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Update gir-files to 1.16.2

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Sebastian Dröge 2019-12-15 12:18:09 +02:00
parent 78525333cc
commit 653b7f1da4
18 changed files with 85403 additions and 22450 deletions
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@ -15,12 +15,239 @@ and/or use gtk-doc annotations. -->
shared-library="libgstwebrtc-1.0.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
<function-macro name="IS_WEBRTC_DTLS_TRANSPORT"
c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT"
introspectable="0">
<source-position filename="dtlstransport.h" line="33"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_DTLS_TRANSPORT_CLASS"
c:identifier="GST_IS_WEBRTC_DTLS_TRANSPORT_CLASS"
introspectable="0">
<source-position filename="dtlstransport.h" line="35"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_ICE_TRANSPORT"
c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT"
introspectable="0">
<source-position filename="icetransport.h" line="32"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_ICE_TRANSPORT_CLASS"
c:identifier="GST_IS_WEBRTC_ICE_TRANSPORT_CLASS"
introspectable="0">
<source-position filename="icetransport.h" line="34"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_RECEIVER"
c:identifier="GST_IS_WEBRTC_RTP_RECEIVER"
introspectable="0">
<source-position filename="rtpreceiver.h" line="33"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_RECEIVER_CLASS"
c:identifier="GST_IS_WEBRTC_RTP_RECEIVER_CLASS"
introspectable="0">
<source-position filename="rtpreceiver.h" line="35"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_SENDER"
c:identifier="GST_IS_WEBRTC_RTP_SENDER"
introspectable="0">
<source-position filename="rtpsender.h" line="33"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_SENDER_CLASS"
c:identifier="GST_IS_WEBRTC_RTP_SENDER_CLASS"
introspectable="0">
<source-position filename="rtpsender.h" line="35"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER"
c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER"
introspectable="0">
<source-position filename="rtptransceiver.h" line="34"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
c:identifier="GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS"
introspectable="0">
<source-position filename="rtptransceiver.h" line="36"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DTLS_TRANSPORT"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT"
introspectable="0">
<source-position filename="dtlstransport.h" line="32"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DTLS_TRANSPORT_CLASS"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_CLASS"
introspectable="0">
<source-position filename="dtlstransport.h" line="34"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_DTLS_TRANSPORT_GET_CLASS"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_GET_CLASS"
introspectable="0">
<source-position filename="dtlstransport.h" line="36"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_ICE_TRANSPORT"
c:identifier="GST_WEBRTC_ICE_TRANSPORT"
introspectable="0">
<source-position filename="icetransport.h" line="31"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_ICE_TRANSPORT_CLASS"
c:identifier="GST_WEBRTC_ICE_TRANSPORT_CLASS"
introspectable="0">
<source-position filename="icetransport.h" line="33"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_ICE_TRANSPORT_GET_CLASS"
c:identifier="GST_WEBRTC_ICE_TRANSPORT_GET_CLASS"
introspectable="0">
<source-position filename="icetransport.h" line="35"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_RECEIVER"
c:identifier="GST_WEBRTC_RTP_RECEIVER"
introspectable="0">
<source-position filename="rtpreceiver.h" line="32"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_RECEIVER_CLASS"
c:identifier="GST_WEBRTC_RTP_RECEIVER_CLASS"
introspectable="0">
<source-position filename="rtpreceiver.h" line="34"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_RECEIVER_GET_CLASS"
c:identifier="GST_WEBRTC_RTP_RECEIVER_GET_CLASS"
introspectable="0">
<source-position filename="rtpreceiver.h" line="36"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_SENDER"
c:identifier="GST_WEBRTC_RTP_SENDER"
introspectable="0">
<source-position filename="rtpsender.h" line="32"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_SENDER_CLASS"
c:identifier="GST_WEBRTC_RTP_SENDER_CLASS"
introspectable="0">
<source-position filename="rtpsender.h" line="34"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_SENDER_GET_CLASS"
c:identifier="GST_WEBRTC_RTP_SENDER_GET_CLASS"
introspectable="0">
<source-position filename="rtpsender.h" line="36"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_TRANSCEIVER"
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER"
introspectable="0">
<source-position filename="rtptransceiver.h" line="33"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_TRANSCEIVER_CLASS"
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_CLASS"
introspectable="0">
<source-position filename="rtptransceiver.h" line="35"/>
<parameters>
<parameter name="klass">
</parameter>
</parameters>
</function-macro>
<function-macro name="WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS"
introspectable="0">
<source-position filename="rtptransceiver.h" line="37"/>
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<enumeration name="WebRTCBundlePolicy"
version="1.16"
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
<doc xml:space="preserve"
filename="webrtc_fwd.h"
line="332">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
@ -51,7 +278,9 @@ for more information.</doc>
glib:type-name="GstWebRTCDTLSSetup"
glib:get-type="gst_webrtc_dtls_setup_get_type"
c:type="GstWebRTCDTLSSetup">
<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
<doc xml:space="preserve"
filename="webrtc_fwd.h"
line="210">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>
@ -83,7 +312,9 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
glib:type-name="GstWebRTCDTLSTransport"
glib:get-type="gst_webrtc_dtls_transport_get_type"
glib:type-struct="WebRTCDTLSTransportClass">
<source-position filename="dtlstransport.h" line="59"/>
<constructor name="new" c:identifier="gst_webrtc_dtls_transport_new">
<source-position filename="dtlstransport.h" line="62"/>
<return-value transfer-ownership="none">
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
</return-value>
@ -98,6 +329,7 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
</constructor>
<method name="set_transport"
c:identifier="gst_webrtc_dtls_transport_set_transport">
<source-position filename="dtlstransport.h" line="65"/>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -163,7 +395,7 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
<type name="Gst.Element" c:type="GstElement*"/>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -171,11 +403,12 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
<record name="WebRTCDTLSTransportClass"
c:type="GstWebRTCDTLSTransportClass"
glib:is-gtype-struct-for="WebRTCDTLSTransport">
<source-position filename="dtlstransport.h" line="59"/>
<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">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -184,7 +417,9 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
glib:type-name="GstWebRTCDTLSTransportState"
glib:get-type="gst_webrtc_dtls_transport_state_get_type"
c:type="GstWebRTCDTLSTransportState">
<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
<doc xml:space="preserve"
filename="webrtc_fwd.h"
line="57">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
@ -220,7 +455,9 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
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
<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
@ -261,20 +498,24 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate"&gt;h
value="0"
c:identifier="GST_WEBRTC_FEC_TYPE_NONE"
glib:nick="none">
<doc xml:space="preserve">none</doc>
<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">ulpfec + red</doc>
