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gstreamer-rs/gir-files/GstWebRTC-1.0.gir

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Add gstreamer-webrtc-sys
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<?xml version="1.0"?>
<!-- This file was automatically generated from C sources - DO NOT EDIT!
To affect the contents of this file, edit the original C definitions,
and/or use gtk-doc annotations. -->
<repository version="1.2"
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">
<include name="Gst" version="1.0"/>
<include name="GstSdp" version="1.0"/>
<package name="gstreamer-webrtc-1.0"/>
<c:include name="gst/webrtc/webrtc.h"/>
<namespace name="GstWebRTC"
version="1.0"
shared-library="libgstwebrtc-1.0.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
Update gir-files to 1.16.0
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<enumeration name="WebRTCBundlePolicy"
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version="1.16"
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glib:type-name="GstWebRTCBundlePolicy"
glib:get-type="gst_webrtc_bundle_policy_get_type"
c:type="GstWebRTCBundlePolicy">
<doc xml:space="preserve">GST_WEBRTC_BUNDLE_POLICY_NONE: none
GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.</doc>
<member name="none"
value="0"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE"
glib:nick="none">
</member>
<member name="balanced"
value="1"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
glib:nick="balanced">
</member>
<member name="max_compat"
value="2"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
glib:nick="max-compat">
</member>
<member name="max_bundle"
value="3"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
glib:nick="max-bundle">
</member>
</enumeration>
Update gir-files to gstreamer 1.14.0 release
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<enumeration name="WebRTCDTLSSetup"
glib:type-name="GstWebRTCDTLSSetup"
glib:get-type="gst_webrtc_dtls_setup_get_type"
c:type="GstWebRTCDTLSSetup">
Add gstreamer-webrtc-sys
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<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
Update gir-files to gstreamer 1.14.0 release
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<member name="none"
value="0"
c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
glib:nick="none">
Add gstreamer-webrtc-sys
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</member>
<member name="actpass"
value="1"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
glib:nick="actpass">
Add gstreamer-webrtc-sys
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</member>
<member name="active"
value="2"
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c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
glib:nick="active">
Add gstreamer-webrtc-sys
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</member>
<member name="passive"
value="3"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
glib:nick="passive">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
<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">
<constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
<return-value transfer-ownership="none">
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
</return-value>
<parameters>
<parameter name="session_id" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="rtcp" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</constructor>
<method name="set_transport"
c:identifier="gst_webrtc_dtls_transport_set_transport">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="transport" transfer-ownership="none">
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
</instance-parameter>
<parameter name="ice" transfer-ownership="none">
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
</parameter>
</parameters>
</method>
<property name="certificate" writable="1" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</property>
<property name="client" writable="1" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
<property name="remote-certificate" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</property>
<property name="rtcp"
writable="1"
construct-only="1"
transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
<property name="session-id"
writable="1"
construct-only="1"
transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
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<property name="state" transfer-ownership="none">
<type name="WebRTCDTLSTransportState"/>
Add gstreamer-webrtc-sys
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</property>
<property name="transport" transfer-ownership="none">
<type name="WebRTCICETransport"/>
</property>
<field name="parent">
<type name="Gst.Object" c:type="GstObject"/>
</field>
<field name="transport">
<type name="WebRTCICETransport" c:type="GstWebRTCICETransport*"/>
</field>
<field name="state">
<type name="WebRTCDTLSTransportState"
c:type="GstWebRTCDTLSTransportState"/>
</field>
<field name="is_rtcp">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="client">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="session_id">
<type name="guint" c:type="guint"/>
</field>
<field name="dtlssrtpenc">
<type name="Gst.Element" c:type="GstElement*"/>
</field>
<field name="dtlssrtpdec">
<type name="Gst.Element" c:type="GstElement*"/>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</class>
<record name="WebRTCDTLSTransportClass"
c:type="GstWebRTCDTLSTransportClass"
glib:is-gtype-struct-for="WebRTCDTLSTransport">
<field name="parent_class">
Fix WebRTCICETransport and WebRTCDTLSTransport parent classes
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<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
Add gstreamer-webrtc-sys
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</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<enumeration name="WebRTCDTLSTransportState"
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glib:type-name="GstWebRTCDTLSTransportState"
glib:get-type="gst_webrtc_dtls_transport_state_get_type"
Add gstreamer-webrtc-sys
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c:type="GstWebRTCDTLSTransportState">
<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
<member name="new"
value="0"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
glib:nick="new">
Add gstreamer-webrtc-sys
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</member>
<member name="closed"
value="1"
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
glib:nick="closed">
Add gstreamer-webrtc-sys
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</member>
<member name="failed"
