GstWebRTC Enumerations
GstWebRTCDTLSTransport
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCDTLSTransport
Class structure
GstWebRTCDTLSTransportClass
GstWebRTC.WebRTCDTLSTransportClass
GstWebRTC.WebRTCDTLSTransportClass
GstWebRTC.WebRTCDTLSTransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCDTLSTransport
GstWebRTC.WebRTCDTLSTransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCDTLSTransport
Properties
GstWebRTCDataChannel
GObject ╰──GstWebRTCDataChannel
Class structure
GstWebRTCDataChannelClass
GstWebRTC.WebRTCDataChannelClass
GstWebRTC.WebRTCDataChannelClass
Methods
gst_webrtc_data_channel_close
gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel)
Close the channel.
Parameters:
channel
–
GstWebRTC.WebRTCDataChannel.prototype.close
function GstWebRTC.WebRTCDataChannel.prototype.close(): {
// javascript wrapper for 'gst_webrtc_data_channel_close'
}
Close the channel.
Parameters:
GstWebRTC.WebRTCDataChannel.close
def GstWebRTC.WebRTCDataChannel.close (self):
#python wrapper for 'gst_webrtc_data_channel_close'
Close the channel.
Parameters:
gst_webrtc_data_channel_send_data
gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel, GBytes * data)
Send data as a data message over channel.
GstWebRTC.WebRTCDataChannel.prototype.send_data
function GstWebRTC.WebRTCDataChannel.prototype.send_data(data: GLib.Bytes): {
// javascript wrapper for 'gst_webrtc_data_channel_send_data'
}
Send data as a data message over channel.
GstWebRTC.WebRTCDataChannel.send_data
def GstWebRTC.WebRTCDataChannel.send_data (self, data):
#python wrapper for 'gst_webrtc_data_channel_send_data'
Send data as a data message over channel.
gst_webrtc_data_channel_send_string
gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel, const gchar * str)
Send str as a string message over channel.
GstWebRTC.WebRTCDataChannel.prototype.send_string
function GstWebRTC.WebRTCDataChannel.prototype.send_string(str: String): {
// javascript wrapper for 'gst_webrtc_data_channel_send_string'
}
Send str as a string message over channel.
Parameters:
GstWebRTC.WebRTCDataChannel.send_string
def GstWebRTC.WebRTCDataChannel.send_string (self, str):
#python wrapper for 'gst_webrtc_data_channel_send_string'
Send str as a string message over channel.
Parameters:
Signals
on-buffered-amount-low
on_buffered_amount_low_callback (GstWebRTCDataChannel * self, gpointer user_data)
Parameters:
self
–
user_data
–
Flags: Run Last
on-buffered-amount-low
function on_buffered_amount_low_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
// javascript callback for the 'on-buffered-amount-low' signal
}
Parameters:
Flags: Run Last
on-buffered-amount-low
def on_buffered_amount_low_callback (self, *user_data):
#python callback for the 'on-buffered-amount-low' signal
Parameters:
Flags: Run Last
on-close
on_close_callback (GstWebRTCDataChannel * self, gpointer user_data)
Parameters:
self
–
user_data
–
Flags: Run Last
on-close
function on_close_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
// javascript callback for the 'on-close' signal
}
Parameters:
Flags: Run Last
on-close
def on_close_callback (self, *user_data):
#python callback for the 'on-close' signal
Parameters:
Flags: Run Last
on-error
on_error_callback (GstWebRTCDataChannel * self, GError * error, gpointer user_data)
Parameters:
self
–
error
–
the GError thrown
user_data
–
Flags: Run Last
on-error
function on_error_callback(self: GstWebRTC.WebRTCDataChannel, error: GLib.Error, user_data: Object): {
// javascript callback for the 'on-error' signal
}
Flags: Run Last
on-error
def on_error_callback (self, error, *user_data):
#python callback for the 'on-error' signal
Flags: Run Last
on-message-data
on_message_data_callback (GstWebRTCDataChannel * self, GBytes * data, gpointer user_data)
Parameters:
self
–
data
(
[nullable])
–
a GBytes of the data received
user_data
–
Flags: Run Last
on-message-data
function on_message_data_callback(self: GstWebRTC.WebRTCDataChannel, data: GLib.Bytes, user_data: Object): {
// javascript callback for the 'on-message-data' signal
}
Flags: Run Last
on-message-data
def on_message_data_callback (self, data, *user_data):
#python callback for the 'on-message-data' signal
Flags: Run Last
on-message-string
on_message_string_callback (GstWebRTCDataChannel * self, gchar * data, gpointer user_data)
Parameters:
self
–
data
(
[nullable])
–
