rtsp media

GstRtspServer.RTSPMedia

GObject.Object
    ╰──GstRtspServer.RTSPMedia
        ╰──GstRtspServer.RTSPOnvifMedia

GstRtspServer.RTSPMedia.prototype.new

function GstRtspServer.RTSPMedia.prototype.new(element: Gst.Element): {
    // javascript wrapper for 'gst_rtsp_media_new'
}

Create a new GstRtspServer.RTSPMedia instance. element is the bin element that provides the different streams. The GstRtspServer.RTSPMedia object contains the element to produce RTP data for one or more related (audio/video/..) streams.

Ownership is taken of element.

GstRtspServer.RTSPMedia.prototype.can_be_shared

function GstRtspServer.RTSPMedia.prototype.can_be_shared(): {
    // javascript wrapper for 'gst_rtsp_media_can_be_shared'
}

Check if the pipeline for media can be shared between multiple clients.

This checks if the media is shareable and whether it is either reusable or was never unprepared before.

The function must be called with GstRtspServer.RTSPMedia.prototype.lock.

Since : 1.24

GstRtspServer.RTSPMedia.prototype.collect_streams

function GstRtspServer.RTSPMedia.prototype.collect_streams(): {
    // javascript wrapper for 'gst_rtsp_media_collect_streams'
}

Find all payloader elements, they should be named pay%d in the element of media, and create GstRtspServer.RTSPStream for them.

Collect all dynamic elements, named dynpay%d, and add them to the list of dynamic elements.

Find all depayloader elements, they should be named depay%d in the element of media, and create GstRtspServer.RTSPStream for them.

GstRtspServer.RTSPMedia.prototype.complete_pipeline

function GstRtspServer.RTSPMedia.prototype.complete_pipeline(transports: [ GstRtsp.RTSPTransport ]): {
    // javascript wrapper for 'gst_rtsp_media_complete_pipeline'
}

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

Since : 1.14

GstRtspServer.RTSPMedia.prototype.create_stream

function GstRtspServer.RTSPMedia.prototype.create_stream(payloader: Gst.Element, pad: Gst.Pad): {
    // javascript wrapper for 'gst_rtsp_media_create_stream'
}

Create a new stream in media that provides RTP data on pad. pad should be a pad of an element inside media->element.

GstRtspServer.RTSPMedia.prototype.find_stream

function GstRtspServer.RTSPMedia.prototype.find_stream(control: String): {
    // javascript wrapper for 'gst_rtsp_media_find_stream'
}

Find a stream in media with control as the control uri.

GstRtspServer.RTSPMedia.prototype.get_address_pool

function GstRtspServer.RTSPMedia.prototype.get_address_pool(): {
    // javascript wrapper for 'gst_rtsp_media_get_address_pool'
}

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

GstRtspServer.RTSPMedia.prototype.get_base_time

function GstRtspServer.RTSPMedia.prototype.get_base_time(): {
    // javascript wrapper for 'gst_rtsp_media_get_base_time'
}

Get the base_time that is used by the pipeline in media.

media must be prepared before this method returns a valid base_time.

GstRtspServer.RTSPMedia.prototype.get_buffer_size

function GstRtspServer.RTSPMedia.prototype.get_buffer_size(): {
    // javascript wrapper for 'gst_rtsp_media_get_buffer_size'
}

Get the kernel UDP buffer size.

GstRtspServer.RTSPMedia.prototype.get_clock

function GstRtspServer.RTSPMedia.prototype.get_clock(): {
    // javascript wrapper for 'gst_rtsp_media_get_clock'
}

Get the clock that is used by the pipeline in media.

media must be prepared before this method returns a valid clock object.

GstRtspServer.RTSPMedia.prototype.get_do_retransmission

function GstRtspServer.RTSPMedia.prototype.get_do_retransmission(): {
    // javascript wrapper for 'gst_rtsp_media_get_do_retransmission'
}

Since : 1.16

GstRtspServer.RTSPMedia.prototype.get_dscp_qos

function GstRtspServer.RTSPMedia.prototype.get_dscp_qos(): {
    // javascript wrapper for 'gst_rtsp_media_get_dscp_qos'
}

Get the configured DSCP QoS of attached media.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.get_element

function GstRtspServer.RTSPMedia.prototype.get_element(): {
    // javascript wrapper for 'gst_rtsp_media_get_element'
}

Get the element that was used when constructing media.

