rtsp stream

GstRtspServer.RTSPStream

GObject.Object
    ╰──GstRtspServer.RTSPStream

GstRtspServer.RTSPStream.prototype.new

function GstRtspServer.RTSPStream.prototype.new(idx: Number, payloader: Gst.Element, pad: Gst.Pad): {
    // javascript wrapper for 'gst_rtsp_stream_new'
}

Create a new media stream with index idx that handles RTP data on pad and has a payloader element payloader if pad is a source pad or a depayloader element payloader if pad is a sink pad.

GstRtspServer.RTSPStream.prototype.add_multicast_client_address

function GstRtspServer.RTSPStream.prototype.add_multicast_client_address(destination: String, rtp_port: Number, rtcp_port: Number, family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_add_multicast_client_address'
}

Add multicast client address to stream. At this point, the sockets that will stream RTP and RTCP data to destination are supposed to be allocated.

Since : 1.16

GstRtspServer.RTSPStream.prototype.add_transport

function GstRtspServer.RTSPStream.prototype.add_transport(trans: GstRtspServer.RTSPStreamTransport): {
    // javascript wrapper for 'gst_rtsp_stream_add_transport'
}

Add the transport in trans to stream. The media of stream will then also be send to the values configured in trans. Adding the same transport twice will not add it a second time.

stream must be joined to a bin.

trans must contain a valid GstRtsp.RTSPTransport.

GstRtspServer.RTSPStream.prototype.allocate_udp_sockets

function GstRtspServer.RTSPStream.prototype.allocate_udp_sockets(family: Gio.SocketFamily, transport: GstRtsp.RTSPTransport, use_client_settings: Number): {
    // javascript wrapper for 'gst_rtsp_stream_allocate_udp_sockets'
}

Allocates RTP and RTCP ports.

GstRtspServer.RTSPStream.prototype.complete_stream

function GstRtspServer.RTSPStream.prototype.complete_stream(transport: GstRtsp.RTSPTransport): {
    // javascript wrapper for 'gst_rtsp_stream_complete_stream'
}

Add a receiver and sender part to the pipeline based on the transport from SETUP.

Since : 1.14

GstRtspServer.RTSPStream.prototype.get_address_pool

function GstRtspServer.RTSPStream.prototype.get_address_pool(): {
    // javascript wrapper for 'gst_rtsp_stream_get_address_pool'
}

Get the GstRtspServer.RTSPAddressPool used as the address pool of stream.

GstRtspServer.RTSPStream.prototype.get_buffer_size

function GstRtspServer.RTSPStream.prototype.get_buffer_size(): {
    // javascript wrapper for 'gst_rtsp_stream_get_buffer_size'
}

Get the size of the UDP transmission buffer (in bytes)

Since : 1.6

GstRtspServer.RTSPStream.prototype.get_caps

function GstRtspServer.RTSPStream.prototype.get_caps(): {
    // javascript wrapper for 'gst_rtsp_stream_get_caps'
}

Retrieve the current caps of stream.

GstRtspServer.RTSPStream.prototype.get_control

function GstRtspServer.RTSPStream.prototype.get_control(): {
    // javascript wrapper for 'gst_rtsp_stream_get_control'
}

Get the control string to identify this stream.

GstRtspServer.RTSPStream.prototype.get_current_seqnum

function GstRtspServer.RTSPStream.prototype.get_current_seqnum(): {
    // javascript wrapper for 'gst_rtsp_stream_get_current_seqnum'
}

GstRtspServer.RTSPStream.prototype.get_dscp_qos

function GstRtspServer.RTSPStream.prototype.get_dscp_qos(): {
    // javascript wrapper for 'gst_rtsp_stream_get_dscp_qos'
}

Get the configured DSCP QoS in of the outgoing sockets.

GstRtspServer.RTSPStream.prototype.get_index

function GstRtspServer.RTSPStream.prototype.get_index(): {
    // javascript wrapper for 'gst_rtsp_stream_get_index'
}

Get the stream index.

Return: the stream index.

GstRtspServer.RTSPStream.prototype.get_joined_bin

function GstRtspServer.RTSPStream.prototype.get_joined_bin(): {
    // javascript wrapper for 'gst_rtsp_stream_get_joined_bin'
}

Get the previous joined bin with GstRtspServer.RTSPStream.prototype.join_bin or NULL.