value="2"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
glib:nick="failed">
Add gstreamer-webrtc-sys
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</member>
<member name="connecting"
value="3"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING"
glib:nick="connecting">
Add gstreamer-webrtc-sys
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</member>
<member name="connected"
value="4"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
glib:nick="connected">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
<enumeration name="WebRTCDataChannelState"
Update gir-files from gstreamer-rs
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version="1.16"
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glib:type-name="GstWebRTCDataChannelState"
glib:get-type="gst_webrtc_data_channel_state_get_type"
c:type="GstWebRTCDataChannelState">
<doc xml:space="preserve">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate&lt;/ulink&gt;</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>
Update gir-files to 1.14.2 Using Debian gir files
2018-09-06 15:17:57 +00:00
<enumeration name="WebRTCFECType"
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version="1.14.1"
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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">
Update gir-files from GStreamer 1.16.1 release
2019-10-04 16:17:21 +00:00
<doc xml:space="preserve">none</doc>
Update gir-files to 1.14.2 Using Debian gir files
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</member>
<member name="ulp_red"
value="1"
c:identifier="GST_WEBRTC_FEC_TYPE_ULP_RED"
glib:nick="ulp-red">
Update gir-files from GStreamer 1.16.1 release
2019-10-04 16:17:21 +00:00
<doc xml:space="preserve">ulpfec + red</doc>
Update gir-files to 1.14.2 Using Debian gir files
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</member>
</enumeration>
Update gir-files to gstreamer 1.14.0 release
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<enumeration name="WebRTCICEComponent"
glib:type-name="GstWebRTCICEComponent"
glib:get-type="gst_webrtc_ice_component_get_type"
c:type="GstWebRTCICEComponent">
Update for GstWebRTCICERole enum rename
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<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
Update gir-files to gstreamer 1.14.0 release
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<member name="rtp"
value="0"
c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP"
glib:nick="rtp">
Add gstreamer-webrtc-sys
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</member>
<member name="rtcp"
value="1"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP"
glib:nick="rtcp">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
<enumeration name="WebRTCICEConnectionState"
Update gir-files to gstreamer 1.14.0 release
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glib:type-name="GstWebRTCICEConnectionState"
glib:get-type="gst_webrtc_ice_connection_state_get_type"
Add gstreamer-webrtc-sys
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c:type="GstWebRTCICEConnectionState">
<doc xml:space="preserve">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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&lt;/ulink&gt;</doc>
<member name="new"
value="0"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW"
glib:nick="new">
Add gstreamer-webrtc-sys
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</member>
<member name="checking"
value="1"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING"
glib:nick="checking">
Add gstreamer-webrtc-sys
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</member>
<member name="connected"
value="2"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED"
glib:nick="connected">
Add gstreamer-webrtc-sys
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</member>
<member name="completed"
value="3"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED"
glib:nick="completed">
Add gstreamer-webrtc-sys
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</member>
<member name="failed"
value="4"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED"
glib:nick="failed">
Add gstreamer-webrtc-sys
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</member>
<member name="disconnected"
value="5"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED"
glib:nick="disconnected">
Add gstreamer-webrtc-sys
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</member>
<member name="closed"
value="6"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
glib:nick="closed">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
<enumeration name="WebRTCICEGatheringState"
Update gir-files to gstreamer 1.14.0 release
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glib:type-name="GstWebRTCICEGatheringState"
glib:get-type="gst_webrtc_ice_gathering_state_get_type"
Add gstreamer-webrtc-sys
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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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&lt;/ulink&gt;</doc>
<member name="new"
value="0"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW"
glib:nick="new">
Add gstreamer-webrtc-sys
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</member>
<member name="gathering"
value="1"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING"
glib:nick="gathering">
Add gstreamer-webrtc-sys
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</member>
<member name="complete"
value="2"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
glib:nick="complete">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
Update gir-files to gstreamer 1.14.0 release
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<enumeration name="WebRTCICERole"
glib:type-name="GstWebRTCICERole"
glib:get-type="gst_webrtc_ice_role_get_type"
c:type="GstWebRTCICERole">
Update for GstWebRTCICERole enum rename
2018-03-15 15:00:38 +00:00
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
<member name="controlled"
value="0"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
glib:nick="controlled">
Update for GstWebRTCICERole enum rename
2018-03-15 15:00:38 +00:00
</member>
<member name="controlling"
value="1"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
glib:nick="controlling">
Update for GstWebRTCICERole enum rename
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</member>
</enumeration>
Add gstreamer-webrtc-sys
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<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">
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
<type name="WebRTCICEComponent"/>
