mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
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1471 lines
58 KiB
XML
1471 lines
58 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 version="1.2"
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xmlns="http://www.gtk.org/introspection/core/1.0"
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xmlns:c="http://www.gtk.org/introspection/c/1.0"
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xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
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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"
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version="1.0"
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shared-library="libgstwebrtc-1.0.so.0"
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c:identifier-prefixes="Gst"
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c:symbol-prefixes="gst">
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<function-macro name="IS_WEBRTC_DTLS_TRANSPORT"
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c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT"
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introspectable="0">
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<source-position filename="dtlstransport.h" line="33"/>
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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"
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c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="dtlstransport.h" line="35"/>
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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"
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c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT"
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introspectable="0">
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<source-position filename="icetransport.h" line="32"/>
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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"
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c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="icetransport.h" line="34"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_RECEIVER"
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introspectable="0">
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<source-position filename="rtpreceiver.h" line="33"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_RECEIVER_CLASS"
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introspectable="0">
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<source-position filename="rtpreceiver.h" line="35"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_SENDER"
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introspectable="0">
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<source-position filename="rtpsender.h" line="33"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_SENDER_CLASS"
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introspectable="0">
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<source-position filename="rtpsender.h" line="35"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER"
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introspectable="0">
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<source-position filename="rtptransceiver.h" line="34"/>
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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"
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c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
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introspectable="0">
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<source-position filename="rtptransceiver.h" line="36"/>
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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"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT"
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introspectable="0">
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<source-position filename="dtlstransport.h" line="32"/>
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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"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="dtlstransport.h" line="34"/>
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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"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS"
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introspectable="0">
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<source-position filename="dtlstransport.h" line="36"/>
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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"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT"
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introspectable="0">
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<source-position filename="icetransport.h" line="31"/>
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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"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT_CLASS"
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introspectable="0">
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<source-position filename="icetransport.h" line="33"/>
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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"
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c:identifier="GST_WEBRTC_ICE_TRANSPORT_GET_CLASS"
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introspectable="0">
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<source-position filename="icetransport.h" line="35"/>
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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"
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c:identifier="GST_WEBRTC_RTP_RECEIVER"
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introspectable="0">
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<source-position filename="rtpreceiver.h" line="32"/>
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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"
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c:identifier="GST_WEBRTC_RTP_RECEIVER_CLASS"
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introspectable="0">
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<source-position filename="rtpreceiver.h" line="34"/>
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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"
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c:identifier="GST_WEBRTC_RTP_RECEIVER_GET_CLASS"
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introspectable="0">
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<source-position filename="rtpreceiver.h" line="36"/>
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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"
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c:identifier="GST_WEBRTC_RTP_SENDER"
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introspectable="0">
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<source-position filename="rtpsender.h" line="32"/>
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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"
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c:identifier="GST_WEBRTC_RTP_SENDER_CLASS"
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introspectable="0">
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<source-position filename="rtpsender.h" line="34"/>
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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"
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c:identifier="GST_WEBRTC_RTP_SENDER_GET_CLASS"
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introspectable="0">
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<source-position filename="rtpsender.h" line="36"/>
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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"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER"
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introspectable="0">
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<source-position filename="rtptransceiver.h" line="33"/>
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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"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS"
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introspectable="0">
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<source-position filename="rtptransceiver.h" line="35"/>