Return: (transfer full) (nullable): the joined bin or NULL.

GstRtspServer.RTSPStream.prototype.get_max_mcast_ttl

function GstRtspServer.RTSPStream.prototype.get_max_mcast_ttl(): {
    // javascript wrapper for 'gst_rtsp_stream_get_max_mcast_ttl'
}

Get the the maximum time-to-live value of outgoing multicast packets.

Since : 1.16

GstRtspServer.RTSPStream.prototype.get_mtu

function GstRtspServer.RTSPStream.prototype.get_mtu(): {
    // javascript wrapper for 'gst_rtsp_stream_get_mtu'
}

Get the configured MTU in the payloader of stream.

GstRtspServer.RTSPStream.prototype.get_multicast_address

function GstRtspServer.RTSPStream.prototype.get_multicast_address(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_multicast_address'
}

Get the multicast address of stream for family. The original GstRtspServer.RTSPAddress is cached and copy is returned, so freeing the return value won't release the address from the pool.

GstRtspServer.RTSPStream.prototype.get_multicast_client_addresses

function GstRtspServer.RTSPStream.prototype.get_multicast_client_addresses(): {
    // javascript wrapper for 'gst_rtsp_stream_get_multicast_client_addresses'
}

Get all multicast client addresses that RTP data will be sent to

Since : 1.16

GstRtspServer.RTSPStream.prototype.get_multicast_iface

function GstRtspServer.RTSPStream.prototype.get_multicast_iface(): {
    // javascript wrapper for 'gst_rtsp_stream_get_multicast_iface'
}

Get the multicast interface used for stream.

GstRtspServer.RTSPStream.prototype.get_profiles

function GstRtspServer.RTSPStream.prototype.get_profiles(): {
    // javascript wrapper for 'gst_rtsp_stream_get_profiles'
}

Get the allowed profiles of stream.

GstRtspServer.RTSPStream.prototype.get_protocols

function GstRtspServer.RTSPStream.prototype.get_protocols(): {
    // javascript wrapper for 'gst_rtsp_stream_get_protocols'
}

Get the allowed protocols of stream.

GstRtspServer.RTSPStream.prototype.get_pt

function GstRtspServer.RTSPStream.prototype.get_pt(): {
    // javascript wrapper for 'gst_rtsp_stream_get_pt'
}

Get the stream payload type.

Return: the stream payload type.

GstRtspServer.RTSPStream.prototype.get_publish_clock_mode

function GstRtspServer.RTSPStream.prototype.get_publish_clock_mode(): {
    // javascript wrapper for 'gst_rtsp_stream_get_publish_clock_mode'
}

Gets if and how the stream clock should be published according to RFC7273.

Since : 1.8

GstRtspServer.RTSPStream.prototype.get_rate_control

function GstRtspServer.RTSPStream.prototype.get_rate_control(): {
    // javascript wrapper for 'gst_rtsp_stream_get_rate_control'
}

Since : 1.18

GstRtspServer.RTSPStream.prototype.get_rates

function GstRtspServer.RTSPStream.prototype.get_rates(): {
    // javascript wrapper for 'gst_rtsp_stream_get_rates'
}

Retrieve the current rate and/or applied_rate.

Since : 1.18

GstRtspServer.RTSPStream.prototype.get_retransmission_pt

function GstRtspServer.RTSPStream.prototype.get_retransmission_pt(): {
    // javascript wrapper for 'gst_rtsp_stream_get_retransmission_pt'
}

Get the payload-type used for retransmission of this stream

GstRtspServer.RTSPStream.prototype.get_retransmission_time

function GstRtspServer.RTSPStream.prototype.get_retransmission_time(): {
    // javascript wrapper for 'gst_rtsp_stream_get_retransmission_time'
}

Get the amount of time to store retransmission data.

GstRtspServer.RTSPStream.prototype.get_rtcp_multicast_socket

function GstRtspServer.RTSPStream.prototype.get_rtcp_multicast_socket(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtcp_multicast_socket'
}

Get the multicast RTCP socket from stream for a family.

Since : 1.14

GstRtspServer.RTSPStream.prototype.get_rtcp_socket

function GstRtspServer.RTSPStream.prototype.get_rtcp_socket(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtcp_socket'
}

Get the RTCP socket from stream for a family.

stream must be joined to a bin.

