rtsp media factory

GstRtspServer.RTSPMediaFactory

GObject.Object
    ╰──GstRtspServer.RTSPMediaFactory
        ╰──GstRtspServer.RTSPMediaFactoryURI
        ╰──GstRtspServer.RTSPOnvifMediaFactory

GstRtspServer.RTSPMediaFactory.prototype.new

function GstRtspServer.RTSPMediaFactory.prototype.new(): {
    // javascript wrapper for 'gst_rtsp_media_factory_new'
}

Create a new GstRtspServer.RTSPMediaFactory instance.

GstRtspServer.RTSPMediaFactory.prototype.add_role_from_structure

function GstRtspServer.RTSPMediaFactory.prototype.add_role_from_structure(structure: Gst.Structure): {
    // javascript wrapper for 'gst_rtsp_media_factory_add_role_from_structure'
}

A convenience wrapper around GstRtspServer.RTSPPermissions.prototype.add_role_from_structure. If factory had no permissions, new permissions will be created and the role will be added to it.

Since : 1.14

GstRtspServer.RTSPMediaFactory.prototype.construct

function GstRtspServer.RTSPMediaFactory.prototype.construct(url: GstRtsp.RTSPUrl): {
    // javascript wrapper for 'gst_rtsp_media_factory_construct'
}

Construct the media object and create its streams. Implementations should create the needed gstreamer elements and add them to the result object. No state changes should be performed on them yet.

One or more GstRTSPStream objects should be created from the result with gst_rtsp_media_create_stream ().

After the media is constructed, it can be configured and then prepared with gst_rtsp_media_prepare ().

The returned media will be locked and must be unlocked afterwards.

GstRtspServer.RTSPMediaFactory.prototype.create_element

function GstRtspServer.RTSPMediaFactory.prototype.create_element(url: GstRtsp.RTSPUrl): {
    // javascript wrapper for 'gst_rtsp_media_factory_create_element'
}

Construct and return a Gst.Element that is a Gst.Bin containing the elements to use for streaming the media.

The bin should contain payloaders pay%d for each stream. The default implementation of this function returns the bin created from the launch parameter.

GstRtspServer.RTSPMediaFactory.prototype.get_address_pool

function GstRtspServer.RTSPMediaFactory.prototype.get_address_pool(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_address_pool'
}

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

GstRtspServer.RTSPMediaFactory.prototype.get_buffer_size

function GstRtspServer.RTSPMediaFactory.prototype.get_buffer_size(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_buffer_size'
}

Get the kernel UDP buffer size.

GstRtspServer.RTSPMediaFactory.prototype.get_clock

function GstRtspServer.RTSPMediaFactory.prototype.get_clock(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_clock'
}

Returns the clock that is going to be used by the pipelines of all medias created from this factory.

Since : 1.8

GstRtspServer.RTSPMediaFactory.prototype.get_do_retransmission

function GstRtspServer.RTSPMediaFactory.prototype.get_do_retransmission(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_do_retransmission'
}

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.get_dscp_qos

function GstRtspServer.RTSPMediaFactory.prototype.get_dscp_qos(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_dscp_qos'
}

Get the configured media DSCP QoS.

Since : 1.18

GstRtspServer.RTSPMediaFactory.prototype.get_ensure_keyunit_on_start

function GstRtspServer.RTSPMediaFactory.prototype.get_ensure_keyunit_on_start(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_ensure_keyunit_on_start'
}

Get ensure-keyunit-on-start flag.

Since : 1.24

GstRtspServer.RTSPMediaFactory.prototype.get_ensure_keyunit_on_start_timeout

function GstRtspServer.RTSPMediaFactory.prototype.get_ensure_keyunit_on_start_timeout(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_ensure_keyunit_on_start_timeout'
}

Get ensure-keyunit-on-start-timeout time.

Since : 1.24

GstRtspServer.RTSPMediaFactory.prototype.get_latency

function GstRtspServer.RTSPMediaFactory.prototype.get_latency(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_latency'
}

Get the latency that is used for receiving media

GstRtspServer.RTSPMediaFactory.prototype.get_launch

function GstRtspServer.RTSPMediaFactory.prototype.get_launch(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_launch'
}

Get the Gst.prototype.parse_launch pipeline description that will be used in the default prepare vmethod.

GstRtspServer.RTSPMediaFactory.prototype.get_max_mcast_ttl

function GstRtspServer.RTSPMediaFactory.prototype.get_max_mcast_ttl(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_max_mcast_ttl'
}

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

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.get_media_gtype

function GstRtspServer.RTSPMediaFactory.prototype.get_media_gtype(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_media_gtype'
}

Return the GType of the GstRTSPMedia subclass this factory will create.

