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Update gir-files to gstreamer 1.14.0 release

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This commit is contained in:
Sebastian Dröge 2018-03-20 10:32:49 +02:00
parent 217a8671a5
commit 7e39cbbfed
4 changed files with 189 additions and 79 deletions
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4
gir-files/Gst-1.0.gir
View file

@ -44717,11 +44717,11 @@ determine a order for the two provided values.</doc>
<doc xml:space="preserve">The major version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="VERSION_MICRO" value="91" c:type="GST_VERSION_MICRO">
<constant name="VERSION_MICRO" value="0" c:type="GST_VERSION_MICRO">
<doc xml:space="preserve">The micro version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="VERSION_MINOR" value="13" c:type="GST_VERSION_MINOR">
<constant name="VERSION_MINOR" value="14" c:type="GST_VERSION_MINOR">
<doc xml:space="preserve">The minor version of GStreamer at compile time:</doc>
<type name="gint" c:type="gint"/>
</constant>

1
gir-files/GstAudio-1.0.gir
View file

@ -8417,6 +8417,7 @@ functionality.</doc>
</record>
<record name="AudioStreamAlign"
c:type="GstAudioStreamAlign"
version="1.14"
glib:type-name="GstAudioStreamAlign"
glib:get-type="gst_audio_stream_align_get_type"
c:symbol-prefix="audio_stream_align">

4
gir-files/GstPbutils-1.0.gir
View file

@ -2744,13 +2744,13 @@ in debugging.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MICRO"
value="91"
value="0"
c:type="GST_PLUGINS_BASE_VERSION_MICRO">
<doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MINOR"
value="13"
value="14"
c:type="GST_PLUGINS_BASE_VERSION_MINOR">
<doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/>