the data received as a string
user_data
–
Flags: Run Last
on-message-string
function on_message_string_callback(self: GstWebRTC.WebRTCDataChannel, data: String, user_data: Object): {
// javascript callback for the 'on-message-string' signal
}
Parameters:
the data received as a string
Flags: Run Last
on-message-string
def on_message_string_callback (self, data, *user_data):
#python callback for the 'on-message-string' signal
Parameters:
the data received as a string
Flags: Run Last
on-open
on_open_callback (GstWebRTCDataChannel * self, gpointer user_data)
Parameters:
self
–
user_data
–
Flags: Run Last
on-open
function on_open_callback(self: GstWebRTC.WebRTCDataChannel, user_data: Object): {
// javascript callback for the 'on-open' signal
}
Parameters:
Flags: Run Last
on-open
def on_open_callback (self, *user_data):
#python callback for the 'on-open' signal
Parameters:
Flags: Run Last
Action Signals
close
g_signal_emit_by_name (self, "close", user_data);
Close the data channel
Parameters:
close
let ret = self.emit ("close", user_data);
Close the data channel
Parameters:
close
ret = self.emit ("close", user_data)
Close the data channel
Parameters:
send-data
g_signal_emit_by_name (self, "send-data", data, user_data);
Parameters:
send-data
let ret = self.emit ("send-data", data, user_data);
send-data
ret = self.emit ("send-data", data, user_data)
send-string
g_signal_emit_by_name (self, "send-string", data, user_data);
Parameters:
the data to send as a string
send-string
let ret = self.emit ("send-string", data, user_data);
Parameters:
the data to send as a string
send-string
ret = self.emit ("send-string", data, user_data)
Parameters:
the data to send as a string
Properties
GstWebRTCICETransport
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCICETransport
Class structure
GstWebRTCICETransportClass
GstWebRTC.WebRTCICETransportClass
GstWebRTC.WebRTCICETransportClass
GstWebRTC.WebRTCICETransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCICETransport
GstWebRTC.WebRTCICETransport
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCICETransport
Signals
on-new-candidate
on_new_candidate_callback (GstWebRTCICETransport * self, gchar * object, gpointer user_data)
Parameters:
self
–
object
–
user_data
–
Flags: Run Last
on-new-candidate
function on_new_candidate_callback(self: GstWebRTC.WebRTCICETransport, object: String, user_data: Object): {
// javascript callback for the 'on-new-candidate' signal
}
Parameters:
Flags: Run Last
on-new-candidate
def on_new_candidate_callback (self, object, *user_data):
#python callback for the 'on-new-candidate' signal
Parameters:
Flags: Run Last
on-selected-candidate-pair-change
on_selected_candidate_pair_change_callback (GstWebRTCICETransport * self, gpointer user_data)
Parameters:
self
–
user_data
–
Flags: Run Last
on-selected-candidate-pair-change
function on_selected_candidate_pair_change_callback(self: GstWebRTC.WebRTCICETransport, user_data: Object): {
// javascript callback for the 'on-selected-candidate-pair-change' signal
}
Parameters:
Flags: Run Last
on-selected-candidate-pair-change
def on_selected_candidate_pair_change_callback (self, *user_data):
#python callback for the 'on-selected-candidate-pair-change' signal
Parameters:
Flags: Run Last
Properties
GstWebRTCRTPReceiver
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCRTPReceiver
An object to track the receiving aspect of the stream
Mostly matches the WebRTC RTCRtpReceiver interface.
Class structure
GstWebRTCRTPReceiverClass
GstWebRTC.WebRTCRTPReceiverClass
GstWebRTC.WebRTCRTPReceiverClass
GstWebRTC.WebRTCRTPReceiver
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPReceiver
An object to track the receiving aspect of the stream
Mostly matches the WebRTC RTCRtpReceiver interface.
GstWebRTC.WebRTCRTPReceiver
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPReceiver
An object to track the receiving aspect of the stream
Mostly matches the WebRTC RTCRtpReceiver interface.
Properties
transport
“transport” GstWebRTC.WebRTCDTLSTransport
The DTLS transport for this receiver
Flags : Read
transport
“self.props.transport” GstWebRTC.WebRTCDTLSTransport
The DTLS transport for this receiver
Flags : Read
GstWebRTCRTPSender
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCRTPSender
An object to track the sending aspect of the stream
Mostly matches the WebRTC RTCRtpSender interface.
