mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
synced 2024-06-27 02:20:41 +00:00
973 lines
47 KiB
XML
973 lines
47 KiB
XML
<?xml version="1.0"?>
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<!-- This file was automatically generated from C sources - DO NOT EDIT!
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To affect the contents of this file, edit the original C definitions,
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and/or use gtk-doc annotations. -->
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<repository xmlns="http://www.gtk.org/introspection/core/1.0" xmlns:c="http://www.gtk.org/introspection/c/1.0" xmlns:glib="http://www.gtk.org/introspection/glib/1.0" version="1.2">
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<include name="Gst" version="1.0"/>
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<include name="GstSdp" version="1.0"/>
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<package name="gstreamer-webrtc-1.0"/>
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<c:include name="gst/webrtc/webrtc.h"/>
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<namespace name="GstWebRTC" version="1.0" shared-library="libgstwebrtc-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst">
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<function-macro name="IS_WEBRTC_DTLS_TRANSPORT" c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_DTLS_TRANSPORT_CLASS" c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_ICE_TRANSPORT" c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_ICE_TRANSPORT_CLASS" c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_RECEIVER" c:identifier="GST_IS_WEBRTC_RTP_RECEIVER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_RECEIVER_CLASS" c:identifier="GST_IS_WEBRTC_RTP_RECEIVER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_SENDER" c:identifier="GST_IS_WEBRTC_RTP_SENDER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_SENDER_CLASS" c:identifier="GST_IS_WEBRTC_RTP_SENDER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER" c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER_CLASS" c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT" c:identifier="GST_WEBRTC_DTLS_TRANSPORT" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT_CLASS" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_DTLS_TRANSPORT_GET_CLASS" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT" c:identifier="GST_WEBRTC_ICE_TRANSPORT" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT_CLASS" c:identifier="GST_WEBRTC_ICE_TRANSPORT_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_ICE_TRANSPORT_GET_CLASS" c:identifier="GST_WEBRTC_ICE_TRANSPORT_GET_CLASS" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER" c:identifier="GST_WEBRTC_RTP_RECEIVER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER_CLASS" c:identifier="GST_WEBRTC_RTP_RECEIVER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_RECEIVER_GET_CLASS" c:identifier="GST_WEBRTC_RTP_RECEIVER_GET_CLASS" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER" c:identifier="GST_WEBRTC_RTP_SENDER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER_CLASS" c:identifier="GST_WEBRTC_RTP_SENDER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_SENDER_GET_CLASS" c:identifier="GST_WEBRTC_RTP_SENDER_GET_CLASS" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER_CLASS" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS" introspectable="0">
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="WEBRTC_RTP_TRANSCEIVER_GET_CLASS" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS" introspectable="0">
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<enumeration name="WebRTCBundlePolicy" version="1.16" glib:type-name="GstWebRTCBundlePolicy" glib:get-type="gst_webrtc_bundle_policy_get_type" c:type="GstWebRTCBundlePolicy">
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<doc xml:space="preserve">GST_WEBRTC_BUNDLE_POLICY_NONE: none
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GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
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GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
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GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
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See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
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for more information.</doc>
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<member name="none" value="0" c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE" glib:nick="none">
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</member>
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<member name="balanced" value="1" c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED" glib:nick="balanced">
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</member>
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<member name="max_compat" value="2" c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT" glib:nick="max-compat">
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</member>
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<member name="max_bundle" value="3" c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE" glib:nick="max-bundle">
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</member>
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</enumeration>
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<enumeration name="WebRTCDTLSSetup" glib:type-name="GstWebRTCDTLSSetup" glib:get-type="gst_webrtc_dtls_setup_get_type" c:type="GstWebRTCDTLSSetup">
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<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
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GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
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GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
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GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
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<member name="none" value="0" c:identifier="GST_WEBRTC_DTLS_SETUP_NONE" glib:nick="none">
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</member>
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<member name="actpass" value="1" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS" glib:nick="actpass">
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</member>
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<member name="active" value="2" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE" glib:nick="active">
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</member>
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<member name="passive" value="3" c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE" glib:nick="passive">
