RTCPeerConnection should not remove its created remote MediaStream objects until getting close
+ https://bugs.webkit.org/show_bug.cgi?id=211070
+
+ Reviewed by Alex Christensen.
+
+ Remove no longer needed code.
+ This aligns with the spec and Firefox implementation.
+ Test: webrtc/direction-change.html
+
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::transceiverBackendFromSender):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+
+2020-04-28 Youenn Fablet <[email protected]>
+
Ensure remote track event gets unmuted after the track event is fired
https://bugs.webkit.org/show_bug.cgi?id=211071
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (260813 => 260814)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2020-04-28 10:08:53 UTC (rev 260814)
@@ -481,22 +481,6 @@
return nullptr;
}
-void LibWebRTCMediaEndpoint::removeRemoteStream(webrtc::MediaStreamInterface& rtcStream)
-{
- bool removed = m_remoteStreamsById.remove(fromStdString(rtcStream.id()));
- ASSERT_UNUSED(removed, removed);
-}
-
-void LibWebRTCMediaEndpoint::OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream)
-{
- callOnMainThread([protectedThis = makeRef(*this), stream = WTFMove(stream)] {
- if (protectedThis->isStopped())
- return;
- ASSERT(stream);
- protectedThis->removeRemoteStream(*stream.get());
- });
-}
-
void LibWebRTCMediaEndpoint::OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver)
{
callOnMainThread([protectedThis = makeRef(*this), transceiver = WTFMove(transceiver)]() mutable {
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (260813 => 260814)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2020-04-28 09:57:41 UTC (rev 260813)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2020-04-28 10:08:53 UTC (rev 260814)
@@ -122,7 +122,6 @@
// webrtc::PeerConnectionObserver API
void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState) final;
- void OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface>) final;
void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface>) final;
void OnTrack(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>) final;
void OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>) final;
@@ -139,7 +138,6 @@
void setLocalSessionDescriptionFailed(ExceptionCode, const char*);
void setRemoteSessionDescriptionSucceeded();
void setRemoteSessionDescriptionFailed(ExceptionCode, const char*);
- void removeRemoteStream(webrtc::MediaStreamInterface&);
void newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&&);
void removeRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&);