GstRtspServer.RTSPMedia.prototype.get_ensure_keyunit_on_start

function GstRtspServer.RTSPMedia.prototype.get_ensure_keyunit_on_start(): {
    // javascript wrapper for 'gst_rtsp_media_get_ensure_keyunit_on_start'
}

Get ensure-keyunit-on-start flag.

Since : 1.24

GstRtspServer.RTSPMedia.prototype.get_ensure_keyunit_on_start_timeout

function GstRtspServer.RTSPMedia.prototype.get_ensure_keyunit_on_start_timeout(): {
    // javascript wrapper for 'gst_rtsp_media_get_ensure_keyunit_on_start_timeout'
}

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

Since : 1.24

GstRtspServer.RTSPMedia.prototype.get_latency

function GstRtspServer.RTSPMedia.prototype.get_latency(): {
    // javascript wrapper for 'gst_rtsp_media_get_latency'
}

Get the latency that is used for receiving media.

GstRtspServer.RTSPMedia.prototype.get_max_mcast_ttl

function GstRtspServer.RTSPMedia.prototype.get_max_mcast_ttl(): {
    // javascript wrapper for 'gst_rtsp_media_get_max_mcast_ttl'
}

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

Since : 1.16

GstRtspServer.RTSPMedia.prototype.get_multicast_iface

function GstRtspServer.RTSPMedia.prototype.get_multicast_iface(): {
    // javascript wrapper for 'gst_rtsp_media_get_multicast_iface'
}

Get the multicast interface used for media.

GstRtspServer.RTSPMedia.prototype.get_permissions

function GstRtspServer.RTSPMedia.prototype.get_permissions(): {
    // javascript wrapper for 'gst_rtsp_media_get_permissions'
}

Get the permissions object from media.

GstRtspServer.RTSPMedia.prototype.get_profiles

function GstRtspServer.RTSPMedia.prototype.get_profiles(): {
    // javascript wrapper for 'gst_rtsp_media_get_profiles'
}

Get the allowed profiles of media.

GstRtspServer.RTSPMedia.prototype.get_protocols

function GstRtspServer.RTSPMedia.prototype.get_protocols(): {
    // javascript wrapper for 'gst_rtsp_media_get_protocols'
}

Get the allowed protocols of media.

GstRtspServer.RTSPMedia.prototype.get_publish_clock_mode

function GstRtspServer.RTSPMedia.prototype.get_publish_clock_mode(): {
    // javascript wrapper for 'gst_rtsp_media_get_publish_clock_mode'
}

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

Since : 1.8

GstRtspServer.RTSPMedia.prototype.get_range_string

function GstRtspServer.RTSPMedia.prototype.get_range_string(play: Number, unit: GstRtsp.RTSPRangeUnit): {
    // javascript wrapper for 'gst_rtsp_media_get_range_string'
}

Get the current range as a string. media must be prepared with gst_rtsp_media_prepare ().

GstRtspServer.RTSPMedia.prototype.get_rate_control

function GstRtspServer.RTSPMedia.prototype.get_rate_control(): {
    // javascript wrapper for 'gst_rtsp_media_get_rate_control'
}

Since : 1.18

GstRtspServer.RTSPMedia.prototype.get_rates

function GstRtspServer.RTSPMedia.prototype.get_rates(): {
    // javascript wrapper for 'gst_rtsp_media_get_rates'
}

Get the rate and applied_rate of the current segment.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.get_retransmission_time

function GstRtspServer.RTSPMedia.prototype.get_retransmission_time(): {
    // javascript wrapper for 'gst_rtsp_media_get_retransmission_time'
}

Get the amount of time to store retransmission data.

GstRtspServer.RTSPMedia.prototype.get_status

function GstRtspServer.RTSPMedia.prototype.get_status(): {
    // javascript wrapper for 'gst_rtsp_media_get_status'
}

Get the status of media. When media is busy preparing, this function waits until media is prepared or in error.

GstRtspServer.RTSPMedia.prototype.get_stream

function GstRtspServer.RTSPMedia.prototype.get_stream(idx: Number): {
    // javascript wrapper for 'gst_rtsp_media_get_stream'
}

Retrieve the stream with index idx from media.

GstRtspServer.RTSPMedia.prototype.get_suspend_mode

function GstRtspServer.RTSPMedia.prototype.get_suspend_mode(): {
    // javascript wrapper for 'gst_rtsp_media_get_suspend_mode'
}

Get how media will be suspended.

