[Webkit-unassigned] [Bug 235885] [GStreamer] ⛱ for GstWebRTC backend

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=235885

Philippe Normand <philn at igalia.com> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
                 CC|                            |philn at igalia.com
         Depends on|                            |255485, 255773, 255614,
                   |                            |255611, 255608, 255605,
                   |                            |255606, 255613, 255612,
                   |                            |255609, 255284, 255285,
                   |                            |255391, 255099, 233731,
                   |                            |255096, 254040, 254163,
                   |                            |254212, 254164, 254150,
                   |                            |253954, 253955, 253833,
                   |                            |251428, 253159, 251146,
                   |                            |252815, 255784, 255786,
                   |                            |254529, 256042


Referenced Bugs:

https://bugs.webkit.org/show_bug.cgi?id=233731
[Bug 233731] [GStreamer] test fast/mediastream/getDisplayMedia-size.html fails
https://bugs.webkit.org/show_bug.cgi?id=251146
[Bug 251146] [GStreamer][1.22] webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html crashing
https://bugs.webkit.org/show_bug.cgi?id=251428
[Bug 251428] [GStreamer][WebRTC] Implement GstMappedRtpBuffer
https://bugs.webkit.org/show_bug.cgi?id=252815
[Bug 252815] [GStreamer][MediaStream] Unify stream collection handling with MSE
https://bugs.webkit.org/show_bug.cgi?id=253159
[Bug 253159] [GStreamer][WebRTC] Logging improvements and disable auto-header-extensions on video payloaders
https://bugs.webkit.org/show_bug.cgi?id=253833
[Bug 253833] [GStreamer][WebRTC] Ensure end-point pipeline is stopped when destroying
https://bugs.webkit.org/show_bug.cgi?id=253954
[Bug 253954] [GStreamer][WebRTC] Improve DataChannel logs
https://bugs.webkit.org/show_bug.cgi?id=253955
[Bug 253955] [GStreamer][WebRTC] StatsCollector: Update GStreamer version checks
https://bugs.webkit.org/show_bug.cgi?id=254040
[Bug 254040] [MediaStream][GStreamer] Add support for InputDeviceInfo
https://bugs.webkit.org/show_bug.cgi?id=254150
[Bug 254150] [GStreamer][WebRTC] Dedicated log categories for incoming/outgoing media sources
https://bugs.webkit.org/show_bug.cgi?id=254163
[Bug 254163] [GStreamer][WebRTC] webrtc/libwebrtc/setLocalDescriptionCrash.html fails
https://bugs.webkit.org/show_bug.cgi?id=254164
[Bug 254164] [GStreamer][WebRTC] fast/mediastream/RTCPeerConnection-inspect-{offer,answer}.html fail
https://bugs.webkit.org/show_bug.cgi?id=254212
[Bug 254212] [Gstreamer][WebRTC] webrtc/video-disabled-black.html fails
https://bugs.webkit.org/show_bug.cgi?id=254529
[Bug 254529] [GStreamer][MediaStream] fast/mediastream/MediaStream-video-element-track-stop.html fails
https://bugs.webkit.org/show_bug.cgi?id=255096
[Bug 255096] [GStreamer][WebRTC] TWCC extension not listed in SDP generated descriptions
https://bugs.webkit.org/show_bug.cgi?id=255099
[Bug 255099] [GStreamer][WebRTC] Outgoing tracks are missing SSRCs
https://bugs.webkit.org/show_bug.cgi?id=255284
[Bug 255284] [GStreamer][WebRTC] Offer/Answer handling is potentially racy
https://bugs.webkit.org/show_bug.cgi?id=255285
[Bug 255285] [GStreamer][MediaStream] Internal API improvements
https://bugs.webkit.org/show_bug.cgi?id=255391
[Bug 255391] [GStreamer][WebRTC] Crash when calling rtpSender.getParameters()
https://bugs.webkit.org/show_bug.cgi?id=255485
[Bug 255485] [GStreamer][MediaStream] fast/mediastream/getDisplayMedia-size.html hits ASSERT in Debug builds
https://bugs.webkit.org/show_bug.cgi?id=255605
[Bug 255605] [GStreamer][WebRTC] Support for data-channel contents logging
https://bugs.webkit.org/show_bug.cgi?id=255606
[Bug 255606] [GStreamer][WebRTC] Ignore DataChannel SDP media when firing incoming track events
https://bugs.webkit.org/show_bug.cgi?id=255608
[Bug 255608] [GStreamer][WebRTC] Relay webrtcbin signaling-state changes to the RTCPeerConnection
https://bugs.webkit.org/show_bug.cgi?id=255609
[Bug 255609] [GStreamer][WebRTC] Plumb exception handling from PC backend to end-point
https://bugs.webkit.org/show_bug.cgi?id=255611
[Bug 255611] [GStreamer][WebRTC] Improve debug logging in RTP sender backend
https://bugs.webkit.org/show_bug.cgi?id=255612
[Bug 255612] [GStreamer][WebRTC] Instrument the SDP ICE candidate parsing with debug logs
https://bugs.webkit.org/show_bug.cgi?id=255613
[Bug 255613] [GStreamer][WebRTC] Opus codec mime-type is incorrect in our SDP offers
https://bugs.webkit.org/show_bug.cgi?id=255614
[Bug 255614] [GStreamer][WebRTC] RTP payload type numbers are incorrect and prone to collisions in our SDP offers
https://bugs.webkit.org/show_bug.cgi?id=255773
[Bug 255773] [GStreamer][WebRTC] fast/mediastream/RTCPeerConnection-inspect-answer.html fails
https://bugs.webkit.org/show_bug.cgi?id=255784
[Bug 255784] [GStreamer][MediaStream] Over-allocation of buffers sourced from the WebAudio bus
https://bugs.webkit.org/show_bug.cgi?id=255786
[Bug 255786] [GStreamer][Debug] mediarecorder tests hitting ASSERT
https://bugs.webkit.org/show_bug.cgi?id=256042
[Bug 256042] [GStreamer][MediaStream] Make the source element behave as a stream from urisourcebin scope
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