Diff
Modified: trunk/LayoutTests/ChangeLog (240151 => 240152)
--- trunk/LayoutTests/ChangeLog 2019-01-18 16:54:23 UTC (rev 240151)
+++ trunk/LayoutTests/ChangeLog 2019-01-18 17:19:26 UTC (rev 240152)
@@ -1,3 +1,14 @@
+2019-01-18 Youenn Fablet <you...@apple.com>
+
+ A track source should be unmuted whenever reenabled after setDirection changes
+ https://bugs.webkit.org/show_bug.cgi?id=193554
+ <rdar://problem/47366196>
+
+ Reviewed by Eric Carlson.
+
+ * webrtc/video-setDirection-expected.txt: Added.
+ * webrtc/video-setDirection.html: Added.
+
2019-01-18 Jonathan Bedard <jbed...@apple.com>
webkitpy: Implement device type specific expected results (Part 2)
Added: trunk/LayoutTests/webrtc/video-setDirection-expected.txt (0 => 240152)
--- trunk/LayoutTests/webrtc/video-setDirection-expected.txt (rev 0)
+++ trunk/LayoutTests/webrtc/video-setDirection-expected.txt 2019-01-18 17:19:26 UTC (rev 240152)
@@ -0,0 +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]"
+
Added: trunk/LayoutTests/webrtc/video-setDirection.html (0 => 240152)
--- trunk/LayoutTests/webrtc/video-setDirection.html (rev 0)
+++ trunk/LayoutTests/webrtc/video-setDirection.html 2019-01-18 17:19:26 UTC (rev 240152)
@@ -0,0 +1,108 @@
+<!doctype html>
+<html>
+ <head>
+ <meta charset="utf-8">
+ <title>Testing video exchange using setDirection with renegotiation from offerer to receiver</title>
+ <script src=""
+ <script src=""
+ </head>
+ <body>
+ <video id="video" autoplay=""></video>
+ <canvas id="canvas" width="640" height="480"></canvas>
+ <script src =""
+ <script>
+video = document.getElementById("video");
+canvas = document.getElementById("canvas");
+
+function grabFrameData(x, y, w, h)
+{
+ canvas.width = video.videoWidth;
+ canvas.height = video.videoHeight;
+
+ canvas.getContext('2d').drawImage(video, x, y, w, h, x, y, w, h);
+ return canvas.getContext('2d').getImageData(x, y, w, h).data;
+}
+
+function testImage()
+{
+ const data = "" 325, 250, 1);
+
+ var index = 20;
+ assert_true(data[index] < 100);
+ assert_true(data[index + 1] < 100);
+ assert_true(data[index + 2] < 100);
+
+ index = 80;
+ assert_true(data[index] > 200);
+ assert_true(data[index + 1] > 200);
+ assert_true(data[index + 2] > 200);
+
+ index += 80;
+ assert_true(data[index] > 200);
+ assert_true(data[index + 1] > 200);
+ assert_true(data[index + 2] < 100);
+}
+
+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)
+ testRunner.setUserMediaPermission(true);
+
+ const localStream = await navigator.mediaDevices.getUserMedia({video: true});
+ const stream = await new Promise((resolve, reject) => {
+ createConnections((firstConnection) => {
+ pc1 = firstConnection;
+ firstConnection.addTrack(localStream.getVideoTracks()[0], localStream);
+ }, (secondConnection) => {
+ pc2 = secondConnection;
+ secondConnection._ontrack_ = (trackEvent) => { resolve(trackEvent.streams[0]); };
+ });
+ setTimeout(() => reject("Test timed out"), 5000);
+ });
+
+ video.srcObject = stream;
+ await video.play();
+
+ testImage();
+
+ let promise = new Promise((resolve) => {
+ pc2.getReceivers()[0].track._onmute_ = resolve;
+ });
+
+ pc1.getTransceivers()[0].direction = "inactive";
+ await renegotiate();
+ await promise;
+
+ promise = new Promise((resolve) => {
+ pc2.getReceivers()[0].track._onunmute_ = resolve;
+ });
+
+ pc1.getTransceivers()[0].direction = "sendrecv";
+ const streamPromise = new Promise(resolve => {
+ pc2._ontrack_ = (trackEvent) => { resolve (trackEvent.streams[0]); };
+ });
+
+ await renegotiate();
+ video.srcObject = await streamPromise;
+ await promise;
+
+ test(() => {
+ assert_equals(stream, video.srcObject);
+ }, "The MediaStream should remain the same");
+ await video.play();
+
+ testImage();
+}, "Going from sendrecv to inactive and back to sendrecv");
+ </script>
+ </body>
+</html>
Modified: trunk/Source/WebCore/ChangeLog (240151 => 240152)
--- trunk/Source/WebCore/ChangeLog 2019-01-18 16:54:23 UTC (rev 240151)
+++ trunk/Source/WebCore/ChangeLog 2019-01-18 17:19:26 UTC (rev 240152)
@@ -1,3 +1,25 @@
+2019-01-18 Youenn Fablet <you...@apple.com>
+
+ A track source should be unmuted whenever reenabled after setDirection changes
+ https://bugs.webkit.org/show_bug.cgi?id=193554
+ <rdar://problem/47366196>
+
+ Reviewed by Eric Carlson.