Since : 1.6

GstRtspServer.RTSPMediaFactory.prototype.get_multicast_iface

function GstRtspServer.RTSPMediaFactory.prototype.get_multicast_iface(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_multicast_iface'
}

Get the multicast interface used for factory.

GstRtspServer.RTSPMediaFactory.prototype.get_permissions

function GstRtspServer.RTSPMediaFactory.prototype.get_permissions(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_permissions'
}

Get the permissions object from factory.

GstRtspServer.RTSPMediaFactory.prototype.get_profiles

function GstRtspServer.RTSPMediaFactory.prototype.get_profiles(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_profiles'
}

Get the allowed profiles of factory.

GstRtspServer.RTSPMediaFactory.prototype.get_protocols

function GstRtspServer.RTSPMediaFactory.prototype.get_protocols(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_protocols'
}

Get the allowed protocols of factory.

GstRtspServer.RTSPMediaFactory.prototype.get_publish_clock_mode

function GstRtspServer.RTSPMediaFactory.prototype.get_publish_clock_mode(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_publish_clock_mode'
}

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

Since : 1.8

GstRtspServer.RTSPMediaFactory.prototype.get_retransmission_time

function GstRtspServer.RTSPMediaFactory.prototype.get_retransmission_time(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_retransmission_time'
}

Get the time that is stored for retransmission purposes

GstRtspServer.RTSPMediaFactory.prototype.get_suspend_mode

function GstRtspServer.RTSPMediaFactory.prototype.get_suspend_mode(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_suspend_mode'
}

Get how media created from this factory will be suspended.

GstRtspServer.RTSPMediaFactory.prototype.get_transport_mode

function GstRtspServer.RTSPMediaFactory.prototype.get_transport_mode(): {
    // javascript wrapper for 'gst_rtsp_media_factory_get_transport_mode'
}

Get if media created from this factory can be used for PLAY or RECORD methods.

GstRtspServer.RTSPMediaFactory.prototype.is_bind_mcast_address

function GstRtspServer.RTSPMediaFactory.prototype.is_bind_mcast_address(): {
    // javascript wrapper for 'gst_rtsp_media_factory_is_bind_mcast_address'
}

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

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.is_enable_rtcp

function GstRtspServer.RTSPMediaFactory.prototype.is_enable_rtcp(): {
    // javascript wrapper for 'gst_rtsp_media_factory_is_enable_rtcp'
}

Check if created media will send and receive RTCP

Since : 1.20

GstRtspServer.RTSPMediaFactory.prototype.is_eos_shutdown

function GstRtspServer.RTSPMediaFactory.prototype.is_eos_shutdown(): {
    // javascript wrapper for 'gst_rtsp_media_factory_is_eos_shutdown'
}

Get if media created from this factory will have an EOS event sent to the pipeline before shutdown.

GstRtspServer.RTSPMediaFactory.prototype.is_shared

function GstRtspServer.RTSPMediaFactory.prototype.is_shared(): {
    // javascript wrapper for 'gst_rtsp_media_factory_is_shared'
}

Get if media created from this factory can be shared between clients.

GstRtspServer.RTSPMediaFactory.prototype.is_stop_on_disonnect

function GstRtspServer.RTSPMediaFactory.prototype.is_stop_on_disonnect(): {
    // javascript wrapper for 'gst_rtsp_media_factory_is_stop_on_disonnect'
}

GstRtspServer.RTSPMediaFactory.prototype.set_address_pool

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

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

GstRtspServer.RTSPMediaFactory.prototype.set_bind_mcast_address

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

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

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.set_buffer_size

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

Set the kernel UDP buffer size.

GstRtspServer.RTSPMediaFactory.prototype.set_clock

function GstRtspServer.RTSPMediaFactory.prototype.set_clock(clock: Gst.Clock): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_clock'
}

Configures a specific clock to be used by the pipelines of all medias created from this factory.

Since : 1.8

GstRtspServer.RTSPMediaFactory.prototype.set_do_retransmission

function GstRtspServer.RTSPMediaFactory.prototype.set_do_retransmission(do_retransmission: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_do_retransmission'
}

Set whether retransmission requests will be sent for receiving media

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.set_dscp_qos

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

Configure the media dscp qos to dscp_qos.