Class structure
GstWebRTCRTPSenderClass
GstWebRTC.WebRTCRTPSenderClass
GstWebRTC.WebRTCRTPSenderClass
GstWebRTC.WebRTCRTPSender
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPSender
An object to track the sending aspect of the stream
Mostly matches the WebRTC RTCRtpSender interface.
GstWebRTC.WebRTCRTPSender
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPSender
An object to track the sending aspect of the stream
Mostly matches the WebRTC RTCRtpSender interface.
Methods
gst_webrtc_rtp_sender_set_priority
gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender * sender, GstWebRTCPriorityType priority)
Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.
Since : 1.20
GstWebRTC.WebRTCRTPSender.prototype.set_priority
function GstWebRTC.WebRTCRTPSender.prototype.set_priority(priority: GstWebRTC.WebRTCPriorityType): {
// javascript wrapper for 'gst_webrtc_rtp_sender_set_priority'
}
Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.
Parameters:
The priority of this sender
Since : 1.20
GstWebRTC.WebRTCRTPSender.set_priority
def GstWebRTC.WebRTCRTPSender.set_priority (self, priority):
#python wrapper for 'gst_webrtc_rtp_sender_set_priority'
Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6.
Parameters:
The priority of this sender
Since : 1.20
Properties
priority
“priority” GstWebRTCPriorityType *
The priority from which to set the DSCP field on packets
Flags : Read / Write
priority
“priority” GstWebRTC.WebRTCPriorityType
The priority from which to set the DSCP field on packets
Flags : Read / Write
priority
“self.props.priority” GstWebRTC.WebRTCPriorityType
The priority from which to set the DSCP field on packets
Flags : Read / Write
transport
“self.props.transport” GstWebRTC.WebRTCDTLSTransport
The DTLS transport for this sender
Flags : Read
GstWebRTCRTPTransceiver
GObject ╰──GInitiallyUnowned ╰──GstObject ╰──GstWebRTCRTPTransceiver
Mostly matches the WebRTC RTCRtpTransceiver interface.
Class structure
GstWebRTCRTPTransceiverClass
GstWebRTC.WebRTCRTPTransceiverClass
GstWebRTC.WebRTCRTPTransceiverClass
GstWebRTC.WebRTCRTPTransceiver
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPTransceiver
Mostly matches the WebRTC RTCRtpTransceiver interface.
GstWebRTC.WebRTCRTPTransceiver
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──GstWebRTC.WebRTCRTPTransceiver
Mostly matches the WebRTC RTCRtpTransceiver interface.
Properties
codec-preferences
“codec-preferences” GstCaps *
Caps representing the codec preferences.
Flags : Read / Write
codec-preferences
“codec-preferences” Gst.Caps
Caps representing the codec preferences.
Flags : Read / Write
codec_preferences
“self.props.codec_preferences” Gst.Caps
Caps representing the codec preferences.
Flags : Read / Write
current-direction
“current-direction” GstWebRTCRTPTransceiverDirection *
The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.
Flags : Read
current-direction
“current-direction” GstWebRTC.WebRTCRTPTransceiverDirection
The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.
Flags : Read
current_direction
“self.props.current_direction” GstWebRTC.WebRTCRTPTransceiverDirection
The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property.
Flags : Read
direction
“direction” GstWebRTCRTPTransceiverDirection *
Direction of the transceiver.
Flags : Read / Write
direction
“direction” GstWebRTC.WebRTCRTPTransceiverDirection
Direction of the transceiver.
Flags : Read / Write
direction
“self.props.direction” GstWebRTC.WebRTCRTPTransceiverDirection
Direction of the transceiver.
Flags : Read / Write
kind
“self.props.kind” GstWebRTC.WebRTCKind
The kind of media this transceiver transports
Flags : Read
mid
“mid” gchar *
The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.
Flags : Read
mid
“mid” String
The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.
Flags : Read
mid
“self.props.mid” str
The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if neither a local or remote description has been applied, or if its associated m-line is rejected by either a remote offer or any answer.