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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"
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c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
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introspectable="0">
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<source-position filename="rtptransceiver.h" line="37"/>
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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"
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version="1.16"
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glib:type-name="GstWebRTCBundlePolicy"
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glib:get-type="gst_webrtc_bundle_policy_get_type"
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c:type="GstWebRTCBundlePolicy">
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<doc xml:space="preserve"
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filename="webrtc_fwd.h"
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line="332">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"
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value="0"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE"
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glib:nick="none">
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</member>
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<member name="balanced"
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value="1"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
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glib:nick="balanced">
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</member>
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<member name="max_compat"
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value="2"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
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glib:nick="max-compat">
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</member>
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<member name="max_bundle"
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value="3"
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c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
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glib:nick="max-bundle">
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</member>
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</enumeration>
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<enumeration name="WebRTCDTLSSetup"
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glib:type-name="GstWebRTCDTLSSetup"
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glib:get-type="gst_webrtc_dtls_setup_get_type"
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c:type="GstWebRTCDTLSSetup">
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<doc xml:space="preserve"
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filename="webrtc_fwd.h"
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line="210">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"
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value="0"
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c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
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glib:nick="none">
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</member>
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<member name="actpass"
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value="1"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
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glib:nick="actpass">
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</member>
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<member name="active"
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value="2"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
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glib:nick="active">
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</member>
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<member name="passive"
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value="3"
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c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
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glib:nick="passive">
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</member>
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</enumeration>
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<class name="WebRTCDTLSTransport"
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c:symbol-prefix="webrtc_dtls_transport"
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c:type="GstWebRTCDTLSTransport"
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parent="Gst.Object"
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glib:type-name="GstWebRTCDTLSTransport"
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glib:get-type="gst_webrtc_dtls_transport_get_type"
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glib:type-struct="WebRTCDTLSTransportClass">
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<source-position filename="dtlstransport.h" line="59"/>
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<constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
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<source-position filename="dtlstransport.h" line="62"/>
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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"
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c:identifier="gst_webrtc_dtls_transport_set_transport">
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<source-position filename="dtlstransport.h" line="65"/>
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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"
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writable="1"
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construct-only="1"
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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"
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writable="1"
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construct-only="1"
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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"
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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"
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c:type="GstWebRTCDTLSTransportClass"
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glib:is-gtype-struct-for="WebRTCDTLSTransport">
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<source-position filename="dtlstransport.h" line="59"/>
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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"
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glib:type-name="GstWebRTCDTLSTransportState"
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glib:get-type="gst_webrtc_dtls_transport_state_get_type"
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c:type="GstWebRTCDTLSTransportState">
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<doc xml:space="preserve"
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filename="webrtc_fwd.h"
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line="57">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"
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value="0"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
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glib:nick="new">
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</member>
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<member name="closed"
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value="1"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
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glib:nick="closed">
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</member>
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<member name="failed"
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value="2"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
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glib:nick="failed">
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</member>
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<member name="connecting"
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value="3"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
</member>
|
|
<member name="connected"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