Since : 1.18

GstRtspServer.RTSPMediaFactory.prototype.set_enable_rtcp

function GstRtspServer.RTSPMediaFactory.prototype.set_enable_rtcp(enable: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_enable_rtcp'
}

Decide whether the created media should send and receive RTCP

Since : 1.20

GstRtspServer.RTSPMediaFactory.prototype.set_ensure_keyunit_on_start

function GstRtspServer.RTSPMediaFactory.prototype.set_ensure_keyunit_on_start(ensure_keyunit_on_start: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_ensure_keyunit_on_start'
}

If media from this factory should ensure a key unit when a client connects.

Since : 1.24

GstRtspServer.RTSPMediaFactory.prototype.set_ensure_keyunit_on_start_timeout

function GstRtspServer.RTSPMediaFactory.prototype.set_ensure_keyunit_on_start_timeout(timeout: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_ensure_keyunit_on_start_timeout'
}

Configures medias from this factory to consider keyunits older than timeout to be expired. Expired keyunits will be discarded.

Since : 1.24

GstRtspServer.RTSPMediaFactory.prototype.set_eos_shutdown

function GstRtspServer.RTSPMediaFactory.prototype.set_eos_shutdown(eos_shutdown: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_eos_shutdown'
}

Configure if media created from this factory will have an EOS sent to the pipeline before shutdown.

GstRtspServer.RTSPMediaFactory.prototype.set_latency

function GstRtspServer.RTSPMediaFactory.prototype.set_latency(latency: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_latency'
}

Configure the latency used for receiving media

GstRtspServer.RTSPMediaFactory.prototype.set_launch

function GstRtspServer.RTSPMediaFactory.prototype.set_launch(launch: String): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_launch'
}

The Gst.prototype.parse_launch line to use for constructing the pipeline in the default prepare vmethod.

The pipeline description should return a GstBin as the toplevel element which can be accomplished by enclosing the description with brackets '(' ')'.

The description should return a pipeline with payloaders named pay0, pay1, etc.. Each of the payloaders will result in a stream.

GstRtspServer.RTSPMediaFactory.prototype.set_max_mcast_ttl

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

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

Since : 1.16

GstRtspServer.RTSPMediaFactory.prototype.set_media_gtype

function GstRtspServer.RTSPMediaFactory.prototype.set_media_gtype(media_gtype: GObject.Type): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_media_gtype'
}

Configure the GType of the GstRTSPMedia subclass to create (by default, overridden construct vmethods may of course do something different)

Since : 1.6

GstRtspServer.RTSPMediaFactory.prototype.set_multicast_iface

function GstRtspServer.RTSPMediaFactory.prototype.set_multicast_iface(multicast_iface: String): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_multicast_iface'
}

configure multicast_iface to be used for factory.

GstRtspServer.RTSPMediaFactory.prototype.set_permissions

function GstRtspServer.RTSPMediaFactory.prototype.set_permissions(permissions: GstRtspServer.RTSPPermissions): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_permissions'
}

Set permissions on factory.

GstRtspServer.RTSPMediaFactory.prototype.set_profiles

function GstRtspServer.RTSPMediaFactory.prototype.set_profiles(profiles: GstRtsp.RTSPProfile): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_profiles'
}

Configure the allowed profiles for factory.

GstRtspServer.RTSPMediaFactory.prototype.set_protocols

function GstRtspServer.RTSPMediaFactory.prototype.set_protocols(protocols: GstRtsp.RTSPLowerTrans): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_protocols'
}

Configure the allowed lower transport for factory.

GstRtspServer.RTSPMediaFactory.prototype.set_publish_clock_mode

function GstRtspServer.RTSPMediaFactory.prototype.set_publish_clock_mode(mode: GstRtspServer.RTSPPublishClockMode): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_publish_clock_mode'
}

Sets if and how the media clock should be published according to RFC7273.

Since : 1.8

GstRtspServer.RTSPMediaFactory.prototype.set_retransmission_time

function GstRtspServer.RTSPMediaFactory.prototype.set_retransmission_time(time: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_retransmission_time'
}

Configure the time to store for possible retransmission

GstRtspServer.RTSPMediaFactory.prototype.set_shared

function GstRtspServer.RTSPMediaFactory.prototype.set_shared(shared: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_shared'
}

Configure if media created from this factory can be shared between clients.

GstRtspServer.RTSPMediaFactory.prototype.set_stop_on_disconnect

function GstRtspServer.RTSPMediaFactory.prototype.set_stop_on_disconnect(stop_on_disconnect: Number): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_stop_on_disconnect'
}

Configure if media created from this factory should be stopped when a client disconnects without sending TEARDOWN.

GstRtspServer.RTSPMediaFactory.prototype.set_suspend_mode

function GstRtspServer.RTSPMediaFactory.prototype.set_suspend_mode(mode: GstRtspServer.RTSPSuspendMode): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_suspend_mode'
}

Configure how media created from this factory will be suspended.