Flags : Read
Enumerations
GstWebRTCBundlePolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GST_WEBRTC_BUNDLE_POLICY_NONE
(0)
–
none
GST_WEBRTC_BUNDLE_POLICY_BALANCED
(1)
–
balanced
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT
(2)
–
max-compat
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
(3)
–
max-bundle
GstWebRTC.WebRTCBundlePolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GstWebRTC.WebRTCBundlePolicy.NONE
(0)
–
none
GstWebRTC.WebRTCBundlePolicy.BALANCED
(1)
–
balanced
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
(2)
–
max-compat
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
(3)
–
max-bundle
GstWebRTC.WebRTCBundlePolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GstWebRTC.WebRTCBundlePolicy.NONE
(0)
–
none
GstWebRTC.WebRTCBundlePolicy.BALANCED
(1)
–
balanced
GstWebRTC.WebRTCBundlePolicy.MAX_COMPAT
(2)
–
max-compat
GstWebRTC.WebRTCBundlePolicy.MAX_BUNDLE
(3)
–
max-bundle
GstWebRTCDTLSSetup
Members
GST_WEBRTC_DTLS_SETUP_NONE
(0)
–
none
GST_WEBRTC_DTLS_SETUP_ACTPASS
(1)
–
actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE
(2)
–
sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE
(3)
–
recvonly
GstWebRTC.WebRTCDTLSSetup
Members
GstWebRTC.WebRTCDTLSSetup.NONE
(0)
–
none
GstWebRTC.WebRTCDTLSSetup.ACTPASS
(1)
–
actpass
GstWebRTC.WebRTCDTLSSetup.ACTIVE
(2)
–
sendonly
GstWebRTC.WebRTCDTLSSetup.PASSIVE
(3)
–
recvonly
GstWebRTC.WebRTCDTLSSetup
Members
GstWebRTC.WebRTCDTLSSetup.NONE
(0)
–
none
GstWebRTC.WebRTCDTLSSetup.ACTPASS
(1)
–
actpass
GstWebRTC.WebRTCDTLSSetup.ACTIVE
(2)
–
sendonly
GstWebRTC.WebRTCDTLSSetup.PASSIVE
(3)
–
recvonly
GstWebRTCDTLSTransportState
Members
GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW
(0)
–
new
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED
(1)
–
closed
GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED
(2)
–
failed
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING
(3)
–
connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED
(4)
–
connected
GstWebRTC.WebRTCDTLSTransportState
Members
GstWebRTC.WebRTCDTLSTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCDTLSTransportState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCDTLSTransportState.FAILED
(2)
–
failed
GstWebRTC.WebRTCDTLSTransportState.CONNECTING
(3)
–
connecting
GstWebRTC.WebRTCDTLSTransportState.CONNECTED
(4)
–
connected
GstWebRTC.WebRTCDTLSTransportState
Members
GstWebRTC.WebRTCDTLSTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCDTLSTransportState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCDTLSTransportState.FAILED
(2)
–
failed
GstWebRTC.WebRTCDTLSTransportState.CONNECTING
(3)
–
connecting
GstWebRTC.WebRTCDTLSTransportState.CONNECTED
(4)
–
connected
GstWebRTCDataChannelState
See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate
Members
GST_WEBRTC_DATA_CHANNEL_STATE_NEW
(0)
–
new
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING
(1)
–
connection
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN
(2)
–
open
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
(3)
–
closing
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED
(4)
–
closed
GstWebRTC.WebRTCDataChannelState
See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate
Members
GstWebRTC.WebRTCDataChannelState.NEW
(0)
–
new
GstWebRTC.WebRTCDataChannelState.CONNECTING
(1)
–
connection
GstWebRTC.WebRTCDataChannelState.OPEN
(2)
–
open
GstWebRTC.WebRTCDataChannelState.CLOSING
(3)
–
closing
GstWebRTC.WebRTCDataChannelState.CLOSED
(4)
–
closed
GstWebRTC.WebRTCDataChannelState
See http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate
Members
GstWebRTC.WebRTCDataChannelState.NEW
(0)
–
new
GstWebRTC.WebRTCDataChannelState.CONNECTING
(1)
–
connection
GstWebRTC.WebRTCDataChannelState.OPEN
(2)
–
open
GstWebRTC.WebRTCDataChannelState.CLOSING
(3)
–
closing
GstWebRTC.WebRTCDataChannelState.CLOSED
(4)
–
closed
GstWebRTCError
See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.