GstRtspServer.RTSPStream.prototype.get_rtp_multicast_socket

function GstRtspServer.RTSPStream.prototype.get_rtp_multicast_socket(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtp_multicast_socket'
}

Get the multicast RTP socket from stream for a family.

GstRtspServer.RTSPStream.prototype.get_rtp_socket

function GstRtspServer.RTSPStream.prototype.get_rtp_socket(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtp_socket'
}

Get the RTP socket from stream for a family.

stream must be joined to a bin.

GstRtspServer.RTSPStream.prototype.get_rtpinfo

function GstRtspServer.RTSPStream.prototype.get_rtpinfo(): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtpinfo'
}

Retrieve the current rtptime, seq and running-time. This is used to construct a RTPInfo reply header.

GstRtspServer.RTSPStream.prototype.get_rtpsession

function GstRtspServer.RTSPStream.prototype.get_rtpsession(): {
    // javascript wrapper for 'gst_rtsp_stream_get_rtpsession'
}

Get the RTP session of this stream.

GstRtspServer.RTSPStream.prototype.get_server_port

function GstRtspServer.RTSPStream.prototype.get_server_port(family: Gio.SocketFamily): {
    // javascript wrapper for 'gst_rtsp_stream_get_server_port'
}

Fill server_port with the port pair used by the server. This function can only be called when stream has been joined.

GstRtspServer.RTSPStream.prototype.get_sinkpad

function GstRtspServer.RTSPStream.prototype.get_sinkpad(): {
    // javascript wrapper for 'gst_rtsp_stream_get_sinkpad'
}

Get the sinkpad associated with stream.

GstRtspServer.RTSPStream.prototype.get_srcpad

function GstRtspServer.RTSPStream.prototype.get_srcpad(): {
    // javascript wrapper for 'gst_rtsp_stream_get_srcpad'
}

Get the srcpad associated with stream.

GstRtspServer.RTSPStream.prototype.get_srtp_encoder

function GstRtspServer.RTSPStream.prototype.get_srtp_encoder(): {
    // javascript wrapper for 'gst_rtsp_stream_get_srtp_encoder'
}

Get the SRTP encoder for this stream.

GstRtspServer.RTSPStream.prototype.get_ssrc

function GstRtspServer.RTSPStream.prototype.get_ssrc(): {
    // javascript wrapper for 'gst_rtsp_stream_get_ssrc'
}

Get the SSRC used by the RTP session of this stream. This function can only be called when stream has been joined.

GstRtspServer.RTSPStream.prototype.get_ulpfec_enabled

function GstRtspServer.RTSPStream.prototype.get_ulpfec_enabled(): {
    // javascript wrapper for 'gst_rtsp_stream_get_ulpfec_enabled'
}

GstRtspServer.RTSPStream.prototype.get_ulpfec_percentage

function GstRtspServer.RTSPStream.prototype.get_ulpfec_percentage(): {
    // javascript wrapper for 'gst_rtsp_stream_get_ulpfec_percentage'
}

Since : 1.16

GstRtspServer.RTSPStream.prototype.get_ulpfec_pt

function GstRtspServer.RTSPStream.prototype.get_ulpfec_pt(): {
    // javascript wrapper for 'gst_rtsp_stream_get_ulpfec_pt'
}

Since : 1.16

GstRtspServer.RTSPStream.prototype.handle_keymgmt

function GstRtspServer.RTSPStream.prototype.handle_keymgmt(keymgmt: String): {
    // javascript wrapper for 'gst_rtsp_stream_handle_keymgmt'
}

Parse and handle a KeyMgmt header.

Since : 1.16

GstRtspServer.RTSPStream.prototype.has_control

function GstRtspServer.RTSPStream.prototype.has_control(control: String): {
    // javascript wrapper for 'gst_rtsp_stream_has_control'
}

Check if stream has the control string control.

GstRtspServer.RTSPStream.prototype.is_bind_mcast_address

function GstRtspServer.RTSPStream.prototype.is_bind_mcast_address(): {
    // javascript wrapper for 'gst_rtsp_stream_is_bind_mcast_address'
}

Check if multicast sockets are configured to be bound to multicast addresses.

