GstWebRTC backports from upstream#1387

Draft

cadubentzen wants to merge 15 commits intowpe-2.38from

cadubentzen/GstWebRTC

Commits on Aug 28, 2024

Move out of OnTrack/OnRemoveTrack libwebrtc callbacks

youennf
authored and
cadubentzen
committed
Incoming video track renderer should rely on the track intrinsic size https://bugs.webkit.org/show_bug.cgi?id=274093

philn
authored and
cadubentzen
committed
Missing a=msid in offer after setting codec preferences https://bugs.webkit.org/show_bug.cgi?id=275157
cadubentzen
committed
Missing media in SDP if setConfiguration() is called after createDataChannel() or addTransceiver() https://bugs.webkit.org/show_bug.cgi?id=273318
cadubentzen
committed
webrtc/connection-state.html started failing after update to 1.24.0 https://bugs.webkit.org/show_bug.cgi?id=271243 <rdar://problem/125014192>

philn
authored and
cadubentzen
committed
`m_maxMessageSize` should be `infinity()` instead of `max()` in `RTCSctpTransport.h`

Ahmad Saleem
authored and
cadubentzen
committed
Most of WPT webrtc/RTCSctpTransport-maxMessageSize.html tests are failing https://bugs.webkit.org/show_bug.cgi?id=274442 rdar://128444396

youennf
authored and
cadubentzen
committed
Add debug logging of TransceiverStates https://bugs.webkit.org/show_bug.cgi?id=275146
cadubentzen
committed
[MediaStream][GStreamer] Hardware decoder/sinks do not have src pads

calvaris
authored and
cadubentzen
committed
[GStreamer][WebRTC] Minor improvements in incoming track processor

philn
authored and
cadubentzen
committed
[GStreamer][WebRTC] Simplified pipeline for incoming tracks

philn
authored and
cadubentzen
committed
[GStreamer][WebRTC] Apply platform-specific quirks for incoming tracks

philn
authored and
cadubentzen
committed
[GStreamer][WebRTC] Emit 'track' event before resolving setRemoteDescription's promise
cadubentzen
committed
[WebRTC] Fix build with GCC 9
cadubentzen
committed

Commits on Aug 30, 2024

wip: do not set immediate-output=TRUE on westerossink
cadubentzen
committed

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GstWebRTC backports from upstream#1387

GstWebRTC backports from upstream#1387
cadubentzen wants to merge 15 commits intowpe-2.38from
cadubentzen/GstWebRTC

Commits on Aug 28, 2024

Move out of OnTrack/OnRemoveTrack libwebrtc callbacks

Incoming video track renderer should rely on the track intrinsic size https://bugs.webkit.org/show_bug.cgi?id=274093

Missing a=msid in offer after setting codec preferences https://bugs.webkit.org/show_bug.cgi?id=275157

Missing media in SDP if setConfiguration() is called after createDataChannel() or addTransceiver() https://bugs.webkit.org/show_bug.cgi?id=273318

webrtc/connection-state.html started failing after update to 1.24.0 https://bugs.webkit.org/show_bug.cgi?id=271243 <rdar://problem/125014192>

`m_maxMessageSize` should be `infinity()` instead of `max()` in `RTCSctpTransport.h`

Most of WPT webrtc/RTCSctpTransport-maxMessageSize.html tests are failing https://bugs.webkit.org/show_bug.cgi?id=274442 rdar://128444396

Add debug logging of TransceiverStates https://bugs.webkit.org/show_bug.cgi?id=275146

[MediaStream][GStreamer] Hardware decoder/sinks do not have src pads

[GStreamer][WebRTC] Minor improvements in incoming track processor

[GStreamer][WebRTC] Simplified pipeline for incoming tracks

[GStreamer][WebRTC] Apply platform-specific quirks for incoming tracks

[GStreamer][WebRTC] Emit 'track' event before resolving setRemoteDescription's promise

[WebRTC] Fix build with GCC 9

Commits on Aug 30, 2024

wip: do not set immediate-output=TRUE on westerossink

GstWebRTC backports from upstream#1387cadubentzen wants to merge 15 commits intowpe-2.38WebPlatformForEmbedded/WPEWebKit:wpe-2.38from cadubentzen/GstWebRTCWebPlatformForEmbedded/WPEWebKit:cadubentzen/GstWebRTCCopy head branch name to clipboard

Commits

Commits on Aug 28, 2024

Commits on Aug 30, 2024

GstWebRTC backports from upstream#1387
cadubentzen wants to merge 15 commits intowpe-2.38from
cadubentzen/GstWebRTC