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</member>
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</enumeration>
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<class name="WebRTCDTLSTransport" c:symbol-prefix="webrtc_dtls_transport" c:type="GstWebRTCDTLSTransport" parent="Gst.Object" glib:type-name="GstWebRTCDTLSTransport" glib:get-type="gst_webrtc_dtls_transport_get_type" glib:type-struct="WebRTCDTLSTransportClass">
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<constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
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<return-value transfer-ownership="none">
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</return-value>
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<parameters>
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<parameter name="session_id" transfer-ownership="none">
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<type name="guint" c:type="guint"/>
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</parameter>
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<parameter name="rtcp" transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</parameter>
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</parameters>
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</constructor>
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<method name="set_transport" c:identifier="gst_webrtc_dtls_transport_set_transport">
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<return-value transfer-ownership="none">
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<type name="none" c:type="void"/>
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</return-value>
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<parameters>
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<instance-parameter name="transport" transfer-ownership="none">
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<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
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</instance-parameter>
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<parameter name="ice" transfer-ownership="none">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</parameter>
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</parameters>
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</method>
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<property name="certificate" writable="1" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</property>
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<property name="client" writable="1" transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</property>
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<property name="remote-certificate" transfer-ownership="none">
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<type name="utf8" c:type="gchar*"/>
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</property>
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<property name="rtcp" writable="1" construct-only="1" transfer-ownership="none">
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<type name="gboolean" c:type="gboolean"/>
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</property>
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<property name="session-id" writable="1" construct-only="1" transfer-ownership="none">
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<type name="guint" c:type="guint"/>
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</property>
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<property name="state" transfer-ownership="none">
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<type name="WebRTCDTLSTransportState"/>
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</property>
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<property name="transport" transfer-ownership="none">
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<type name="WebRTCICETransport"/>
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</property>
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<field name="parent">
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<type name="Gst.Object" c:type="GstObject"/>
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</field>
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<field name="transport">
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<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
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</field>
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<field name="state">
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<type name="WebRTCDTLSTransportState" c:type="GstWebRTCDTLSTransportState"/>
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</field>
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<field name="is_rtcp">
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<type name="gboolean" c:type="gboolean"/>
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</field>
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<field name="client">
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<type name="gboolean" c:type="gboolean"/>
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</field>
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<field name="session_id">
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<type name="guint" c:type="guint"/>
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</field>
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<field name="dtlssrtpenc">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="dtlssrtpdec">
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<type name="Gst.Element" c:type="GstElement*"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</class>
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<record name="WebRTCDTLSTransportClass" c:type="GstWebRTCDTLSTransportClass" glib:is-gtype-struct-for="WebRTCDTLSTransport">
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<field name="parent_class">
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<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
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</field>
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<field name="_padding">
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<array zero-terminated="0" fixed-size="4">
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<type name="gpointer" c:type="gpointer"/>
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</array>
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</field>
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</record>