GstRtspServer.RTSPMedia.prototype.get_time_provider

function GstRtspServer.RTSPMedia.prototype.get_time_provider(address: String, port: Number): {
    // javascript wrapper for 'gst_rtsp_media_get_time_provider'
}

Get the GstNet.NetTimeProvider for the clock used by media. The time provider will listen on address and port for client time requests.

GstRtspServer.RTSPMedia.prototype.get_transport_mode

function GstRtspServer.RTSPMedia.prototype.get_transport_mode(): {
    // javascript wrapper for 'gst_rtsp_media_get_transport_mode'
}

Check if the pipeline for media can be used for PLAY or RECORD methods.

GstRtspServer.RTSPMedia.prototype.handle_sdp

function GstRtspServer.RTSPMedia.prototype.handle_sdp(sdp: GstSdp.SDPMessage): {
    // javascript wrapper for 'gst_rtsp_media_handle_sdp'
}

Configure an SDP on media for receiving streams

GstRtspServer.RTSPMedia.prototype.has_completed_sender

function GstRtspServer.RTSPMedia.prototype.has_completed_sender(): {
    // javascript wrapper for 'gst_rtsp_media_has_completed_sender'
}

See GstRtspServer.RTSPStream.prototype.is_complete, GstRtspServer.RTSPStream.prototype.is_sender.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.is_bind_mcast_address

function GstRtspServer.RTSPMedia.prototype.is_bind_mcast_address(): {
    // javascript wrapper for 'gst_rtsp_media_is_bind_mcast_address'
}

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

Since : 1.16

GstRtspServer.RTSPMedia.prototype.is_eos_shutdown

function GstRtspServer.RTSPMedia.prototype.is_eos_shutdown(): {
    // javascript wrapper for 'gst_rtsp_media_is_eos_shutdown'
}

Check if the pipeline for media will send an EOS down the pipeline before unpreparing.

GstRtspServer.RTSPMedia.prototype.is_receive_only

function GstRtspServer.RTSPMedia.prototype.is_receive_only(): {
    // javascript wrapper for 'gst_rtsp_media_is_receive_only'
}

Since : 1.18

GstRtspServer.RTSPMedia.prototype.is_reusable

function GstRtspServer.RTSPMedia.prototype.is_reusable(): {
    // javascript wrapper for 'gst_rtsp_media_is_reusable'
}

Check if the pipeline for media can be reused after an unprepare.

GstRtspServer.RTSPMedia.prototype.is_shared

function GstRtspServer.RTSPMedia.prototype.is_shared(): {
    // javascript wrapper for 'gst_rtsp_media_is_shared'
}

Check if the pipeline for media can be shared between multiple clients in theory. This simply returns the value set via GstRtspServer.RTSPMedia.prototype.set_shared.

To know if a media can be shared in practice, i.e. if it's shareable and either reusable or was never unprepared before, use GstRtspServer.RTSPMedia.prototype.can_be_shared.

GstRtspServer.RTSPMedia.prototype.is_stop_on_disconnect

function GstRtspServer.RTSPMedia.prototype.is_stop_on_disconnect(): {
    // javascript wrapper for 'gst_rtsp_media_is_stop_on_disconnect'
}

Check if the pipeline for media will be stopped when a client disconnects without sending TEARDOWN.

GstRtspServer.RTSPMedia.prototype.is_time_provider

function GstRtspServer.RTSPMedia.prototype.is_time_provider(): {
    // javascript wrapper for 'gst_rtsp_media_is_time_provider'
}

Check if media can provide a GstNet.NetTimeProvider for its pipeline clock.

Use GstRtspServer.RTSPMedia.prototype.get_time_provider to get the network clock.

GstRtspServer.RTSPMedia.prototype.lock

function GstRtspServer.RTSPMedia.prototype.lock(): {
    // javascript wrapper for 'gst_rtsp_media_lock'
}

Lock the entire media. This is needed by callers such as rtsp_client to protect the media when it is shared by many clients. The lock prevents that concurrent clients alters the shared media, while one client already is working with it. Typically the lock is taken in external RTSP API calls that uses shared media such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.

As best practice take the lock as soon as the function get hold of a shared media object. Release the lock right before the function returns.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.n_streams

function GstRtspServer.RTSPMedia.prototype.n_streams(): {
    // javascript wrapper for 'gst_rtsp_media_n_streams'
}

Get the number of streams in this media.

GstRtspServer.RTSPMedia.prototype.prepare

function GstRtspServer.RTSPMedia.prototype.prepare(thread: GstRtspServer.RTSPThread): {
    // javascript wrapper for 'gst_rtsp_media_prepare'
}

Prepare media for streaming. This function will create the objects to manage the streaming. A pipeline must have been set on media with GstRtspServer.RTSPMedia.prototype.take_pipeline.

