gstreamer-rs/docs/gstreamer-webrtc/docs.md
2020-06-19 13:09:39 +03:00

9.2 KiB

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-24section-4.1.1 for more information.

Feature: v1_16

none

actpass

sendonly

recvonly

Implements

glib::object::ObjectExt

new

closed

failed

connecting

connected

Feature: v1_18

Implements

glib::object::ObjectExt

Close the self.

Feature: v1_18

Signal that the data channel reached a low buffered amount. Should only be used by subclasses.

Feature: v1_18

Signal that the data channel was closed. Should only be used by subclasses.

Feature: v1_18

Signal that the data channel had an error. Should only be used by subclasses.

Feature: v1_18

error

a glib::Error

Signal that the data channel received a data message. Should only be used by subclasses.

Feature: v1_18

data

a glib::Bytes or None

Signal that the data channel received a string message. Should only be used by subclasses.

Feature: v1_18

str

a string or None

Signal that the data channel was opened. Should only be used by subclasses.

Feature: v1_18

Send data as a data message over self.

Feature: v1_18

data

a glib::Bytes or None

Send str as a string message over self.

Feature: v1_18

str

a string or None

Close the data channel

error

the glib::Error thrown

data

a glib::Bytes of the data received

data

the data received as a string

data

a glib::Bytes with the data

data

the data to send as a string

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 http://w3c.github.io/webrtc-pc/`dom`-rtcdatachannelstate

Feature: v1_16

none

ulpfec + red

Feature: v1_14_1

RTP component

RTCP component

See http://w3c.github.io/webrtc-pc/`dom`-rtciceconnectionstate

new

checking

connected

completed

failed

disconnected

closed

See http://w3c.github.io/webrtc-pc/`dom`-rtcicegatheringstate

new

gathering

complete

controlled

controlling

Implements

glib::object::ObjectExt

GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24section-4.1.1 for more information.

Feature: v1_16

See http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate

new

connecting

connected

disconnected

failed

closed

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 http://w3c.github.io/webrtc-pc/`dom`-rtcprioritytype

Feature: v1_16

Implements

glib::object::ObjectExt

Implements

glib::object::ObjectExt

Implements

glib::object::ObjectExt

Direction of the transceiver.

Feature: v1_18

Direction of the transceiver.

Feature: v1_18

none

inactive

sendonly

recvonly

sendrecv

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 http://w3c.github.io/webrtc-pc/`dom`-rtcsctptransportstate

Feature: v1_16

See http://w3c.github.io/webrtc-pc/`rtcsdptype`

offer

pranswer

answer

rollback

See https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class

type_

a WebRTCSDPType

sdp

a gst_sdp::SDPMessage

Returns

a new WebRTCSessionDescription from type_ and sdp

Returns

a new copy of self

Free self and all associated resources

See http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate

stable

closed

have-local-offer

have-remote-offer

have-local-pranswer

have-remote-pranswer

codec

inbound-rtp

outbound-rtp

remote-inbound-rtp

remote-outbound-rtp

csrc

peer-connectiion

data-channel

stream

transport

candidate-pair

local-candidate

remote-candidate

certificate