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</property>
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
<property name="gathering-state" transfer-ownership="none">
<type name="WebRTCICEGatheringState"/>
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</property>
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
<property name="state" transfer-ownership="none">
<type name="WebRTCICEConnectionState"/>
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</property>
<field name="parent">
<type name="Gst.Object" c:type="GstObject"/>
</field>
<field name="role">
Update for GstWebRTCICERole enum rename
2018-03-15 15:00:38 +00:00
<type name="WebRTCICERole" c:type="GstWebRTCICERole"/>
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</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" c:type="gpointer" 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">
Fix WebRTCICETransport and WebRTCDTLSTransport parent classes
2018-06-05 21:25:51 +00:00
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
Add gstreamer-webrtc-sys
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</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" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
<enumeration name="WebRTCICETransportPolicy"
Update gir-files from gstreamer-rs
2019-10-04 16:08:29 +00:00
version="1.16"
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
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>
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
<enumeration name="WebRTCPeerConnectionState"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
glib:type-name="GstWebRTCPeerConnectionState"
glib:get-type="gst_webrtc_peer_connection_state_get_type"
Add gstreamer-webrtc-sys
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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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&lt;/ulink&gt;</doc>
<member name="new"
value="0"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW"
glib:nick="new">
Add gstreamer-webrtc-sys
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</member>
<member name="connecting"
value="1"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING"
glib:nick="connecting">
Add gstreamer-webrtc-sys
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</member>
<member name="connected"
value="2"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED"
glib:nick="connected">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="disconnected"
value="3"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED"
glib:nick="disconnected">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="failed"
value="4"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED"
glib:nick="failed">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="closed"
value="5"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
glib:nick="closed">
Add gstreamer-webrtc-sys
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</member>
</enumeration>
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
<enumeration name="WebRTCPriorityType"
Update gir-files from gstreamer-rs
2019-10-04 16:08:29 +00:00
version="1.16"
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype&lt;/ulink&gt;</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>
Add gstreamer-webrtc-sys
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<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" c:type="gpointer" 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" c:type="gpointer" 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">
More webrtc API fixup
2018-03-15 15:34:40 +00:00
<constructor name="new" c:identifier="gst_webrtc_rtp_sender_new">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
<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" c:type="gpointer" 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" c:type="gpointer" 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" c:type="gpointer" 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" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<enumeration name="WebRTCRTPTransceiverDirection"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
glib:type-name="GstWebRTCRTPTransceiverDirection"
glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type"
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
c:type="GstWebRTCRTPTransceiverDirection">
<member name="none"
value="0"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE"
glib:nick="none">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="inactive"
value="1"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE"
glib:nick="inactive">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="sendonly"
value="2"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY"
glib:nick="sendonly">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="recvonly"
value="3"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY"
glib:nick="recvonly">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="sendrecv"
value="4"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
glib:nick="sendrecv">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
</enumeration>
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
<enumeration name="WebRTCSCTPTransportState"
Update gir-files from gstreamer-rs
2019-10-04 16:08:29 +00:00
version="1.16"
Update gir-files to 1.16.0
2019-04-23 09:05:44 +00:00
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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate&lt;/ulink&gt;</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>
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
<enumeration name="WebRTCSDPType"
glib:type-name="GstWebRTCSDPType"
glib:get-type="gst_webrtc_sdp_type_get_type"
c:type="GstWebRTCSDPType">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
<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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.github.io/webrtc-pc/#rtcsdptype&lt;/ulink&gt;</doc>
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
<member name="offer"
value="1"
c:identifier="GST_WEBRTC_SDP_TYPE_OFFER"
glib:nick="offer">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="pranswer"
value="2"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
glib:nick="pranswer">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="answer"
value="3"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
glib:nick="answer">
Add gstreamer-webrtc-sys
2018-03-15 10:18:06 +00:00
</member>
<member name="rollback"
value="4"
Update gir-files to gstreamer 1.14.0 release
2018-03-20 08:32:49 +00:00
c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
glib:nick="rollback">
