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607ef6db60
GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214> |
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adaptivedemux | ||
audio | ||
basecamerabinsrc | ||
codecparsers | ||
codecs | ||
d3d11 | ||
insertbin | ||
interfaces | ||
isoff | ||
mpegts | ||
opencv | ||
play | ||
player | ||
sctp | ||
transcoder | ||
uridownloader | ||
va | ||
vulkan | ||
wayland | ||
webrtc | ||
gettext.h | ||
glib-compat-private.h | ||
gst-i18n-plugin.h | ||
meson.build |