mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer-rs.git
synced 2024-11-18 23:51:05 +00:00
332 lines
9.1 KiB
Markdown
332 lines
9.1 KiB
Markdown
<!-- file * -->
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<!-- enum WebRTCBundlePolicy -->
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GST_WEBRTC_BUNDLE_POLICY_NONE: none
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GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
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GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
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GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
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See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24`section`-4.1.1
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for more information.
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Feature: `v1_16`
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<!-- enum WebRTCDTLSSetup -->
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<!-- enum WebRTCDTLSSetup::variant None -->
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none
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<!-- enum WebRTCDTLSSetup::variant Actpass -->
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actpass
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<!-- enum WebRTCDTLSSetup::variant Active -->
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sendonly
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<!-- enum WebRTCDTLSSetup::variant Passive -->
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recvonly
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<!-- struct WebRTCDTLSTransport -->
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- enum WebRTCDTLSTransportState -->
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<!-- enum WebRTCDTLSTransportState::variant New -->
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new
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<!-- enum WebRTCDTLSTransportState::variant Closed -->
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closed
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<!-- enum WebRTCDTLSTransportState::variant Failed -->
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failed
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<!-- enum WebRTCDTLSTransportState::variant Connecting -->
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connecting
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<!-- enum WebRTCDTLSTransportState::variant Connected -->
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connected
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<!-- struct WebRTCDataChannel -->
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This is an Abstract Base Class, you cannot instantiate it.
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Feature: `v1_18`
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- impl WebRTCDataChannel::fn close -->
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Close the `self`.
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Feature: `v1_18`
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<!-- impl WebRTCDataChannel::fn on_buffered_amount_low -->
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Signal that the data channel reached a low buffered amount. Should only be used by subclasses.
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Feature: `v1_18`
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<!-- impl WebRTCDataChannel::fn on_close -->
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Signal that the data channel was closed. Should only be used by subclasses.
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Feature: `v1_18`
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<!-- impl WebRTCDataChannel::fn on_error -->
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Signal that the data channel had an error. Should only be used by subclasses.
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Feature: `v1_18`
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## `error`
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a `glib::Error`
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<!-- impl WebRTCDataChannel::fn on_message_data -->
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Signal that the data channel received a data message. Should only be used by subclasses.
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Feature: `v1_18`
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## `data`
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a `glib::Bytes` or `None`
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<!-- impl WebRTCDataChannel::fn on_message_string -->
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Signal that the data channel received a string message. Should only be used by subclasses.
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Feature: `v1_18`
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## `str`
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a string or `None`
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<!-- impl WebRTCDataChannel::fn on_open -->
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Signal that the data channel was opened. Should only be used by subclasses.
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Feature: `v1_18`
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<!-- impl WebRTCDataChannel::fn send_data -->
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Send `data` as a data message over `self`.
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Feature: `v1_18`
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## `data`
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a `glib::Bytes` or `None`
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<!-- impl WebRTCDataChannel::fn send_string -->
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Send `str` as a string message over `self`.
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Feature: `v1_18`
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## `str`
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a string or `None`
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<!-- impl WebRTCDataChannel::fn connect_close -->
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Close the data channel
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<!-- impl WebRTCDataChannel::fn connect_on_error -->
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## `error`
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the `glib::Error` thrown
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<!-- impl WebRTCDataChannel::fn connect_on_message_data -->
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## `data`
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a `glib::Bytes` of the data received
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<!-- impl WebRTCDataChannel::fn connect_on_message_string -->
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## `data`
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the data received as a string
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<!-- impl WebRTCDataChannel::fn connect_send_data -->
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## `data`
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a `glib::Bytes` with the data
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<!-- impl WebRTCDataChannel::fn connect_send_string -->
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## `data`
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the data to send as a string
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<!-- enum WebRTCDataChannelState -->
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GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
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GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
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GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
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GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
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See <http://w3c.github.io/webrtc-pc/`dom`-rtcdatachannelstate>
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Feature: `v1_16`
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<!-- enum WebRTCFECType -->
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<!-- enum WebRTCFECType::variant None -->
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none
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<!-- enum WebRTCFECType::variant UlpRed -->
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ulpfec + red
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Feature: `v1_14_1`
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<!-- enum WebRTCICEComponent -->
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<!-- enum WebRTCICEComponent::variant Rtp -->
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RTP component
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<!-- enum WebRTCICEComponent::variant Rtcp -->
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RTCP component
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<!-- enum WebRTCICEConnectionState -->
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See <http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate>
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<!-- enum WebRTCICEConnectionState::variant New -->
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new
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<!-- enum WebRTCICEConnectionState::variant Checking -->
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checking
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<!-- enum WebRTCICEConnectionState::variant Connected -->
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connected
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<!-- enum WebRTCICEConnectionState::variant Completed -->
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completed
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<!-- enum WebRTCICEConnectionState::variant Failed -->
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failed
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<!-- enum WebRTCICEConnectionState::variant Disconnected -->
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disconnected
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<!-- enum WebRTCICEConnectionState::variant Closed -->
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closed
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<!-- enum WebRTCICEGatheringState -->
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See <http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate>
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<!-- enum WebRTCICEGatheringState::variant New -->
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new
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<!-- enum WebRTCICEGatheringState::variant Gathering -->
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gathering
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<!-- enum WebRTCICEGatheringState::variant Complete -->
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complete
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<!-- enum WebRTCICERole -->
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<!-- enum WebRTCICERole::variant Controlled -->
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controlled
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<!-- enum WebRTCICERole::variant Controlling -->
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controlling
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<!-- struct WebRTCICETransport -->
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This is an Abstract Base Class, you cannot instantiate it.