Members
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE
(0)
–
data-channel-failure
GST_WEBRTC_ERROR_DTLS_FAILURE
(1)
–
dtls-failure
GST_WEBRTC_ERROR_FINGERPRINT_FAILURE
(2)
–
fingerprint-failure
GST_WEBRTC_ERROR_SCTP_FAILURE
(3)
–
sctp-failure
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR
(4)
–
sdp-syntax-error
GST_WEBRTC_ERROR_HARDWARE_ENCODER_NOT_AVAILABLE
(5)
–
hardware-encoder-not-available
GST_WEBRTC_ERROR_ENCODER_ERROR
(6)
–
encoder-error
GST_WEBRTC_ERROR_INVALID_STATE
(7)
–
invalid-state (part of WebIDL specification)
GST_WEBRTC_ERROR_INTERNAL_FAILURE
(8)
–
GStreamer-specific failure, not matching any other value from the specification
GstWebRTC.WebRTCError
See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.
Members
GstWebRTC.WebRTCError.DATA_CHANNEL_FAILURE
(0)
–
data-channel-failure
GstWebRTC.WebRTCError.DTLS_FAILURE
(1)
–
dtls-failure
GstWebRTC.WebRTCError.FINGERPRINT_FAILURE
(2)
–
fingerprint-failure
GstWebRTC.WebRTCError.SCTP_FAILURE
(3)
–
sctp-failure
GstWebRTC.WebRTCError.SDP_SYNTAX_ERROR
(4)
–
sdp-syntax-error
GstWebRTC.WebRTCError.HARDWARE_ENCODER_NOT_AVAILABLE
(5)
–
hardware-encoder-not-available
GstWebRTC.WebRTCError.ENCODER_ERROR
(6)
–
encoder-error
GstWebRTC.WebRTCError.INVALID_STATE
(7)
–
invalid-state (part of WebIDL specification)
GstWebRTC.WebRTCError.INTERNAL_FAILURE
(8)
–
GStreamer-specific failure, not matching any other value from the specification
GstWebRTC.WebRTCError
See https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype for more information.
Members
GstWebRTC.WebRTCError.DATA_CHANNEL_FAILURE
(0)
–
data-channel-failure
GstWebRTC.WebRTCError.DTLS_FAILURE
(1)
–
dtls-failure
GstWebRTC.WebRTCError.FINGERPRINT_FAILURE
(2)
–
fingerprint-failure
GstWebRTC.WebRTCError.SCTP_FAILURE
(3)
–
sctp-failure
GstWebRTC.WebRTCError.SDP_SYNTAX_ERROR
(4)
–
sdp-syntax-error
GstWebRTC.WebRTCError.HARDWARE_ENCODER_NOT_AVAILABLE
(5)
–
hardware-encoder-not-available
GstWebRTC.WebRTCError.ENCODER_ERROR
(6)
–
encoder-error
GstWebRTC.WebRTCError.INVALID_STATE
(7)
–
invalid-state (part of WebIDL specification)
GstWebRTC.WebRTCError.INTERNAL_FAILURE
(8)
–
GStreamer-specific failure, not matching any other value from the specification
GstWebRTCFECType
Members
GST_WEBRTC_FEC_TYPE_NONE
(0)
–
none
GST_WEBRTC_FEC_TYPE_ULP_RED
(1)
–
ulpfec + red
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
(0)
–
none
GstWebRTC.WebRTCFECType.ULP_RED
(1)
–
ulpfec + red
GstWebRTC.WebRTCFECType
Members
GstWebRTC.WebRTCFECType.NONE
(0)
–
none
GstWebRTC.WebRTCFECType.ULP_RED
(1)
–
ulpfec + red
GstWebRTCICEComponent
Members
GST_WEBRTC_ICE_COMPONENT_RTP
(0)
–
RTP component
GST_WEBRTC_ICE_COMPONENT_RTCP
(1)
–
RTCP component
GstWebRTC.WebRTCICEComponent
Members
GstWebRTC.WebRTCICEComponent.RTP
(0)
–
RTP component
GstWebRTC.WebRTCICEComponent.RTCP
(1)
–
RTCP component
GstWebRTC.WebRTCICEComponent
Members
GstWebRTC.WebRTCICEComponent.RTP
(0)
–
RTP component
GstWebRTC.WebRTCICEComponent.RTCP
(1)
–
RTCP component
GstWebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GST_WEBRTC_ICE_CONNECTION_STATE_NEW
(0)
–
new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING
(1)
–
checking
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED
(2)
–
connected
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED
(3)
–
completed
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED
(4)
–
failed
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED
(5)
–
disconnected
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED
(6)
–
closed
GstWebRTC.WebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GstWebRTC.WebRTCICEConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCICEConnectionState.CHECKING
(1)
–
checking
GstWebRTC.WebRTCICEConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCICEConnectionState.COMPLETED
(3)
–
completed