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</member>
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<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>
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</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">
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<doc xml:space="preserve">See &lt;ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class"&gt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&lt;/ulink&gt;</doc>
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<field name="type" writable="1">
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<doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
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<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
</field>
<field name="sdp" writable="1">
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<doc xml:space="preserve">the #GstSDPMessage of the description</doc>
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<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>
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<parameter name="sdp" transfer-ownership="full">
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<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>
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<enumeration name="WebRTCSignalingState"
glib:type-name="GstWebRTCSignalingState"
glib:get-type="gst_webrtc_signaling_state_get_type"
c:type="GstWebRTCSignalingState">
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<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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&lt;/ulink&gt;</doc>
<member name="stable"
value="0"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
glib:nick="stable">
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</member>
<member name="closed"
value="1"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED"
glib:nick="closed">
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</member>
<member name="have_local_offer"
value="2"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER"
glib:nick="have-local-offer">
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</member>
<member name="have_remote_offer"
value="3"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER"
glib:nick="have-remote-offer">
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</member>
<member name="have_local_pranswer"
value="4"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER"
glib:nick="have-local-pranswer">
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</member>
<member name="have_remote_pranswer"
value="5"
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c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
glib:nick="have-remote-pranswer">
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</member>
</enumeration>
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<enumeration name="WebRTCStatsType"
glib:type-name="GstWebRTCStatsType"
glib:get-type="gst_webrtc_stats_type_get_type"
c:type="GstWebRTCStatsType">
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<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>
Update gir-files to gstreamer 1.14.0 release
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<member name="codec"
value="1"
c:identifier="GST_WEBRTC_STATS_CODEC"
glib:nick="codec">
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</member>
<member name="inbound_rtp"
value="2"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_INBOUND_RTP"
glib:nick="inbound-rtp">
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</member>
<member name="outbound_rtp"
value="3"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP"
glib:nick="outbound-rtp">
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</member>
<member name="remote_inbound_rtp"
value="4"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
glib:nick="remote-inbound-rtp">
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</member>
<member name="remote_outbound_rtp"
value="5"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
glib:nick="remote-outbound-rtp">
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</member>
Update gir-files to gstreamer 1.14.0 release
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<member name="csrc"
value="6"
c:identifier="GST_WEBRTC_STATS_CSRC"
glib:nick="csrc">
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</member>
<member name="peer_connection"
value="7"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION"
glib:nick="peer-connection">
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</member>
<member name="data_channel"
value="8"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
glib:nick="data-channel">
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</member>
Update gir-files to gstreamer 1.14.0 release
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<member name="stream"
value="9"
c:identifier="GST_WEBRTC_STATS_STREAM"
glib:nick="stream">
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</member>
<member name="transport"
value="10"
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c:identifier="GST_WEBRTC_STATS_TRANSPORT"
glib:nick="transport">
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</member>
<member name="candidate_pair"
value="11"
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c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR"
glib:nick="candidate-pair">
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</member>
<member name="local_candidate"
value="12"
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c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE"
glib:nick="local-candidate">
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</member>
<member name="remote_candidate"
value="13"
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c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
glib:nick="remote-candidate">
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</member>
<member name="certificate"
value="14"
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c:identifier="GST_WEBRTC_STATS_CERTIFICATE"
glib:nick="certificate">
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</member>
</enumeration>
<function name="webrtc_sdp_type_to_string"
Update gir-files to gstreamer 1.14.0 release
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c:identifier="gst_webrtc_sdp_type_to_string"
moved-to="WebRTCSDPType.to_string">
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<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>
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