Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 06463409ee8aa2a75c70ac1fcc8abb4c363755c2
https://github.com/WebKit/WebKit/commit/06463409ee8aa2a75c70ac1fcc8abb4c363755c2
Author: Carlos Bentzen <[email protected]>
Date: 2024-06-19 (Wed, 19 Jun 2024)
Changed paths:
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
M Source/WebCore/platform/audio/gstreamer/AudioEncoderGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp
M Source/WebCore/platform/graphics/gstreamer/VideoEncoderGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
Log Message:
-----------
[GStreamer] Make use of gstStructureGet<T>
https://bugs.webkit.org/show_bug.cgi?id=275658
Reviewed by Philippe Normand.
Make use of gstStructureGet<T> where it makes sense.
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::processStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerStatsCollector.cpp:
(WebCore::RTCStatsReport::Stats::Stats):
(WebCore::RTCStatsReport::RtpStreamStats::RtpStreamStats):
(WebCore::RTCStatsReport::SentRtpStreamStats::SentRtpStreamStats):
(WebCore::RTCStatsReport::CodecStats::CodecStats):
(WebCore::RTCStatsReport::ReceivedRtpStreamStats::ReceivedRtpStreamStats):
(WebCore::RTCStatsReport::RemoteInboundRtpStreamStats::RemoteInboundRtpStreamStats):
(WebCore::RTCStatsReport::RemoteOutboundRtpStreamStats::RemoteOutboundRtpStreamStats):
(WebCore::RTCStatsReport::InboundRtpStreamStats::InboundRtpStreamStats):
(WebCore::RTCStatsReport::OutboundRtpStreamStats::OutboundRtpStreamStats):
(WebCore::RTCStatsReport::PeerConnectionStats::PeerConnectionStats):
(WebCore::RTCStatsReport::IceCandidateStats::IceCandidateStats):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::toRTCEncodingParameters):
* Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
(WebCore::maximumNumberOfOutputChannels):
* Source/WebCore/platform/audio/gstreamer/AudioEncoderGStreamer.cpp:
(WebCore::GStreamerInternalAudioEncoder::GStreamerInternalAudioEncoder):
* Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp:
(WebCore::AudioTrackPrivateGStreamer::updateConfigurationFromCaps):
* Source/WebCore/platform/graphics/gstreamer/VideoEncoderGStreamer.cpp:
(WebCore::retrieveTemporalIndex):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingAudioSourceGStreamer::setPayloadType):
*
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp:
(WebCore::RealtimeOutgoingVideoSourceGStreamer::setPayloadType):
Canonical link: https://commits.webkit.org/280177@main
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