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<enumeration name="WebRTCDTLSTransportState" glib:type-name="GstWebRTCDTLSTransportState" glib:get-type="gst_webrtc_dtls_transport_state_get_type" c:type="GstWebRTCDTLSTransportState">
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<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
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<member name="new" value="0" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" glib:nick="new">
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</member>
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<member name="closed" value="1" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" glib:nick="closed">
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</member>
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<member name="failed" value="2" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" glib:nick="failed">
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</member>
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<member name="connecting" value="3" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" glib:nick="connecting">
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</member>
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<member name="connected" value="4" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" glib:nick="connected">
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</member>
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</enumeration>
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<enumeration name="WebRTCDataChannelState" version="1.16" glib:type-name="GstWebRTCDataChannelState" glib:get-type="gst_webrtc_data_channel_state_get_type" c:type="GstWebRTCDataChannelState">
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<doc xml:space="preserve">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
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GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
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GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink></doc>
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<member name="new" value="0" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW" glib:nick="new">
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</member>
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<member name="connecting" value="1" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING" glib:nick="connecting">
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</member>
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<member name="open" value="2" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN" glib:nick="open">
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</member>
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<member name="closing" value="3" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING" glib:nick="closing">
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</member>
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<member name="closed" value="4" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED" glib:nick="closed">
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</member>
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</enumeration>
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<enumeration name="WebRTCFECType" version="1.14.1" glib:type-name="GstWebRTCFECType" glib:get-type="gst_webrtc_fec_type_get_type" c:type="GstWebRTCFECType">
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<member name="none" value="0" c:identifier="GST_WEBRTC_FEC_TYPE_NONE" glib:nick="none">
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<doc xml:space="preserve">none</doc>
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</member>
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<member name="ulp_red" value="1" c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED" glib:nick="ulp-red">
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<doc xml:space="preserve">ulpfec + red</doc>
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</member>
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</enumeration>
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<enumeration name="WebRTCICEComponent" glib:type-name="GstWebRTCICEComponent" glib:get-type="gst_webrtc_ice_component_get_type" c:type="GstWebRTCICEComponent">
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<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
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GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
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<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP" glib:nick="rtp">
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</member>
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<member name="rtcp" value="1" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP" glib:nick="rtcp">
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</member>
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</enumeration>
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<enumeration name="WebRTCICEConnectionState" glib:type-name="GstWebRTCICEConnectionState" glib:get-type="gst_webrtc_ice_connection_state_get_type" c:type="GstWebRTCICEConnectionState">
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<doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
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GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
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GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
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GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
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GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
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GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
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GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink></doc>
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<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" glib:nick="new">
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</member>
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<member name="checking" value="1" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" glib:nick="checking">
|
|
</member>
|
|
<member name="connected" value="2" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" glib:nick="connected">
|
|
</member>
|
|
<member name="completed" value="3" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" glib:nick="completed">
|
|
</member>
|
|
<member name="failed" value="4" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" glib:nick="failed">
|
|
</member>