<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">GST_WEBRTC_ICE_COMPONENT_RTP,
<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"
@ -291,7 +532,9 @@ GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
glib:type-name="GstWebRTCICEConnectionState"
glib:get-type="gst_webrtc_ice_connection_state_get_type"
c:type="GstWebRTCICEConnectionState">
<doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
<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
@ -339,7 +582,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate"&gt
glib:type-name="GstWebRTCICEGatheringState"
glib:get-type="gst_webrtc_ice_gathering_state_get_type"
c:type="GstWebRTCICEGatheringState">
<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
<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 &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&lt;/ulink&gt;</doc>
@ -363,7 +608,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;
glib:type-name="GstWebRTCICERole"
glib:get-type="gst_webrtc_ice_role_get_type"
c:type="GstWebRTCICERole">
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
<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"
@ -384,7 +631,9 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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>
@ -396,6 +645,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</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>
@ -411,6 +661,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</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>
@ -426,6 +677,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</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>
@ -446,6 +698,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</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>
@ -491,7 +744,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
<type name="Gst.Element" c:type="GstElement*"/>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -514,11 +767,13 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
<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>
@ -530,7 +785,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</callback>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -540,7 +795,9 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
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
<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>
@ -559,7 +816,9 @@ for more information.</doc>
glib:type-name="GstWebRTCPeerConnectionState"
glib:get-type="gst_webrtc_peer_connection_state_get_type"
c:type="GstWebRTCPeerConnectionState">
<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
<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
@ -602,7 +861,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
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
<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
@ -635,13 +896,16 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
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>
@ -656,6 +920,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
</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>
@ -678,7 +943,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
<type name="WebRTCDTLSTransport" c:type="GstWebRTCDTLSTransport*"/>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -686,11 +951,12 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
<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" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -702,13 +968,16 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
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>
@ -723,6 +992,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
</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>
@ -750,7 +1020,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
</array>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -758,11 +1028,12 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
<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" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -775,6 +1046,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
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"
@ -823,7 +1095,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
<type name="Gst.Caps" c:type="GstCaps*"/>
</field>
<field name="_padding">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -831,11 +1103,12 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
<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" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -875,7 +1148,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
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
<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
@ -905,7 +1180,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate"&gt
glib:type-name="GstWebRTCSDPType"
glib:get-type="gst_webrtc_sdp_type_get_type"
c:type="GstWebRTCSDPType">
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
<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
@ -931,14 +1208,19 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.git
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">the string representation of @type or "unknown" when @type is not
<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">a #GstWebRTCSDPType</doc>
<doc xml:space="preserve"
filename="rtcsessiondescription.c"
line="39">a #GstWebRTCSDPType</doc>
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
</parameter>
</parameters>
@ -949,56 +1231,80 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.git
glib:type-name="GstWebRTCSessionDescription"
glib:get-type="gst_webrtc_session_description_get_type"
c:symbol-prefix="webrtc_session_description">
<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>
<doc xml:space="preserve"
filename="rtcsessiondescription.h"
line="36">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>
<source-position filename="rtcsessiondescription.h" line="47"/>
<field name="type" writable="1">
<doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
<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">the #GstSDPMessage of the description</doc>
<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">a new #GstWebRTCSessionDescription from @type
<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">a #GstWebRTCSDPType</doc>
<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">a #GstSDPMessage</doc>
<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">a new copy of @src</doc>
<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">a #GstWebRTCSessionDescription</doc>
<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">Free @desc and all associated resources</doc>
<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">a #GstWebRTCSessionDescription</doc>
<doc xml:space="preserve"
filename="rtcsessiondescription.c"
line="85">a #GstWebRTCSessionDescription</doc>
<type name="WebRTCSessionDescription"
c:type="GstWebRTCSessionDescription*"/>
</instance-parameter>
@ -1009,7 +1315,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.git
glib:type-name="GstWebRTCSignalingState"
glib:get-type="gst_webrtc_signaling_state_get_type"
c:type="GstWebRTCSignalingState">
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
<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
@ -1051,7 +1359,9 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate"&gt;htt
glib:type-name="GstWebRTCStatsType"
glib:get-type="gst_webrtc_stats_type_get_type"
c:type="GstWebRTCStatsType">
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
<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
@ -1139,14 +1449,19 @@ GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
<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">the string representation of @type or "unknown" when @type is not
<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">a #GstWebRTCSDPType</doc>
<doc xml:space="preserve"
filename="rtcsessiondescription.c"
line="39">a #GstWebRTCSDPType</doc>
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
</parameter>
</parameters>