diff options
Diffstat (limited to 'girs/GstWebRTC-1.0.gir')
| -rw-r--r-- | girs/GstWebRTC-1.0.gir | 152 |
1 files changed, 149 insertions, 3 deletions
diff --git a/girs/GstWebRTC-1.0.gir b/girs/GstWebRTC-1.0.gir index 951089f..fae95eb 100644 --- a/girs/GstWebRTC-1.0.gir +++ b/girs/GstWebRTC-1.0.gir @@ -15,6 +15,37 @@ and/or use gtk-doc annotations. --> shared-library="libgstwebrtc-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst"> + <enumeration name="WebRTCBundlePolicy" + 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> <enumeration name="WebRTCDTLSSetup" glib:type-name="GstWebRTCDTLSSetup" glib:get-type="gst_webrtc_dtls_setup_get_type" @@ -140,7 +171,7 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc> c:type="GstWebRTCDTLSTransportClass" glib:is-gtype-struct-for="WebRTCDTLSTransport"> <field name="parent_class"> - <type name="Gst.BinClass" c:type="GstBinClass"/> + <type name="Gst.ObjectClass" c:type="GstObjectClass"/> </field> <field name="_padding"> <array zero-terminated="0" c:type="gpointer" fixed-size="4"> @@ -183,6 +214,42 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc> glib:nick="connected"> </member> </enumeration> + <enumeration name="WebRTCDataChannelState" + 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 <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" glib:type-name="GstWebRTCFECType" glib:get-type="gst_webrtc_fec_type_get_type" @@ -445,7 +512,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc> c:type="GstWebRTCICETransportClass" glib:is-gtype-struct-for="WebRTCICETransport"> <field name="parent_class"> - <type name="Gst.BinClass" c:type="GstBinClass"/> + <type name="Gst.ObjectClass" c:type="GstObjectClass"/> </field> <field name="gather_candidates"> <callback name="gather_candidates"> @@ -465,6 +532,25 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc> </array> </field> </record> + <enumeration name="WebRTCICETransportPolicy" + glib:type-name="GstWebRTCICETransportPolicy" + glib:get-type="gst_webrtc_ice_transport_policy_get_type" + c:type="GstWebRTCICETransportPolicy"> + <doc xml:space="preserve">GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all +GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay +See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 +for more information.</doc> + <member name="all" + value="0" + c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL" + glib:nick="all"> + </member> + <member name="relay" + value="1" + c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY" + glib:nick="relay"> + </member> + </enumeration> <enumeration name="WebRTCPeerConnectionState" glib:type-name="GstWebRTCPeerConnectionState" glib:get-type="gst_webrtc_peer_connection_state_get_type" @@ -507,6 +593,36 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g glib:nick="closed"> </member> </enumeration> + <enumeration name="WebRTCPriorityType" + glib:type-name="GstWebRTCPriorityType" + glib:get-type="gst_webrtc_priority_type_get_type" + c:type="GstWebRTCPriorityType"> + <doc xml:space="preserve">GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low +GST_WEBRTC_PRIORITY_TYPE_LOW: low +GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium +GST_WEBRTC_PRIORITY_TYPE_HIGH: high +See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink></doc> + <member name="very_low" + value="1" + c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW" + glib:nick="very-low"> + </member> + <member name="low" + value="2" + c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW" + glib:nick="low"> + </member> + <member name="medium" + value="3" + c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM" + glib:nick="medium"> + </member> + <member name="high" + value="4" + c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH" + glib:nick="high"> + </member> + </enumeration> <class name="WebRTCRTPReceiver" c:symbol-prefix="webrtc_rtp_receiver" c:type="GstWebRTCRTPReceiver" @@ -749,6 +865,36 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g glib:nick="sendrecv"> </member> </enumeration> + <enumeration name="WebRTCSCTPTransportState" + glib:type-name="GstWebRTCSCTPTransportState" + glib:get-type="gst_webrtc_sctp_transport_state_get_type" + c:type="GstWebRTCSCTPTransportState"> + <doc xml:space="preserve">GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new +GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting +GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected +GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed +See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink></doc> + <member name="new" + value="0" + c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW" + glib:nick="new"> + </member> + <member name="connecting" + value="1" + c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING" + glib:nick="connecting"> + </member> + <member name="connected" + value="2" + c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED" + glib:nick="connected"> + </member> + <member name="closed" + value="3" + c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED" + glib:nick="closed"> + </member> + </enumeration> <enumeration name="WebRTCSDPType" glib:type-name="GstWebRTCSDPType" glib:get-type="gst_webrtc_sdp_type_get_type" @@ -819,7 +965,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.git <doc xml:space="preserve">a #GstWebRTCSDPType</doc> <type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/> </parameter> - <parameter name="sdp" transfer-ownership="none"> + <parameter name="sdp" transfer-ownership="full"> <doc xml:space="preserve">a #GstSDPMessage</doc> <type name="GstSdp.SDPMessage" c:type="GstSDPMessage*"/> </parameter> |