259
gir-files/GstWebRTC-1.0.gir
View file

@ -15,24 +15,33 @@ and/or use gtk-doc annotations. -->
shared-library="libgstwebrtc-1.0.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
<enumeration name="WebRTCDTLSSetup" c:type="GstWebRTCDTLSSetup">
<enumeration name="WebRTCDTLSSetup"
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
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
<member name="none" value="0" c:identifier="GST_WEBRTC_DTLS_SETUP_NONE">
<member name="none"
value="0"
c:identifier="GST_WEBRTC_DTLS_SETUP_NONE"
glib:nick="none">
</member>
<member name="actpass"
value="1"
c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS">
c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS"
glib:nick="actpass">
</member>
<member name="active"
value="2"
c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE">
c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE"
glib:nick="active">
</member>
<member name="passive"
value="3"
c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE">
c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE"
glib:nick="passive">
</member>
</enumeration>
<class name="WebRTCDTLSTransport"
@ -90,8 +99,8 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="state" introspectable="0" transfer-ownership="none">
<type/>
<property name="state" transfer-ownership="none">
<type name="WebRTCDTLSTransportState"/>
</property>
<property name="transport" transfer-ownership="none">
<type name="WebRTCICETransport"/>
@ -140,6 +149,8 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
</field>
</record>
<enumeration name="WebRTCDTLSTransportState"
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
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
@ -148,36 +159,50 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
<member name="new"
value="0"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW">
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW"
glib:nick="new">
</member>
<member name="closed"
value="1"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED">
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED"
glib:nick="closed">
</member>
<member name="failed"
value="2"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED">
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED"
glib:nick="failed">
</member>
<member name="connecting"
value="3"
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING">
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">
c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED"
glib:nick="connected">
</member>
</enumeration>
<enumeration name="WebRTCICEComponent" c:type="GstWebRTCICEComponent">
<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,
GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP">
<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">
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">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
@ -189,34 +214,43 @@ 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"
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW">
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">
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">
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">
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">
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">
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">
c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED"
glib:nick="closed">
</member>
</enumeration>
<enumeration name="WebRTCICEGatheringState"
glib:type-name="GstWebRTCICEGatheringState"
glib:get-type="gst_webrtc_ice_gathering_state_get_type"
c:type="GstWebRTCICEGatheringState">
<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
@ -224,27 +258,35 @@ 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"
c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW">
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">
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">
c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE"
glib:nick="complete">
</member>
</enumeration>
<enumeration name="WebRTCICERole" c:type="GstWebRTCICERole">
<enumeration name="WebRTCICERole"
glib:type-name="GstWebRTCICERole"
glib:get-type="gst_webrtc_ice_role_get_type"
c:type="GstWebRTCICERole">
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
<member name="controlled"
value="0"
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED">
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED"
glib:nick="controlled">
</member>
<member name="controlling"
value="1"
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING">
c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING"
glib:nick="controlling">
</member>
</enumeration>
<class name="WebRTCICETransport"
@ -327,19 +369,16 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</parameters>
</method>
<property name="component"
introspectable="0"
writable="1"
construct-only="1"
transfer-ownership="none">
<type/>
<type name="WebRTCICEComponent"/>
</property>
<property name="gathering-state"
introspectable="0"
transfer-ownership="none">
<type/>
<property name="gathering-state" transfer-ownership="none">
<type name="WebRTCICEGatheringState"/>
</property>
<property name="state" introspectable="0" transfer-ownership="none">
<type/>
<property name="state" transfer-ownership="none">
<type name="WebRTCICEConnectionState"/>
</property>
<field name="parent">
<type name="Gst.Object" c:type="GstObject"/>
@ -410,6 +449,8 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</field>
</record>
<enumeration name="WebRTCPeerConnectionState"
glib:type-name="GstWebRTCPeerConnectionState"
glib:get-type="gst_webrtc_peer_connection_state_get_type"
c:type="GstWebRTCPeerConnectionState">
<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
@ -420,27 +461,33 @@ 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"
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW">
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">
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">
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">
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">
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">
c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED"
glib:nick="closed">
</member>
</enumeration>
<class name="WebRTCRTPReceiver"
@ -656,48 +703,77 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
</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">
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">
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">
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">
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">
c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV"
glib:nick="sendrecv">
</member>
</enumeration>
<enumeration name="WebRTCSDPType" c:type="GstWebRTCSDPType">
<enumeration name="WebRTCSDPType"
glib:type-name="GstWebRTCSDPType"
glib:get-type="gst_webrtc_sdp_type_get_type"
c:type="GstWebRTCSDPType">
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
GST_WEBRTC_SDP_TYPE_ANSWER: answer
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.github.io/webrtc-pc/#rtcsdptype&lt;/ulink&gt;</doc>
<member name="offer" value="1" c:identifier="GST_WEBRTC_SDP_TYPE_OFFER">
<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">
c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER"
glib:nick="pranswer">
</member>
<member name="answer"
value="3"
c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER">
c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER"
glib:nick="answer">
</member>
<member name="rollback"
value="4"
c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK">
c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK"
glib:nick="rollback">
</member>
<function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
<return-value transfer-ownership="none">
<doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
recognized.</doc>
<type name="utf8" c:type="const gchar*"/>
</return-value>
<parameters>
<parameter name="type" transfer-ownership="none">
<doc xml:space="preserve">a #GstWebRTCSDPType</doc>
<type name="WebRTCSDPType" c:type="GstWebRTCSDPType"/>
</parameter>
</parameters>
</function>
</enumeration>
<record name="WebRTCSessionDescription"
c:type="GstWebRTCSessionDescription"
@ -759,7 +835,10 @@ See &lt;ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class"&gt
</parameters>
</method>
</record>
<enumeration name="WebRTCSignalingState" c:type="GstWebRTCSignalingState">
<enumeration name="WebRTCSignalingState"
glib:type-name="GstWebRTCSignalingState"
glib:get-type="gst_webrtc_signaling_state_get_type"
c:type="GstWebRTCSignalingState">
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
@ -769,30 +848,39 @@ 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"
c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE">
c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE"
glib:nick="stable">
</member>
<member name="closed"
value="1"
c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED">
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">
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">
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">
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">
c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER"
glib:nick="have-remote-pranswer">
</member>
</enumeration>
<enumeration name="WebRTCStatsType" c:type="GstWebRTCStatsType">
<enumeration name="WebRTCStatsType"
glib:type-name="GstWebRTCStatsType"
glib:get-type="gst_webrtc_stats_type_get_type"
c:type="GstWebRTCStatsType">
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
@ -807,59 +895,80 @@ 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">
<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">
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">
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">
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">
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">
<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">
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">
c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL"
glib:nick="data-channel">
</member>
<member name="stream" value="9" c:identifier="GST_WEBRTC_STATS_STREAM">
<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">
c:identifier="GST_WEBRTC_STATS_TRANSPORT"
glib:nick="transport">
</member>
<member name="candidate_pair"
value="11"
c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR">
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">
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">
c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE"
glib:nick="remote-candidate">
</member>
<member name="certificate"
value="14"
c:identifier="GST_WEBRTC_STATS_CERTIFICATE">
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">
c:identifier="gst_webrtc_sdp_type_to_string"
moved-to="WebRTCSDPType.to_string">
<return-value transfer-ownership="none">
<doc xml:space="preserve">the string representation of @type or "unknown" when @type is not
recognized.</doc>