|
|
<member name="disconnected" value="5" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
|
|
</member>
|
|
<member name="closed" value="6" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" glib:nick="closed">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEGatheringState" glib:type-name="GstWebRTCICEGatheringState" glib:get-type="gst_webrtc_ice_gathering_state_get_type" c:type="GstWebRTCICEGatheringState">
|
|
<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
|
|
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
|
|
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink></doc>
|
|
<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW" glib:nick="new">
|
|
</member>
|
|
<member name="gathering" value="1" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" glib:nick="gathering">
|
|
</member>
|
|
<member name="complete" value="2" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" glib:nick="complete">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICERole" glib:type-name="GstWebRTCICERole" glib:get-type="gst_webrtc_ice_role_get_type" c:type="GstWebRTCICERole">
|
|
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
|
|
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
|
|
<member name="controlled" value="0" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED" glib:nick="controlled">
|
|
</member>
|
|
<member name="controlling" value="1" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING" glib:nick="controlling">
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCICETransport" c:symbol-prefix="webrtc_ice_transport" c:type="GstWebRTCICETransport" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCICETransport" glib:get-type="gst_webrtc_ice_transport_get_type" glib:type-struct="WebRTCICETransportClass">
|
|
|
|
<virtual-method name="gather_candidates">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="connection_state_change" c:identifier="gst_webrtc_ice_transport_connection_state_change">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="ice" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</instance-parameter>
|
|
<parameter name="new_state" transfer-ownership="none">
|
|
<type name="WebRTCICEConnectionState" c:type="GstWebRTCICEConnectionState"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="gathering_state_change" c:identifier="gst_webrtc_ice_transport_gathering_state_change">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="ice" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</instance-parameter>
|
|
<parameter name="new_state" transfer-ownership="none">
|
|
<type name="WebRTCICEGatheringState" c:type="GstWebRTCICEGatheringState"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="new_candidate" c:identifier="gst_webrtc_ice_transport_new_candidate">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="ice" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</instance-parameter>
|
|
<parameter name="stream_id" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="component" transfer-ownership="none">
|
|
<type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
|
|
</parameter>
|
|
<parameter name="attr" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="selected_pair_change" c:identifier="gst_webrtc_ice_transport_selected_pair_change">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="ice" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="component" writable="1" construct-only="1" transfer-ownership="none">
|
|
<type name="WebRTCICEComponent"/>
|
|
</property>
|
|
<property name="gathering-state" transfer-ownership="none">
|
|
<type name="WebRTCICEGatheringState"/>
|
|
</property>
|
|
<property name="state" transfer-ownership="none">
|
|
<type name="WebRTCICEConnectionState"/>
|
|
</property>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="role">
|
|
<type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
|
|
</field>
|
|
<field name="component">
|
|
<type name="WebRTCICEComponent" c:type="GstWebRTCICEComponent"/>
|
|
</field>
|
|
<field name="state">
|
|
<type name="WebRTCICEConnectionState" c:type="GstWebRTCICEConnectionState"/>
|
|
</field>
|
|
<field name="gathering_state">
|
|
<type name="WebRTCICEGatheringState" c:type="GstWebRTCICEGatheringState"/>
|
|
</field>
|
|
<field name="src">
|
|
<type name="Gst.Element" c:type="GstElement*"/>
|
|
</field>
|
|
<field name="sink">
|
|
<type name="Gst.Element" c:type="GstElement*"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<glib:signal name="on-new-candidate" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="object" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</glib:signal>
|
|
<glib:signal name="on-selected-candidate-pair-change" when="last">
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
</glib:signal>
|
|
</class>
|
|
<record name="WebRTCICETransportClass" c:type="GstWebRTCICETransportClass" glib:is-gtype-struct-for="WebRTCICETransport">
|
|
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="gather_candidates">
|
|
<callback name="gather_candidates">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="WebRTCICETransportPolicy" version="1.16" glib:type-name="GstWebRTCICETransportPolicy" glib:get-type="gst_webrtc_ice_transport_policy_get_type" c:type="GstWebRTCICETransportPolicy">
|
|
<doc xml:space="preserve">GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
|
|
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
|
|
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
|
|
for more information.</doc>
|
|
<member name="all" value="0" c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL" glib:nick="all">
|
|
</member>
|
|
<member name="relay" value="1" c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY" glib:nick="relay">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCPeerConnectionState" glib:type-name="GstWebRTCPeerConnectionState" glib:get-type="gst_webrtc_peer_connection_state_get_type" c:type="GstWebRTCPeerConnectionState">
|
|
<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink></doc>
|
|
<member name="new" value="0" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" glib:nick="new">
|
|
</member>
|
|
<member name="connecting" value="1" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" glib:nick="connecting">
|
|
</member>
|
|
<member name="connected" value="2" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" glib:nick="connected">
|
|
</member>
|
|
<member name="disconnected" value="3" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
|
|
</member>
|
|
<member name="failed" value="4" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" glib:nick="failed">
|
|
</member>
|
|
<member name="closed" value="5" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" glib:nick="closed">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCPriorityType" version="1.16" glib:type-name="GstWebRTCPriorityType" glib:get-type="gst_webrtc_priority_type_get_type" c:type="GstWebRTCPriorityType">
