mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
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120 lines
4.5 KiB
Markdown
120 lines
4.5 KiB
Markdown
<!-- file * -->
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<!-- enum WebRTCDTLSSetup -->
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GST_WEBRTC_DTLS_SETUP_NONE: none
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GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
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GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
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GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
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<!-- struct WebRTCDTLSTransport -->
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# Implements
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[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
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<!-- enum WebRTCDTLSTransportState -->
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GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
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GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected
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<!-- enum WebRTCICEComponent -->
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GST_WEBRTC_ICE_COMPONENT_RTP,
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GST_WEBRTC_ICE_COMPONENT_RTCP,
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<!-- enum WebRTCICEConnectionState -->
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GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
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GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
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GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
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GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
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GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
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GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
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GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate`</ulink>`
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<!-- enum WebRTCICEGatheringState -->
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GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
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GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
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GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
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See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate`</ulink>`
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<!-- enum WebRTCICERole -->
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GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
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GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
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<!-- struct WebRTCICETransport -->
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# Implements
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[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
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<!-- enum WebRTCPeerConnectionState -->
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GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
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GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
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GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
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GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
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GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
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See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate`</ulink>`
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<!-- struct WebRTCRTPReceiver -->
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# Implements
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[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
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<!-- struct WebRTCRTPSender -->
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# Implements
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[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
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<!-- struct WebRTCRTPTransceiver -->
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# Implements
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[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
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<!-- enum WebRTCRTPTransceiverDirection -->
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<!-- enum WebRTCSDPType -->
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GST_WEBRTC_SDP_TYPE_OFFER: offer
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GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
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GST_WEBRTC_SDP_TYPE_ANSWER: answer
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GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
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See <ulink url="http://w3c.github.io/webrtc-pc/`rtcsdptype`">http://w3c.github.io/webrtc-pc/`rtcsdptype``</ulink>`
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<!-- struct WebRTCSessionDescription -->
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See <ulink url="https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class">https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class`</ulink>`
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<!-- impl WebRTCSessionDescription::fn new -->
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## `type_`
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a `WebRTCSDPType`
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## `sdp`
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a `gst_sdp::SDPMessage`
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# Returns
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a new `WebRTCSessionDescription` from `type_`
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and `sdp`
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<!-- impl WebRTCSessionDescription::fn copy -->
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# Returns
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a new copy of `self`
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<!-- impl WebRTCSessionDescription::fn free -->
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Free `self` and all associated resources
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<!-- enum WebRTCSignalingState -->
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GST_WEBRTC_SIGNALING_STATE_STABLE: stable
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GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
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GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
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GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
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GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
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GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
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See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate">http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate`</ulink>`
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<!-- enum WebRTCStatsType -->
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GST_WEBRTC_STATS_CODEC: codec
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GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
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GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
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GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
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GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
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GST_WEBRTC_STATS_CSRC: csrc
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GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
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GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
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GST_WEBRTC_STATS_STREAM: stream
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GST_WEBRTC_STATS_TRANSPORT: transport
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GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
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GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
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GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
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GST_WEBRTC_STATS_CERTIFICATE: certificate
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