+
+ Ensure that track gets unmuted after being fired as part of track event.
+ Test is triggering some existing issues with MediaPlayerPrivateMediaStreamAVFObjC.
+ Given the enqueuing of samples happens in a different frame than the thread used to update media stream and the active video track,
+ some enqueued samples might not be from the right active video track or there might be no active video track.
+
+ Test: webrtc/video-setDirection.html
+
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
+ (WebCore::LibWebRTCMediaEndpoint::fireTrackEvent):
+ * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
+ * platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm:
+ (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::enqueueVideoSample):
+ (WebCore::MediaPlayerPrivateMediaStreamAVFObjC::requestNotificationWhenReadyForVideoData):
+
2019-01-18 Charlie Turner <ctur...@igalia.com>
[GStreamer][EME][ClearKey] Request keys from CDMInstance rather than passing via bus messages
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp (240151 => 240152)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2019-01-18 16:54:23 UTC (rev 240151)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp 2019-01-18 17:19:26 UTC (rev 240152)
@@ -413,21 +413,24 @@
fireTrackEvent(receiver.releaseNonNull(), track, rtcStreams, nullptr);
}
-void LibWebRTCMediaEndpoint::fireTrackEvent(Ref<RTCRtpReceiver>&& receiver, Ref<MediaStreamTrack>&& track, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams, RefPtr<RTCRtpTransceiver>&& transceiver)
+void LibWebRTCMediaEndpoint::fireTrackEvent(Ref<RTCRtpReceiver>&& receiver, MediaStreamTrack& track, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& rtcStreams, RefPtr<RTCRtpTransceiver>&& transceiver)
{
Vector<RefPtr<MediaStream>> streams;
for (auto& rtcStream : rtcStreams) {
auto& mediaStream = mediaStreamFromRTCStream(*rtcStream.get());
streams.append(&mediaStream);
- mediaStream.addTrackFromPlatform(track.get());
+ mediaStream.addTrackFromPlatform(track);
}
auto streamIds = WTF::map(streams, [](auto& stream) -> String {
return stream->id();
});
- m_remoteStreamsFromRemoteTrack.add(track.ptr(), WTFMove(streamIds));
+ m_remoteStreamsFromRemoteTrack.add(&track, WTFMove(streamIds));
m_peerConnectionBackend.connection().fireEvent(RTCTrackEvent::create(eventNames().trackEvent,
- Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), WTFMove(track), WTFMove(streams), WTFMove(transceiver)));
+ Event::CanBubble::No, Event::IsCancelable::No, WTFMove(receiver), &track, WTFMove(streams), WTFMove(transceiver)));
+
+ // FIXME: As per spec, we should set muted to 'false' when starting to receive the content from network.
+ track.source().setMuted(false);
}
static inline void setExistingReceiverSourceTrack(RealtimeMediaSource& existingSource, webrtc::RtpReceiverInterface& rtcReceiver)
Modified: trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h (240151 => 240152)
--- trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2019-01-18 16:54:23 UTC (rev 240151)
+++ trunk/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h 2019-01-18 17:19:26 UTC (rev 240152)
@@ -143,7 +143,7 @@
void newTransceiver(rtc::scoped_refptr<webrtc::RtpTransceiverInterface>&&);
void removeRemoteTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>&&);
- void fireTrackEvent(Ref<RTCRtpReceiver>&&, Ref<MediaStreamTrack>&&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&, RefPtr<RTCRtpTransceiver>&&);
+ void fireTrackEvent(Ref<RTCRtpReceiver>&&, MediaStreamTrack&, const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&, RefPtr<RTCRtpTransceiver>&&);
template<typename T>
Optional<Backends> createTransceiverBackends(T&&, const RTCRtpTransceiverInit&, LibWebRTCRtpSenderBackend::Source&&);
Modified: trunk/Source/WebCore/platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm (240151 => 240152)
--- trunk/Source/WebCore/platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm 2019-01-18 16:54:23 UTC (rev 240151)
+++ trunk/Source/WebCore/platform/graphics/avfoundation/objc/MediaPlayerPrivateMediaStreamAVFObjC.mm 2019-01-18 17:19:26 UTC (rev 240152)
@@ -359,8 +359,6 @@
void MediaPlayerPrivateMediaStreamAVFObjC::enqueueVideoSample(MediaStreamTrackPrivate& track, MediaSample& sample)
{
- ASSERT(m_videoTrackMap.contains(track.id()));
-
if (&track != m_mediaStreamPrivate->activeVideoTrack())
return;
@@ -414,6 +412,11 @@
[m_sampleBufferDisplayLayer stopRequestingMediaData];
+ if (!m_activeVideoTrack) {
+ m_pendingVideoSampleQueue.clear();
+ return;
+ }
+
while (!m_pendingVideoSampleQueue.isEmpty()) {
if (![m_sampleBufferDisplayLayer isReadyForMoreMediaData]) {
requestNotificationWhenReadyForVideoData();
