-
Notifications
You must be signed in to change notification settings - Fork 1.8k
[GStreamer][WebRTC] Tighten EndPoint pipeline with playback pipeline #13235
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
Conversation
|
EWS run on previous version of this PR (hash 1035584)
|
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
Shouldn't you assert if that has a value and then then it?
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
can't parse the question :)
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
Do you mean
ASSERT(internalSource->m_webrtcSourceClientId)
auto clientId = internalSource->m_webrtcSourceClientId.value();
¿
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
Sorry, I meant that I think you should assert on the value to have an actual value and then retrieving it. Now you're doing the other way around and it looks weird, but I might be missing something.
|
EWS run on current version of this PR (hash 993b131) |
https://bugs.webkit.org/show_bug.cgi?id=256041 Reviewed by Xabier Rodriguez-Calvar. The RTP depayloaders and parsers are now wrapped by the RealtimeIncomingSourceGStreamer sub-classes, using parsebin. The relationship between the incoming source appsink and the mediastreamsrc appsrc elements is also stronger, latency is now properly propagated and queries are relayed. This all helps improving lip-sync of incoming WebRTC tracks. * Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp: (WebCore::MediaPlayerPrivateGStreamer::configureElement): (WebCore::MediaPlayerPrivateGStreamer::configureDepayloader): Deleted. * Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h: * Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp: (webkitMediaStreamSrcAddTrack): * Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingSourceGStreamer.cpp: (WebCore::RealtimeIncomingSourceGStreamer::RealtimeIncomingSourceGStreamer): (WebCore::RealtimeIncomingSourceGStreamer::registerClient): (WebCore::RealtimeIncomingSourceGStreamer::unregisterClient): (WebCore::RealtimeIncomingSourceGStreamer::handleUpstreamEvent): (WebCore::RealtimeIncomingSourceGStreamer::handleUpstreamQuery): * Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingSourceGStreamer.h: (WebCore::RealtimeIncomingSourceGStreamer::bin): Canonical link: https://commits.webkit.org/263499@main
993b131 to
72fe397
Compare
|
Committed 263499@main (72fe397): https://commits.webkit.org/263499@main Reviewed commits have been landed. Closing PR #13235 and removing active labels. |
🛠 🧪 jsc-arm64
72fe397
993b131
🧪 wpe-wk2🧪 gtk-wk2🧪 mac-AS-debug-wk2