GstWebRTC.WebRTCICEConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCICEConnectionState.DISCONNECTED
(5)
–
disconnected
GstWebRTC.WebRTCICEConnectionState.CLOSED
(6)
–
closed
GstWebRTC.WebRTCICEConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
Members
GstWebRTC.WebRTCICEConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCICEConnectionState.CHECKING
(1)
–
checking
GstWebRTC.WebRTCICEConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCICEConnectionState.COMPLETED
(3)
–
completed
GstWebRTC.WebRTCICEConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCICEConnectionState.DISCONNECTED
(5)
–
disconnected
GstWebRTC.WebRTCICEConnectionState.CLOSED
(6)
–
closed
GstWebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GST_WEBRTC_ICE_GATHERING_STATE_NEW
(0)
–
new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING
(1)
–
gathering
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE
(2)
–
complete
GstWebRTC.WebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GstWebRTC.WebRTCICEGatheringState.NEW
(0)
–
new
GstWebRTC.WebRTCICEGatheringState.GATHERING
(1)
–
gathering
GstWebRTC.WebRTCICEGatheringState.COMPLETE
(2)
–
complete
GstWebRTC.WebRTCICEGatheringState
See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
Members
GstWebRTC.WebRTCICEGatheringState.NEW
(0)
–
new
GstWebRTC.WebRTCICEGatheringState.GATHERING
(1)
–
gathering
GstWebRTC.WebRTCICEGatheringState.COMPLETE
(2)
–
complete
GstWebRTCICERole
Members
GST_WEBRTC_ICE_ROLE_CONTROLLED
(0)
–
controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING
(1)
–
controlling
GstWebRTC.WebRTCICERole
Members
GstWebRTC.WebRTCICERole.CONTROLLED
(0)
–
controlled
GstWebRTC.WebRTCICERole.CONTROLLING
(1)
–
controlling
GstWebRTC.WebRTCICERole
Members
GstWebRTC.WebRTCICERole.CONTROLLED
(0)
–
controlled
GstWebRTC.WebRTCICERole.CONTROLLING
(1)
–
controlling
GstWebRTCICETransportPolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL
(0)
–
all
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY
(1)
–
relay
GstWebRTC.WebRTCICETransportPolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
(0)
–
all
GstWebRTC.WebRTCICETransportPolicy.RELAY
(1)
–
relay
GstWebRTC.WebRTCICETransportPolicy
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information.
Members
GstWebRTC.WebRTCICETransportPolicy.ALL
(0)
–
all
GstWebRTC.WebRTCICETransportPolicy.RELAY
(1)
–
relay
GstWebRTCKind
https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
Members
GST_WEBRTC_KIND_UNKNOWN
(0)
–
Kind has not yet been set
GST_WEBRTC_KIND_AUDIO
(1)
–
Kind is audio
GST_WEBRTC_KIND_VIDEO
(2)
–
Kind is audio
GstWebRTC.WebRTCKind
https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
Members
GstWebRTC.WebRTCKind.UNKNOWN
(0)
–
Kind has not yet been set
GstWebRTC.WebRTCKind.AUDIO
(1)
–
Kind is audio
GstWebRTC.WebRTCKind.VIDEO
(2)
–
Kind is audio
GstWebRTC.WebRTCKind
https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind
Members
GstWebRTC.WebRTCKind.UNKNOWN
(0)
–
Kind has not yet been set
GstWebRTC.WebRTCKind.AUDIO
(1)
–
Kind is audio
GstWebRTC.WebRTCKind.VIDEO
(2)
–
Kind is audio
GstWebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GST_WEBRTC_PEER_CONNECTION_STATE_NEW
(0)
–
new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING
(1)
–
connecting
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED
(2)
–
connected
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED
(3)
–
disconnected
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED
(4)
–
failed
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED
(5)
–
closed
GstWebRTC.WebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GstWebRTC.WebRTCPeerConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCPeerConnectionState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCPeerConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED
(3)
–
disconnected