|
|
<doc xml:space="preserve">GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
|
|
GST_WEBRTC_PRIORITY_TYPE_LOW: low
|
|
GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
|
|
GST_WEBRTC_PRIORITY_TYPE_HIGH: high
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink></doc>
|
|
<member name="very_low" value="1" c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW" glib:nick="very-low">
|
|
</member>
|
|
<member name="low" value="2" c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW" glib:nick="low">
|
|
</member>
|
|
<member name="medium" value="3" c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM" glib:nick="medium">
|
|
</member>
|
|
<member name="high" value="4" c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH" glib:nick="high">
|
|
</member>
|
|
</enumeration>
|
|
<class name="WebRTCRTPReceiver" c:symbol-prefix="webrtc_rtp_receiver" c:type="GstWebRTCRTPReceiver" parent="Gst.Object" glib:type-name="GstWebRTCRTPReceiver" glib:get-type="gst_webrtc_rtp_receiver_get_type" glib:type-struct="WebRTCRTPReceiverClass">
|
|
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</return-value>
|
|
</constructor>
|
|
<method name="set_rtcp_transport" c:identifier="gst_webrtc_rtp_receiver_set_rtcp_transport">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="receiver" transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_transport" c:identifier="gst_webrtc_rtp_receiver_set_transport">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="receiver" transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="rtcp_transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPReceiverClass" c:type="GstWebRTCRTPReceiverClass" glib:is-gtype-struct-for="WebRTCRTPReceiver">
|
|
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="WebRTCRTPSender" c:symbol-prefix="webrtc_rtp_sender" c:type="GstWebRTCRTPSender" parent="Gst.Object" glib:type-name="GstWebRTCRTPSender" glib:get-type="gst_webrtc_rtp_sender_get_type" glib:type-struct="WebRTCRTPSenderClass">
|
|
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</return-value>
|
|
</constructor>
|
|
<method name="set_rtcp_transport" c:identifier="gst_webrtc_rtp_sender_set_rtcp_transport">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sender" transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_transport" c:identifier="gst_webrtc_rtp_sender_set_transport">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sender" transfer-ownership="none">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</instance-parameter>
|
|
<parameter name="transport" transfer-ownership="none">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="rtcp_transport">
|
|
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
|
|
</field>
|
|
<field name="send_encodings">
|
|
<array name="GLib.Array" c:type="GArray*">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPSenderClass" c:type="GstWebRTCRTPSenderClass" glib:is-gtype-struct-for="WebRTCRTPSender">
|
|
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="WebRTCRTPTransceiver" c:symbol-prefix="webrtc_rtp_transceiver" c:type="GstWebRTCRTPTransceiver" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCRTPTransceiver" glib:get-type="gst_webrtc_rtp_transceiver_get_type" glib:type-struct="WebRTCRTPTransceiverClass">
|
|
|
|
<property name="mlineindex" writable="1" construct-only="1" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</property>
|
|
<property name="receiver" writable="1" construct-only="1" transfer-ownership="none">
|
|
<type name="WebRTCRTPReceiver"/>
|
|
</property>
|
|
<property name="sender" writable="1" construct-only="1" transfer-ownership="none">
|
|
<type name="WebRTCRTPSender"/>
|
|
</property>
|
|
<field name="parent">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="mline">
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="mid">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</field>
|
|
<field name="stopped">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="sender">
|
|
<type name="WebRTCRTPSender" c:type="GstWebRTCRTPSender*"/>
|
|
</field>
|
|
<field name="receiver">
|
|
<type name="WebRTCRTPReceiver" c:type="GstWebRTCRTPReceiver*"/>
|
|
</field>
|
|
<field name="direction">
|
|
<type name="WebRTCRTPTransceiverDirection" c:type="GstWebRTCRTPTransceiverDirection"/>
|
|
</field>
|
|
<field name="current_direction">
|
|
<type name="WebRTCRTPTransceiverDirection" c:type="GstWebRTCRTPTransceiverDirection"/>
|
|
</field>
|
|
<field name="codec_preferences">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="WebRTCRTPTransceiverClass" c:type="GstWebRTCRTPTransceiverClass" glib:is-gtype-struct-for="WebRTCRTPTransceiver">
|
|
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="_padding">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="WebRTCRTPTransceiverDirection" glib:type-name="GstWebRTCRTPTransceiverDirection" glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type" c:type="GstWebRTCRTPTransceiverDirection">
|
|
<member name="none" value="0" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" glib:nick="none">
|
|
</member>
|
|
<member name="inactive" value="1" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" glib:nick="inactive">
|
|
</member>
|
|
<member name="sendonly" value="2" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" glib:nick="sendonly">
|
|
</member>
|
|
<member name="recvonly" value="3" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" glib:nick="recvonly">
|
|
</member>
|
|
<member name="sendrecv" value="4" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" glib:nick="sendrecv">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCSCTPTransportState" version="1.16" glib:type-name="GstWebRTCSCTPTransportState" glib:get-type="gst_webrtc_sctp_transport_state_get_type" c:type="GstWebRTCSCTPTransportState">
|
|
<doc xml:space="preserve">GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
|
|
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
|
|
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
|
|
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink></doc>
|
|
<member name="new" value="0" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW" glib:nick="new">
|
|
</member>
|
|
<member name="connecting" value="1" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING" glib:nick="connecting">
|
|
</member>
|
|
<member name="connected" value="2" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED" glib:nick="connected">
|
|
</member>
|
|
<member name="closed" value="3" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED" glib:nick="closed">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCSDPType" glib:type-name="GstWebRTCSDPType" glib:get-type="gst_webrtc_sdp_type_get_type" c:type="GstWebRTCSDPType">