Since : 1.16

GstRtspServer.RTSPStream.prototype.is_blocking

function GstRtspServer.RTSPStream.prototype.is_blocking(): {
    // javascript wrapper for 'gst_rtsp_stream_is_blocking'
}

Check if stream is blocking on a Gst.Buffer.

GstRtspServer.RTSPStream.prototype.is_client_side

function GstRtspServer.RTSPStream.prototype.is_client_side(): {
    // javascript wrapper for 'gst_rtsp_stream_is_client_side'
}

See GstRtspServer.RTSPStream.prototype.set_client_side

GstRtspServer.RTSPStream.prototype.is_complete

function GstRtspServer.RTSPStream.prototype.is_complete(): {
    // javascript wrapper for 'gst_rtsp_stream_is_complete'
}

Checks whether the stream is complete, contains the receiver and the sender parts. As the stream contains sink(s) element(s), it's possible to perform seek operations on it.

Since : 1.14

GstRtspServer.RTSPStream.prototype.is_receiver

function GstRtspServer.RTSPStream.prototype.is_receiver(): {
    // javascript wrapper for 'gst_rtsp_stream_is_receiver'
}

Checks whether the stream is a receiver.

Since : 1.14

GstRtspServer.RTSPStream.prototype.is_sender

function GstRtspServer.RTSPStream.prototype.is_sender(): {
    // javascript wrapper for 'gst_rtsp_stream_is_sender'
}

Checks whether the stream is a sender.

Since : 1.14

GstRtspServer.RTSPStream.prototype.is_transport_supported

function GstRtspServer.RTSPStream.prototype.is_transport_supported(transport: GstRtsp.RTSPTransport): {
    // javascript wrapper for 'gst_rtsp_stream_is_transport_supported'
}

Check if transport can be handled by stream

GstRtspServer.RTSPStream.prototype.join_bin

function GstRtspServer.RTSPStream.prototype.join_bin(bin: Gst.Bin, rtpbin: Gst.Element, state: Gst.State): {
    // javascript wrapper for 'gst_rtsp_stream_join_bin'
}

Join the Gst.Bin bin that contains the element rtpbin.

stream will link to rtpbin, which must be inside bin. The elements added to bin will be set to the state given in state.

GstRtspServer.RTSPStream.prototype.leave_bin

function GstRtspServer.RTSPStream.prototype.leave_bin(bin: Gst.Bin, rtpbin: Gst.Element): {
    // javascript wrapper for 'gst_rtsp_stream_leave_bin'
}

Remove the elements of stream from bin.

Return: true on success.

GstRtspServer.RTSPStream.prototype.query_position

function GstRtspServer.RTSPStream.prototype.query_position(): {
    // javascript wrapper for 'gst_rtsp_stream_query_position'
}

Query the position of the stream in Gst.Format.TIME. This only considers the RTP parts of the pipeline and not the RTCP parts.

GstRtspServer.RTSPStream.prototype.query_stop

function GstRtspServer.RTSPStream.prototype.query_stop(): {
    // javascript wrapper for 'gst_rtsp_stream_query_stop'
}

Query the stop of the stream in Gst.Format.TIME. This only considers the RTP parts of the pipeline and not the RTCP parts.

GstRtspServer.RTSPStream.prototype.recv_rtcp

function GstRtspServer.RTSPStream.prototype.recv_rtcp(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtsp_stream_recv_rtcp'
}

Handle an RTCP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport.

This function takes ownership of buffer.

GstRtspServer.RTSPStream.prototype.recv_rtp

function GstRtspServer.RTSPStream.prototype.recv_rtp(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtsp_stream_recv_rtp'
}

Handle an RTP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport.

This function takes ownership of buffer.

GstRtspServer.RTSPStream.prototype.remove_transport

function GstRtspServer.RTSPStream.prototype.remove_transport(trans: GstRtspServer.RTSPStreamTransport): {
    // javascript wrapper for 'gst_rtsp_stream_remove_transport'
}

Remove the transport in trans from stream. The media of stream will not be sent to the values configured in trans.

stream must be joined to a bin.

trans must contain a valid GstRtsp.RTSPTransport.