It will preroll the pipeline and collect vital information about the streams such as the duration.

GstRtspServer.RTSPMedia.prototype.seek

function GstRtspServer.RTSPMedia.prototype.seek(range: GstRtsp.RTSPTimeRange): {
    // javascript wrapper for 'gst_rtsp_media_seek'
}

Seek the pipeline of media to range. media must be prepared with GstRtspServer.RTSPMedia.prototype.prepare.

GstRtspServer.RTSPMedia.prototype.seek_full

function GstRtspServer.RTSPMedia.prototype.seek_full(range: GstRtsp.RTSPTimeRange, flags: Gst.SeekFlags): {
    // javascript wrapper for 'gst_rtsp_media_seek_full'
}

Seek the pipeline of media to range with the given flags. media must be prepared with GstRtspServer.RTSPMedia.prototype.prepare.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.seek_trickmode

function GstRtspServer.RTSPMedia.prototype.seek_trickmode(range: GstRtsp.RTSPTimeRange, flags: Gst.SeekFlags, rate: Number, trickmode_interval: Number): {
    // javascript wrapper for 'gst_rtsp_media_seek_trickmode'
}

Seek the pipeline of media to range with the given flags and rate, and trickmode_interval. media must be prepared with GstRtspServer.RTSPMedia.prototype.prepare. In order to perform the seek operation, the pipeline must contain all needed transport parts (transport sinks).

Since : 1.18

GstRtspServer.RTSPMedia.prototype.seekable

function GstRtspServer.RTSPMedia.prototype.seekable(): {
    // javascript wrapper for 'gst_rtsp_media_seekable'
}

Check if the pipeline for media seek and up to what point in time, it can seek.

Since : 1.14

GstRtspServer.RTSPMedia.prototype.set_address_pool

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

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

GstRtspServer.RTSPMedia.prototype.set_bind_mcast_address

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

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

Since : 1.16

GstRtspServer.RTSPMedia.prototype.set_buffer_size

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

Set the kernel UDP buffer size.

GstRtspServer.RTSPMedia.prototype.set_clock

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

Configure the clock used for the media.

GstRtspServer.RTSPMedia.prototype.set_do_retransmission

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

Set whether retransmission requests will be sent

Since : 1.16

GstRtspServer.RTSPMedia.prototype.set_dscp_qos

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

Configure the dscp qos of attached streams to dscp_qos.

Since : 1.18

GstRtspServer.RTSPMedia.prototype.set_ensure_keyunit_on_start

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

Set whether or not a keyunit should be ensured when a client connects. It will also configure the streams to drop delta units to ensure that they start on a keyunit.

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

Since : 1.24

GstRtspServer.RTSPMedia.prototype.set_ensure_keyunit_on_start_timeout

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

Sets the maximum allowed time before the first keyunit is considered expired.

Note that this will only have an effect when ensure-keyunit-on-start is enabled.

Since : 1.24

GstRtspServer.RTSPMedia.prototype.set_eos_shutdown

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

Set or unset if an EOS event will be sent to the pipeline for media before it is unprepared.

GstRtspServer.RTSPMedia.prototype.set_latency

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

Configure the latency used for receiving media.

GstRtspServer.RTSPMedia.prototype.set_max_mcast_ttl

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

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

Since : 1.16

GstRtspServer.RTSPMedia.prototype.set_multicast_iface

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

configure multicast_iface to be used for media.

GstRtspServer.RTSPMedia.prototype.set_permissions

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

Set permissions on media.

GstRtspServer.RTSPMedia.prototype.set_pipeline_state

function GstRtspServer.RTSPMedia.prototype.set_pipeline_state(state: Gst.State): {
    // javascript wrapper for 'gst_rtsp_media_set_pipeline_state'
}

Set the state of the pipeline managed by media to state

GstRtspServer.RTSPMedia.prototype.set_profiles

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

Configure the allowed lower transport for media.

GstRtspServer.RTSPMedia.prototype.set_protocols

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

Configure the allowed lower transport for media.

GstRtspServer.RTSPMedia.prototype.set_publish_clock_mode

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

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

Since : 1.8

GstRtspServer.RTSPMedia.prototype.set_rate_control

function GstRtspServer.RTSPMedia.prototype.set_rate_control(enabled: Number): {
    // javascript wrapper for 'gst_rtsp_media_set_rate_control'
}

Define whether media will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec.

Since : 1.18