Branch: refs/heads/main
Home: https://github.com/WebKit/WebKit
Commit: 7c3380a57b31d1ab78391dba50b0549fd0287f46
https://github.com/WebKit/WebKit/commit/7c3380a57b31d1ab78391dba50b0549fd0287f46
Author: Philippe Normand <[email protected]>
Date: 2024-04-10 (Wed, 10 Apr 2024)
Changed paths:
M
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp
M Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp
M
Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp
Log Message:
-----------
[GStreamer][WebRTC] Outgoing Opus improvements
https://bugs.webkit.org/show_bug.cgi?id=272390
Reviewed by Xabier Rodriguez-Calvar.
Some WebRTC stacks expect Opus in SDP as "opus" and others "OPUS", so we are
now more relient in
that regard. Our encoder now emits packets with "perfect" timestamps, because
Chrome expects that.
Support for multi-channel is also slightly improved, by checking the
encoding-params SDP attribute.
*
LayoutTests/platform/glib/fast/mediastream/RTCPeerConnection-inspect-offer-expected.txt:
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerMediaEndpoint.cpp:
(WebCore::GStreamerMediaEndpoint::setDescription):
* Source/WebCore/Modules/mediastream/gstreamer/GStreamerWebRTCUtils.cpp:
(WebCore::capsFromRtpCapabilities):
(WebCore::capsFromSDPMedia):
*
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/277296@main
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