[webkit-changes] [WebKit/WebKit] 84c90a: [GStreamer][WebRTC] Handle GObject unsigned pt pro...

[Philippe Normand]

  Branch: refs/heads/main
  Home:   https://github.com/WebKit/WebKit
  Commit: 84c90a1cdcd4d11f253d1f1c5a2ad483efa58c77
      https://github.com/WebKit/WebKit/commit/84c90a1cdcd4d11f253d1f1c5a2ad483efa58c77
  Author: Philippe Normand <philn at igalia.com>
  Date:   2025-01-23 (Thu, 23 Jan 2025)

  Changed paths:
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.h
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.h
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp
    M Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.h
    M Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp

  Log Message:
  -----------
  [GStreamer][WebRTC] Handle GObject unsigned pt properties in RTP payloaders
https://bugs.webkit.org/show_bug.cgi?id=286346

Reviewed by Xabier Rodriguez-Calvar.

The GStreamer rtp*pay2 payloaders' "pt" property is an unsigned int while on the legacy payloaders
it is an int, so both cases need to be handled. We also now emit an error early on if an attempt to
set an out-of-range value is made.

* Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.cpp:
(WebCore::GStreamerAudioRTPPacketizer::create):
(WebCore::GStreamerAudioRTPPacketizer::GStreamerAudioRTPPacketizer):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioRTPPacketizer.h:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.cpp:
(WebCore::GStreamerRTPPacketizer::GStreamerRTPPacketizer):
(WebCore::GStreamerRTPPacketizer::setPayloadType):
(WebCore::GStreamerRTPPacketizer::payloadType const):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerRTPPacketizer.h:
* Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.cpp:
(WebCore::GStreamerVideoRTPPacketizer::create):
(WebCore::GStreamerVideoRTPPacketizer::GStreamerVideoRTPPacketizer):
* Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoRTPPacketizer.h:

Canonical link: https://commits.webkit.org/289286@main



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