GstWebRTC.WebRTCPeerConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCPeerConnectionState.CLOSED
(5)
–
closed
GstWebRTC.WebRTCPeerConnectionState
See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
Members
GstWebRTC.WebRTCPeerConnectionState.NEW
(0)
–
new
GstWebRTC.WebRTCPeerConnectionState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCPeerConnectionState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCPeerConnectionState.DISCONNECTED
(3)
–
disconnected
GstWebRTC.WebRTCPeerConnectionState.FAILED
(4)
–
failed
GstWebRTC.WebRTCPeerConnectionState.CLOSED
(5)
–
closed
GstWebRTCPriorityType
See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype
Members
GST_WEBRTC_PRIORITY_TYPE_VERY_LOW
(1)
–
very-low
GST_WEBRTC_PRIORITY_TYPE_LOW
(2)
–
low
GST_WEBRTC_PRIORITY_TYPE_MEDIUM
(3)
–
medium
GST_WEBRTC_PRIORITY_TYPE_HIGH
(4)
–
high
GstWebRTC.WebRTCPriorityType
See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
(1)
–
very-low
GstWebRTC.WebRTCPriorityType.LOW
(2)
–
low
GstWebRTC.WebRTCPriorityType.MEDIUM
(3)
–
medium
GstWebRTC.WebRTCPriorityType.HIGH
(4)
–
high
GstWebRTC.WebRTCPriorityType
See http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype
Members
GstWebRTC.WebRTCPriorityType.VERY_LOW
(1)
–
very-low
GstWebRTC.WebRTCPriorityType.LOW
(2)
–
low
GstWebRTC.WebRTCPriorityType.MEDIUM
(3)
–
medium
GstWebRTC.WebRTCPriorityType.HIGH
(4)
–
high
GstWebRTCRTPTransceiverDirection
Members
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
(0)
–
none
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
(1)
–
inactive
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY
(2)
–
sendonly
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
(3)
–
recvonly
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV
(4)
–
sendrecv
GstWebRTC.WebRTCRTPTransceiverDirection
Members
GstWebRTC.WebRTCRTPTransceiverDirection.NONE
(0)
–
none
GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE
(1)
–
inactive
GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
(2)
–
sendonly
GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
(3)
–
recvonly
GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
(4)
–
sendrecv
GstWebRTC.WebRTCRTPTransceiverDirection
Members
GstWebRTC.WebRTCRTPTransceiverDirection.NONE
(0)
–
none
GstWebRTC.WebRTCRTPTransceiverDirection.INACTIVE
(1)
–
inactive
GstWebRTC.WebRTCRTPTransceiverDirection.SENDONLY
(2)
–
sendonly
GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
(3)
–
recvonly
GstWebRTC.WebRTCRTPTransceiverDirection.SENDRECV
(4)
–
sendrecv
GstWebRTCSCTPTransportState
See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate
Members
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW
(0)
–
new
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING
(1)
–
connecting
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED
(2)
–
connected
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED
(3)
–
closed
GstWebRTC.WebRTCSCTPTransportState
See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCSCTPTransportState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCSCTPTransportState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCSCTPTransportState.CLOSED
(3)
–
closed
GstWebRTC.WebRTCSCTPTransportState
See http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate
Members
GstWebRTC.WebRTCSCTPTransportState.NEW
(0)
–
new
GstWebRTC.WebRTCSCTPTransportState.CONNECTING
(1)
–
connecting
GstWebRTC.WebRTCSCTPTransportState.CONNECTED
(2)
–
connected
GstWebRTC.WebRTCSCTPTransportState.CLOSED
(3)
–
closed
GstWebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GST_WEBRTC_SDP_TYPE_OFFER
(1)
–
offer
GST_WEBRTC_SDP_TYPE_PRANSWER
(2)
–
pranswer
GST_WEBRTC_SDP_TYPE_ANSWER
(3)
–
answer
GST_WEBRTC_SDP_TYPE_ROLLBACK
(4)
–
rollback
GstWebRTC.WebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GstWebRTC.WebRTCSDPType.OFFER
(1)
–
offer