|
|
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
|
|
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
|
|
GST_WEBRTC_SDP_TYPE_ANSWER: answer
|
|
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink></doc>
|
|
<member name="offer" value="1" c:identifier="GST_WEBRTC_SDP_TYPE_OFFER" glib:nick="offer">
|
|
</member>
|
|
<member name="pranswer" value="2" c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER" glib:nick="pranswer">
|
|
</member>
|
|
<member name="answer" value="3" c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER" glib:nick="answer">
|
|
</member>
|
|
<member name="rollback" value="4" c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK" glib:nick="rollback">
|
|
</member>
|
|
<function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
|
|
recognized.</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</enumeration>
|
|
<record name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription" glib:type-name="GstWebRTCSessionDescription" glib:get-type="gst_webrtc_session_description_get_type" c:symbol-prefix="webrtc_session_description">
|
|
<doc xml:space="preserve">See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink></doc>
|
|
|
|
<field name="type" writable="1">
|
|
<doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</field>
|
|
<field name="sdp" writable="1">
|
|
<doc xml:space="preserve">the #GstSDPMessage of the description</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</field>
|
|
<constructor name="new" c:identifier="gst_webrtc_session_description_new">
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a new #GstWebRTCSessionDescription from @type
|
|
and @sdp</doc>
|
|
<type name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
<parameter name="sdp" transfer-ownership="full">
|
|
<doc xml:space="preserve">a #GstSDPMessage</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="copy" c:identifier="gst_webrtc_session_description_copy">
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a new copy of @src</doc>
|
|
<type name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription" c:type="const GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free" c:identifier="gst_webrtc_session_description_free">
|
|
<doc xml:space="preserve">Free @desc and all associated resources</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="desc" transfer-ownership="full">
|
|
<doc xml:space="preserve">a #GstWebRTCSessionDescription</doc>
|
|
<type name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<enumeration name="WebRTCSignalingState" glib:type-name="GstWebRTCSignalingState" glib:get-type="gst_webrtc_signaling_state_get_type" c:type="GstWebRTCSignalingState">
|
|
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
|
|
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
|
|
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink></doc>
|
|
<member name="stable" value="0" c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE" glib:nick="stable">
|
|
</member>
|
|
<member name="closed" value="1" c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED" glib:nick="closed">
|
|
</member>
|
|
<member name="have_local_offer" value="2" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" glib:nick="have-local-offer">
|
|
</member>
|
|
<member name="have_remote_offer" value="3" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" glib:nick="have-remote-offer">
|
|
</member>
|
|
<member name="have_local_pranswer" value="4" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" glib:nick="have-local-pranswer">
|
|
</member>
|
|
<member name="have_remote_pranswer" value="5" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" glib:nick="have-remote-pranswer">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCStatsType" glib:type-name="GstWebRTCStatsType" glib:get-type="gst_webrtc_stats_type_get_type" c:type="GstWebRTCStatsType">
|
|
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
|
|
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
|
|
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
|
|
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
|
|
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
|
|
GST_WEBRTC_STATS_CSRC: csrc
|
|
GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
|
|
GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
|
|
GST_WEBRTC_STATS_STREAM: stream
|
|
GST_WEBRTC_STATS_TRANSPORT: transport
|
|
GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
|
|
GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
|
|
GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
|
|
GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
|
|
<member name="codec" value="1" c:identifier="GST_WEBRTC_STATS_CODEC" glib:nick="codec">
|
|
</member>
|
|
<member name="inbound_rtp" value="2" c:identifier="GST_WEBRTC_STATS_INBOUND_RTP" glib:nick="inbound-rtp">
|
|
</member>
|
|
<member name="outbound_rtp" value="3" c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP" glib:nick="outbound-rtp">
|
|
</member>
|
|
<member name="remote_inbound_rtp" value="4" c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" glib:nick="remote-inbound-rtp">
|
|
</member>
|
|
<member name="remote_outbound_rtp" value="5" c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" glib:nick="remote-outbound-rtp">
|
|
</member>
|
|
<member name="csrc" value="6" c:identifier="GST_WEBRTC_STATS_CSRC" glib:nick="csrc">
|
|
</member>
|
|
<member name="peer_connection" value="7" c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION" glib:nick="peer-connection">
|
|
</member>
|
|
<member name="data_channel" value="8" c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL" glib:nick="data-channel">
|
|
</member>
|
|
<member name="stream" value="9" c:identifier="GST_WEBRTC_STATS_STREAM" glib:nick="stream">
|
|
</member>
|
|
<member name="transport" value="10" c:identifier="GST_WEBRTC_STATS_TRANSPORT" glib:nick="transport">
|
|
</member>
|
|
<member name="candidate_pair" value="11" c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR" glib:nick="candidate-pair">
|
|
</member>
|
|
<member name="local_candidate" value="12" c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE" glib:nick="local-candidate">
|
|
</member>
|
|
<member name="remote_candidate" value="13" c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE" glib:nick="remote-candidate">
|
|
</member>
|
|
<member name="certificate" value="14" c:identifier="GST_WEBRTC_STATS_CERTIFICATE" glib:nick="certificate">
|
|
</member>
|
|
</enumeration>
|
|
<function name="webrtc_sdp_type_to_string" c:identifier="gst_webrtc_sdp_type_to_string" moved-to="WebRTCSDPType.to_string">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
|
|
recognized.</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="type" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|