|
|
glib:nick="connected">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCDataChannelState"
|
|
version="1.16"
|
|
glib:type-name="GstWebRTCDataChannelState"
|
|
glib:get-type="gst_webrtc_data_channel_state_get_type"
|
|
c:type="GstWebRTCDataChannelState">
|
|
<doc xml:space="preserve"
|
|
filename="webrtc_fwd.h"
|
|
line="311">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
|
|
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
|
|
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
|
|
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
|
|
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW"
|
|
glib:nick="new">
|
|
</member>
|
|
<member name="connecting"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING"
|
|
glib:nick="connecting">
|
|
</member>
|
|
<member name="open"
|
|
value="2"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN"
|
|
glib:nick="open">
|
|
</member>
|
|
<member name="closing"
|
|
value="3"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING"
|
|
glib:nick="closing">
|
|
</member>
|
|
<member name="closed"
|
|
value="4"
|
|
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED"
|
|
glib:nick="closed">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCFECType"
|
|
version="1.14.1"
|
|
glib:type-name="GstWebRTCFECType"
|
|
glib:get-type="gst_webrtc_fec_type_get_type"
|
|
c:type="GstWebRTCFECType">
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve" filename="webrtc_fwd.h" line="262">none</doc>
|
|
</member>
|
|
<member name="ulp_red"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
|
|
glib:nick="ulp-red">
|
|
<doc xml:space="preserve"
|
|
filename="webrtc_fwd.h"
|
|
line="263">ulpfec + red</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEComponent"
|
|
glib:type-name="GstWebRTCICEComponent"
|
|
glib:get-type="gst_webrtc_ice_component_get_type"
|
|
c:type="GstWebRTCICEComponent">
|
|
<doc xml:space="preserve"
|
|
filename="webrtc_fwd.h"
|
|
line="165">GST_WEBRTC_ICE_COMPONENT_RTP,
|
|
GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
|
|
<member name="rtp"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
|
|
glib:nick="rtp">
|
|
</member>
|
|
<member name="rtcp"
|
|
value="1"
|
|
c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
|
|
glib:nick="rtcp">
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="WebRTCICEConnectionState"
|
|
glib:type-name="GstWebRTCICEConnectionState"
|
|
glib:get-type="gst_webrtc_ice_connection_state_get_type"
|
|
c:type="GstWebRTCICEConnectionState">
|
|
<doc xml:space="preserve"
|
|
filename="webrtc_fwd.h"
|
|
line="89">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
|
|
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink></doc>
|
|
<member name="new"
|
|
value="0"
|
|
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
|
|
glib:nick="new">
|
|
</member>
|
|
<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"
|
|
filename="webrtc_fwd.h"
|
|
line="74">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"
|
|
filename="webrtc_fwd.h"
|
|
line="154">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">
|
|
<source-position filename="icetransport.h" line="61"/>
|
|
<virtual-method name="gather_candidates">
|
|
<source-position filename="icetransport.h" line="58"/>
|
|
<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">
|
|
<source-position filename="icetransport.h" line="64"/>
|
|
<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">
|
|
<source-position filename="icetransport.h" line="67"/>
|
|
<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">
|
|
<source-position filename="icetransport.h" line="72"/>
|
|
<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">
|
|
<source-position filename="icetransport.h" line="70"/>
|
|
<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">
|
|
<source-position filename="icetransport.h" line="61"/>
|
|
<field name="parent_class">
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="gather_candidates">
|
|
<callback name="gather_candidates">
|
|
<source-position filename="icetransport.h" line="58"/>
|
|
<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"
|
|
filename="webrtc_fwd.h"
|
|
line="352">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"
|
|
filename="webrtc_fwd.h"
|
|
line="133">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"
|
|
filename="webrtc_fwd.h"
|
|
line="292">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">
|
|
<source-position filename="rtpreceiver.h" line="54"/>
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_receiver_new">
|
|
<source-position filename="rtpreceiver.h" line="57"/>
|
|
<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">
|
|
<source-position filename="rtpreceiver.h" line="62"/>
|
|
<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">
|
|
<source-position filename="rtpreceiver.h" line="59"/>
|
|
<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">
|
|
<source-position filename="rtpreceiver.h" line="54"/>
|
|
<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">
|
|
<source-position filename="rtpsender.h" line="56"/>
|
|
<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
|
|
<source-position filename="rtpsender.h" line="59"/>
|
|
<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">
|
|
<source-position filename="rtpsender.h" line="65"/>
|
|
<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">
|
|
<source-position filename="rtpsender.h" line="62"/>
|
|
<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">
|
|
<source-position filename="rtpsender.h" line="56"/>
|
|
<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">
|
|
<source-position filename="rtptransceiver.h" line="62"/>
|
|
<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">
|
|
<source-position filename="rtptransceiver.h" line="62"/>
|
|
<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"
|
|
filename="webrtc_fwd.h"
|
|
line="273">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"
|
|
filename="webrtc_fwd.h"
|
|
line="176">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">
|
|
<source-position filename="rtcsessiondescription.h" line="30"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.c"
|
|
line="41">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"
|
|
filename="rtcsessiondescription.c"
|
|
line="39">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"
|
|
filename="rtcsessiondescription.h"
|
|
line="36">See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink></doc>
|
|
<source-position filename="rtcsessiondescription.h" line="47"/>
|
|
<field name="type" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.h"
|
|
line="38">the #GstWebRTCSDPType of the description</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</field>
|
|
<field name="sdp" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.h"
|
|
line="39">the #GstSDPMessage of the description</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</field>
|
|
<constructor name="new"
|
|
c:identifier="gst_webrtc_session_description_new">
|
|
<source-position filename="rtcsessiondescription.h" line="50"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.c"
|
|
line="103">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"
|
|
filename="rtcsessiondescription.c"
|
|
line="100">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
<parameter name="sdp" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.c"
|
|
line="101">a #GstSDPMessage</doc>
|
|
<type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="copy" c:identifier="gst_webrtc_session_description_copy">
|
|
<source-position filename="rtcsessiondescription.h" line="52"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.c"
|
|
line="65">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"
|
|
filename="rtcsessiondescription.c"
|
|
line="63">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"
|
|
filename="rtcsessiondescription.c"
|
|
line="83">Free @desc and all associated resources</doc>
|
|
<source-position filename="rtcsessiondescription.h" line="54"/>
|
|
<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"
|
|
filename="rtcsessiondescription.c"
|
|
line="85">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"
|
|
filename="webrtc_fwd.h"
|
|
line="112">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"
|
|
filename="webrtc_fwd.h"
|
|
line="225">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">
|
|
<source-position filename="rtcsessiondescription.h" line="30"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="rtcsessiondescription.c"
|
|
line="41">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"
|
|
filename="rtcsessiondescription.c"
|
|
line="39">a #GstWebRTCSDPType</doc>
|
|
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|