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- enum WebRTCICETransportPolicy -->
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GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
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GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
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See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24`section`-4.1.1
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for more information.
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Feature: `v1_16`
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<!-- enum WebRTCPeerConnectionState -->
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See <http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate>
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<!-- enum WebRTCPeerConnectionState::variant New -->
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new
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<!-- enum WebRTCPeerConnectionState::variant Connecting -->
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connecting
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<!-- enum WebRTCPeerConnectionState::variant Connected -->
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connected
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<!-- enum WebRTCPeerConnectionState::variant Disconnected -->
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disconnected
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<!-- enum WebRTCPeerConnectionState::variant Failed -->
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failed
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<!-- enum WebRTCPeerConnectionState::variant Closed -->
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closed
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<!-- enum WebRTCPriorityType -->
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GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
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GST_WEBRTC_PRIORITY_TYPE_LOW: low
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GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
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GST_WEBRTC_PRIORITY_TYPE_HIGH: high
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See <http://w3c.github.io/webrtc-pc/`dom`-rtcprioritytype>
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Feature: `v1_16`
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<!-- struct WebRTCRTPReceiver -->
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- struct WebRTCRTPSender -->
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- struct WebRTCRTPTransceiver -->
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This is an Abstract Base Class, you cannot instantiate it.
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# Implements
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[`trait@glib::object::ObjectExt`]
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<!-- impl WebRTCRTPTransceiver::fn get_property_direction -->
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Direction of the transceiver.
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Feature: `v1_18`
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<!-- impl WebRTCRTPTransceiver::fn set_property_direction -->
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Direction of the transceiver.
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Feature: `v1_18`
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<!-- enum WebRTCRTPTransceiverDirection -->
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<!-- enum WebRTCRTPTransceiverDirection::variant None -->
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none
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<!-- enum WebRTCRTPTransceiverDirection::variant Inactive -->
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inactive
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<!-- enum WebRTCRTPTransceiverDirection::variant Sendonly -->
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sendonly
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<!-- enum WebRTCRTPTransceiverDirection::variant Recvonly -->
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recvonly
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<!-- enum WebRTCRTPTransceiverDirection::variant Sendrecv -->
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sendrecv
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<!-- enum WebRTCSCTPTransportState -->
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GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
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GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
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See <http://w3c.github.io/webrtc-pc/`dom`-rtcsctptransportstate>
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Feature: `v1_16`
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<!-- enum WebRTCSDPType -->
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See <http://w3c.github.io/webrtc-pc/`rtcsdptype`>
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<!-- enum WebRTCSDPType::variant Offer -->
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offer
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<!-- enum WebRTCSDPType::variant Pranswer -->
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pranswer
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<!-- enum WebRTCSDPType::variant Answer -->
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answer
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<!-- enum WebRTCSDPType::variant Rollback -->
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rollback
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<!-- struct WebRTCSessionDescription -->
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See <https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class>
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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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See <http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate>
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<!-- enum WebRTCSignalingState::variant Stable -->
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stable
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<!-- enum WebRTCSignalingState::variant Closed -->
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closed
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<!-- enum WebRTCSignalingState::variant HaveLocalOffer -->
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have-local-offer
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<!-- enum WebRTCSignalingState::variant HaveRemoteOffer -->
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have-remote-offer
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<!-- enum WebRTCSignalingState::variant HaveLocalPranswer -->
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have-local-pranswer
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<!-- enum WebRTCSignalingState::variant HaveRemotePranswer -->
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have-remote-pranswer
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<!-- enum WebRTCStatsType -->
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<!-- enum WebRTCStatsType::variant Codec -->
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codec
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<!-- enum WebRTCStatsType::variant InboundRtp -->
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inbound-rtp
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<!-- enum WebRTCStatsType::variant OutboundRtp -->
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outbound-rtp
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<!-- enum WebRTCStatsType::variant RemoteInboundRtp -->
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remote-inbound-rtp
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<!-- enum WebRTCStatsType::variant RemoteOutboundRtp -->
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remote-outbound-rtp
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<!-- enum WebRTCStatsType::variant Csrc -->
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csrc
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<!-- enum WebRTCStatsType::variant PeerConnection -->
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peer-connectiion
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<!-- enum WebRTCStatsType::variant DataChannel -->
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data-channel
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<!-- enum WebRTCStatsType::variant Stream -->
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stream
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<!-- enum WebRTCStatsType::variant Transport -->
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transport
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<!-- enum WebRTCStatsType::variant CandidatePair -->
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candidate-pair
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<!-- enum WebRTCStatsType::variant LocalCandidate -->
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local-candidate
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<!-- enum WebRTCStatsType::variant RemoteCandidate -->
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remote-candidate
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<!-- enum WebRTCStatsType::variant Certificate -->
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certificate
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