GstRtspServer.RTSPMediaFactory.prototype.set_transport_mode

function GstRtspServer.RTSPMediaFactory.prototype.set_transport_mode(mode: GstRtspServer.RTSPTransportMode): {
    // javascript wrapper for 'gst_rtsp_media_factory_set_transport_mode'
}

Configure if this factory creates media for PLAY or RECORD modes.

media-configure

function media_configure_callback(self: GstRtspServer.RTSPMediaFactory, object: GstRtspServer.RTSPMedia, user_data: Object): {
    // javascript callback for the 'media-configure' signal
}

Flags: Run Last

media-constructed

function media_constructed_callback(self: GstRtspServer.RTSPMediaFactory, object: GstRtspServer.RTSPMedia, user_data: Object): {
    // javascript callback for the 'media-constructed' signal
}

Flags: Run Last

bind-mcast-address

“bind-mcast-address” Number

Flags : Read / Write

buffer-size

“buffer-size” Number

Flags : Read / Write

clock

“clock” Gst.Clock

Flags : Read / Write

dscp-qos

“dscp-qos” Number

Flags : Read / Write

enable-rtcp

“enable-rtcp” Number

Whether the created media should send and receive RTCP

Flags : Read / Write

Since : 1.20

ensure-keyunit-on-start

“ensure-keyunit-on-start” Number

If media from this factory should ensure a key unit when a client connects.

This property will ensure that the stream always starts on a key unit instead of a delta unit which the client would not be able to decode.

Note that this will only affect non-shared medias for now.

Flags : Read / Write

Since : 1.24

ensure-keyunit-on-start-timeout

“ensure-keyunit-on-start-timeout” Number

Timeout in milliseconds used to determine if a keyunit should be discarded when a client connects.

If the timeout has been reached a new keyframe will be forced, otherwise the currently blocking keyframe will be used.

This options is only relevant when ensure-keyunit-on-start is enabled.

Flags : Read / Write

Since : 1.24

eos-shutdown

“eos-shutdown” Number

Flags : Read / Write

latency

“latency” Number

Flags : Read / Write

launch

“launch” String

Flags : Read / Write

max-mcast-ttl

“max-mcast-ttl” Number

Flags : Read / Write

profiles

“profiles” GstRtsp.RTSPProfile

Flags : Read / Write

protocols

“protocols” GstRtsp.RTSPLowerTrans

Flags : Read / Write

shared

“shared” Number

Flags : Read / Write

stop-on-disconnect

“stop-on-disconnect” Number

Flags : Read / Write

suspend-mode

“suspend-mode” GstRtspServer.RTSPSuspendMode

Flags : Read / Write

transport-mode

“transport-mode” GstRtspServer.RTSPTransportMode

Flags : Read / Write

vfunc_configure

function vfunc_configure(factory: GstRtspServer.RTSPMediaFactory, media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'configure' virtual method
}

configure the media created with construct. The default implementation will configure the 'shared' property of the media.

vfunc_construct

function vfunc_construct(factory: GstRtspServer.RTSPMediaFactory, url: GstRtsp.RTSPUrl): {
    // javascript implementation of the 'construct' virtual method
}

the vmethod that will be called when the factory has to create the GstRtspServer.RTSPMedia for url. The default implementation of this function calls create_element to retrieve an element and then looks for pay%d to create the streams.

vfunc_create_element

function vfunc_create_element(factory: GstRtspServer.RTSPMediaFactory, url: GstRtsp.RTSPUrl): {
    // javascript implementation of the 'create_element' virtual method
}

Construct and return a Gst.Element that is a Gst.Bin containing the elements to use for streaming the media. The bin should contain payloaders pay%d for each stream. The default implementation of this function returns the bin created from the launch parameter.

vfunc_gen_key

function vfunc_gen_key(factory: GstRtspServer.RTSPMediaFactory, url: GstRtsp.RTSPUrl): {
    // javascript implementation of the 'gen_key' virtual method
}

convert url to a key for caching shared GstRtspServer.RTSPMedia objects. The default implementation of this function will use the complete URL including the query parameters to return a key.

vfunc_media_configure

function vfunc_media_configure(factory: GstRtspServer.RTSPMediaFactory, media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'media_configure' virtual method
}

signal emitted when a media should be configured

vfunc_media_constructed

function vfunc_media_constructed(factory: GstRtspServer.RTSPMediaFactory, media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'media_constructed' virtual method
}

signal emitted when a media was constructed