GstRtspServer.RTSPStream.prototype.request_aux_receiver

function GstRtspServer.RTSPStream.prototype.request_aux_receiver(sessid: Number): {
    // javascript wrapper for 'gst_rtsp_stream_request_aux_receiver'
}

Creating a rtxreceive bin

Since : 1.16

GstRtspServer.RTSPStream.prototype.request_aux_sender

function GstRtspServer.RTSPStream.prototype.request_aux_sender(sessid: Number): {
    // javascript wrapper for 'gst_rtsp_stream_request_aux_sender'
}

Creating a rtxsend bin

Since : 1.6

GstRtspServer.RTSPStream.prototype.request_ulpfec_decoder

function GstRtspServer.RTSPStream.prototype.request_ulpfec_decoder(rtpbin: Gst.Element, sessid: Number): {
    // javascript wrapper for 'gst_rtsp_stream_request_ulpfec_decoder'
}

Creating a rtpulpfecdec element

Since : 1.16

GstRtspServer.RTSPStream.prototype.request_ulpfec_encoder

function GstRtspServer.RTSPStream.prototype.request_ulpfec_encoder(sessid: Number): {
    // javascript wrapper for 'gst_rtsp_stream_request_ulpfec_encoder'
}

Creating a rtpulpfecenc element

Since : 1.16

GstRtspServer.RTSPStream.prototype.reserve_address

function GstRtspServer.RTSPStream.prototype.reserve_address(address: String, port: Number, n_ports: Number, ttl: Number): {
    // javascript wrapper for 'gst_rtsp_stream_reserve_address'
}

Reserve address and port as the address and port of stream. The original GstRtspServer.RTSPAddress is cached and copy is returned, so freeing the return value won't release the address from the pool.

GstRtspServer.RTSPStream.prototype.seekable

function GstRtspServer.RTSPStream.prototype.seekable(): {
    // javascript wrapper for 'gst_rtsp_stream_seekable'
}

Checks whether the individual stream is seekable.

Since : 1.14

GstRtspServer.RTSPStream.prototype.set_address_pool

function GstRtspServer.RTSPStream.prototype.set_address_pool(pool: GstRtspServer.RTSPAddressPool): {
    // javascript wrapper for 'gst_rtsp_stream_set_address_pool'
}

configure pool to be used as the address pool of stream.

GstRtspServer.RTSPStream.prototype.set_bind_mcast_address

function GstRtspServer.RTSPStream.prototype.set_bind_mcast_address(bind_mcast_addr: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_bind_mcast_address'
}

Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY.

Since : 1.16

GstRtspServer.RTSPStream.prototype.set_blocked

function GstRtspServer.RTSPStream.prototype.set_blocked(blocked: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_blocked'
}

Blocks or unblocks the dataflow on stream.

GstRtspServer.RTSPStream.prototype.set_buffer_size

function GstRtspServer.RTSPStream.prototype.set_buffer_size(size: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_buffer_size'
}

Set the size of the UDP transmission buffer (in bytes) Needs to be set before the stream is joined to a bin.

Since : 1.6

GstRtspServer.RTSPStream.prototype.set_client_side

function GstRtspServer.RTSPStream.prototype.set_client_side(client_side: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_client_side'
}

Sets the GstRtspServer.RTSPStream as a 'client side' stream - used for sending streams to an RTSP server via RECORD. This has the practical effect of changing which UDP port numbers are used when setting up the local side of the stream sending to be either the 'server' or 'client' pair of a configured UDP transport.

GstRtspServer.RTSPStream.prototype.set_control

function GstRtspServer.RTSPStream.prototype.set_control(control: String): {
    // javascript wrapper for 'gst_rtsp_stream_set_control'
}

Set the control string in stream.

GstRtspServer.RTSPStream.prototype.set_dscp_qos

function GstRtspServer.RTSPStream.prototype.set_dscp_qos(dscp_qos: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_dscp_qos'
}

Configure the dscp qos of the outgoing sockets to dscp_qos.

GstRtspServer.RTSPStream.prototype.set_max_mcast_ttl

function GstRtspServer.RTSPStream.prototype.set_max_mcast_ttl(ttl: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_max_mcast_ttl'
}

Set the maximum time-to-live value of outgoing multicast packets.

Since : 1.16

GstRtspServer.RTSPStream.prototype.set_mtu

function GstRtspServer.RTSPStream.prototype.set_mtu(mtu: Number): {
    // javascript wrapper for 'gst_rtsp_stream_set_mtu'
}

Configure the mtu in the payloader of stream to mtu.

GstRtspServer.RTSPStream.prototype.set_multicast_iface