GstWebRTC.WebRTCSDPType.PRANSWER
(2)
–
pranswer
GstWebRTC.WebRTCSDPType.ANSWER
(3)
–
answer
GstWebRTC.WebRTCSDPType.ROLLBACK
(4)
–
rollback
GstWebRTC.WebRTCSDPType
See http://w3c.github.io/webrtc-pc/#rtcsdptype
Members
GstWebRTC.WebRTCSDPType.OFFER
(1)
–
offer
GstWebRTC.WebRTCSDPType.PRANSWER
(2)
–
pranswer
GstWebRTC.WebRTCSDPType.ANSWER
(3)
–
answer
GstWebRTC.WebRTCSDPType.ROLLBACK
(4)
–
rollback
GstWebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GST_WEBRTC_SIGNALING_STATE_STABLE
(0)
–
stable
GST_WEBRTC_SIGNALING_STATE_CLOSED
(1)
–
closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTC.WebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GstWebRTC.WebRTCSignalingState.STABLE
(0)
–
stable
GstWebRTC.WebRTCSignalingState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTC.WebRTCSignalingState
See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
Members
GstWebRTC.WebRTCSignalingState.STABLE
(0)
–
stable
GstWebRTC.WebRTCSignalingState.CLOSED
(1)
–
closed
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_OFFER
(2)
–
have-local-offer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_OFFER
(3)
–
have-remote-offer
GstWebRTC.WebRTCSignalingState.HAVE_LOCAL_PRANSWER
(4)
–
have-local-pranswer
GstWebRTC.WebRTCSignalingState.HAVE_REMOTE_PRANSWER
(5)
–
have-remote-pranswer
GstWebRTCStatsType
Members
GST_WEBRTC_STATS_CODEC
(1)
–
codec
GST_WEBRTC_STATS_INBOUND_RTP
(2)
–
inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP
(3)
–
outbound-rtp
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GST_WEBRTC_STATS_CSRC
(6)
–
csrc
GST_WEBRTC_STATS_PEER_CONNECTION
(7)
–
peer-connectiion
GST_WEBRTC_STATS_DATA_CHANNEL
(8)
–
data-channel
GST_WEBRTC_STATS_STREAM
(9)
–
stream
GST_WEBRTC_STATS_TRANSPORT
(10)
–
transport
GST_WEBRTC_STATS_CANDIDATE_PAIR
(11)
–
candidate-pair
GST_WEBRTC_STATS_LOCAL_CANDIDATE
(12)
–
local-candidate
GST_WEBRTC_STATS_REMOTE_CANDIDATE
(13)
–
remote-candidate
GST_WEBRTC_STATS_CERTIFICATE
(14)
–
certificate
GstWebRTC.WebRTCStatsType
Members
GstWebRTC.WebRTCStatsType.CODEC
(1)
–
codec
GstWebRTC.WebRTCStatsType.INBOUND_RTP
(2)
–
inbound-rtp
GstWebRTC.WebRTCStatsType.OUTBOUND_RTP
(3)
–
outbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GstWebRTC.WebRTCStatsType.CSRC
(6)
–
csrc
GstWebRTC.WebRTCStatsType.PEER_CONNECTION
(7)
–
peer-connectiion
GstWebRTC.WebRTCStatsType.DATA_CHANNEL
(8)
–
data-channel
GstWebRTC.WebRTCStatsType.STREAM
(9)
–
stream
GstWebRTC.WebRTCStatsType.TRANSPORT
(10)
–
transport
GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR
(11)
–
candidate-pair
GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE
(12)
–
local-candidate
GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE
(13)
–
remote-candidate
GstWebRTC.WebRTCStatsType.CERTIFICATE
(14)
–
certificate
GstWebRTC.WebRTCStatsType
Members
GstWebRTC.WebRTCStatsType.CODEC
(1)
–
codec
GstWebRTC.WebRTCStatsType.INBOUND_RTP
(2)
–
inbound-rtp
GstWebRTC.WebRTCStatsType.OUTBOUND_RTP
(3)
–
outbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_INBOUND_RTP
(4)
–
remote-inbound-rtp
GstWebRTC.WebRTCStatsType.REMOTE_OUTBOUND_RTP
(5)
–
remote-outbound-rtp
GstWebRTC.WebRTCStatsType.CSRC
(6)
–
csrc
GstWebRTC.WebRTCStatsType.PEER_CONNECTION
(7)
–
peer-connectiion
GstWebRTC.WebRTCStatsType.DATA_CHANNEL
(8)
–
data-channel
GstWebRTC.WebRTCStatsType.STREAM
(9)
–
stream
GstWebRTC.WebRTCStatsType.TRANSPORT
(10)
–
transport
GstWebRTC.WebRTCStatsType.CANDIDATE_PAIR
(11)
–
candidate-pair
GstWebRTC.WebRTCStatsType.LOCAL_CANDIDATE
(12)
–
local-candidate
GstWebRTC.WebRTCStatsType.REMOTE_CANDIDATE
(13)
–
remote-candidate
GstWebRTC.WebRTCStatsType.CERTIFICATE
(14)
–
certificate
Constants
GST_WEBRTC_API
#define GST_WEBRTC_API GST_API_EXPORT /* from config.h */
GST_WEBRTC_ERROR
#define GST_WEBRTC_ERROR gst_webrtc_error_quark ()
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