[webkit-changes] [WebKit/WebKit] d6d7f3: [GStreamer][WebRTC] Caps negotiation failure when ...

[Philippe Normand]

  Branch: refs/heads/main
  Home:   https://github.com/WebKit/WebKit
  Commit: d6d7f3edf374504363fece083b65e38793b27df0
      https://github.com/WebKit/WebKit/commit/d6d7f3edf374504363fece083b65e38793b27df0
  Author: Philippe Normand <philn at igalia.com>
  Date:   2024-05-14 (Tue, 14 May 2024)

  Changed paths:
    M LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt
    M LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
    M Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp

  Log Message:
  -----------
  [GStreamer][WebRTC] Caps negotiation failure when ssrc-audio-level RTP extension is requested
https://bugs.webkit.org/show_bug.cgi?id=271519

Reviewed by Xabier Rodriguez-Calvar.

When not present in caps, the vad support of the ssrc-audio-level extension should be enabled. In
order to prevent caps negotiation issues with downstream, explicitely set it.

* LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-answer-expected.txt:
* LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt:
* Source/WebCore/platform/graphics/gstreamer/GStreamerRegistryScanner.h:
* Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType):

Canonical link: https://commits.webkit.org/278738@main



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