eb789072b4
https://bugs.webkit.org/show_bug.cgi?id=218436 LayoutTests/imported/w3c: Reviewed by Eric Carlson. * web-platform-tests/webrtc/RTCPeerConnection-createDataChannel-expected.txt: * web-platform-tests/webrtc/RTCPeerConnection-transceivers.https-expected.txt: * web-platform-tests/webrtc/RTCRtpTransceiver-stop-expected.txt: Source/ThirdParty/libwebrtc: Reviewed by Eric Carlson. * CMakeLists.txt: * Configurations/libwebrtc.iOS.exp: * Configurations/libwebrtc.iOSsim.exp: * Configurations/libwebrtc.mac.exp: * Source/webrtc: Resynced. * WebKit/libwebrtc-m87-diff: Added. * libwebrtc.xcodeproj/project.pbxproj: Source/WebCore: Reviewed by Eric Carlson. Move from deprecated to new APIs. Covered by existing tests. * Modules/mediastream/libwebrtc/LibWebRTCRtpTransceiverBackend.cpp: (WebCore::LibWebRTCRtpTransceiverBackend::setDirection): (WebCore::LibWebRTCRtpTransceiverBackend::stop): * Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp: (WebCore::fillRTCDataChannelStats): * platform/mediastream/RealtimeOutgoingVideoSource.h: * platform/mediastream/gstreamer/GStreamerVideoFrameLibWebRTC.cpp: (WebCore::GStreamerVideoFrameLibWebRTC::ToI420): * platform/mediastream/gstreamer/GStreamerVideoFrameLibWebRTC.h: * platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp: (WebCore::GStreamerEncodedImageBuffer::create): (WebCore::GStreamerEncodedImageBuffer::GStreamerEncodedImageBuffer): (WebCore::GStreamerVideoEncoderFactory::CreateVideoEncoder): (WebCore::GStreamerVideoEncoder::Fragmentize): Deleted. * platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.h: * testing/MockLibWebRTCPeerConnection.h: Source/WebKit: Reviewed by Eric Carlson. Update code now that fragmentation headers are computed at packetization time. * Configurations/WebKit.xcconfig: * GPUProcess/webrtc/LibWebRTCCodecsProxy.mm: (WebKit::LibWebRTCCodecsProxy::createEncoder): * Scripts/webkit/messages.py: * Shared/RTCNetwork.h: * WebProcess/GPU/webrtc/LibWebRTCCodecs.cpp: (WebKit::LibWebRTCCodecs::completedEncoding): * WebProcess/GPU/webrtc/LibWebRTCCodecs.h: * WebProcess/GPU/webrtc/LibWebRTCCodecs.messages.in: LayoutTests: Reviewed by Eric Carlson. We now have to explicitly support all packetization modes, which is similar to what Chrome is doing. * webrtc/h264-packetization-mode.html: Canonical link: https://commits.webkit.org/231430@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@269642 268f45cc-cd09-0410-ab3c-d52691b4dbfc |
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|---|---|---|
| .. | ||
| 0001-Close-sockets-with-file-descriptors-above-FD_SETSIZE.patch | ||
| 0001-Fix-RTCVideoDecoderH264 | ||
| 0001-Fix-RTCVideoEncoderH264 | ||
| 0001-Fix-ctype.h-include.patch | ||
| 001-GTK-changes.patch | ||
| 0001-fix-195930.patch | ||
| 0001-fix-197301.patch | ||
| 0001-fix-216314.patch | ||
| 0001-fix-fd-clr.patch | ||
| 0002-Fixing-usrctp-library-compilation-errors.patch | ||
| 0002-libwebrtc-fix-unnecessary-copy-of-for-loop-variables.diff | ||
| 0003-libwebrtc-fix-vp8e_mr_alloc_mem-leak.diff | ||
| libwebrtc-m87-diff | ||
| libwebrtc.diff | ||
| patch-boringssl | ||
| patch-opus.diff | ||
| patch-usrsctp.diff | ||