rtsp media

a GstRTSPMedia contains the complete GStreamer pipeline to manage the streaming to the clients. The actual data transfer is done by the GstRTSPStream objects that are created and exposed by the GstRTSPMedia.

The GstRTSPMedia is usually created from a GstRTSPMediaFactory when the client does a DESCRIBE or SETUP of a resource.

A media is created with gst_rtsp_media_new that takes the element that will provide the streaming elements. For each of the streams, a new GstRTSPStream object needs to be made with the gst_rtsp_media_create_stream which takes the payloader element and the source pad that produces the RTP stream.

The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare. The prepare method will add rtpbin and sinks and sources to send and receive RTP and RTCP packets from the clients. Each stream srcpad is connected to an input into the internal rtpbin.

It is also possible to dynamically create GstRTSPStream objects during the prepare phase. With gst_rtsp_media_get_status you can check the status of the prepare phase.

After the media is prepared, it is ready for streaming. It will usually be managed in a session with gst_rtsp_session_manage_media. See GstRTSPSession and GstRTSPSessionMedia.

The state of the media can be controlled with gst_rtsp_media_set_state (). Seeking can be done with gst_rtsp_media_seek, or gst_rtsp_media_seek_full or gst_rtsp_media_seek_trickmode for finer control of the seek.

With gst_rtsp_media_unprepare the pipeline is stopped and shut down. When gst_rtsp_media_set_eos_shutdown an EOS will be sent to the pipeline to cleanly shut down.

With gst_rtsp_media_set_shared, the media can be shared between multiple clients. With gst_rtsp_media_set_reusable you can control if the pipeline can be prepared again after an unprepare.

Last reviewed on 2013-07-11 (1.0.0)

GstRTSPMedia

GObject
    ╰──GstRTSPMedia
        ╰──GstRTSPOnvifMedia

A class that contains the GStreamer element along with a list of GstRTSPStream objects that can produce data.

This object is usually created from a GstRTSPMediaFactory.

Members

parent (GObject) –
No description available

Class structure

GstRTSPMediaClass

The RTSP media class

Fields
parent_class (GObjectClass) –
No description available

GstRtspServer.RTSPMediaClass

The RTSP media class

Attributes
parent_class (GObject.ObjectClass) –
No description available

GstRtspServer.RTSPMediaClass

The RTSP media class

Attributes
parent_class (GObject.ObjectClass) –
No description available

GstRtspServer.RTSPMedia

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

A class that contains the GStreamer element along with a list of GstRtspServer.RTSPStream objects that can produce data.

This object is usually created from a GstRtspServer.RTSPMediaFactory.

Members

parent (GObject.Object) –
No description available

GstRtspServer.RTSPMedia

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

A class that contains the GStreamer element along with a list of GstRtspServer.RTSPStream objects that can produce data.

This object is usually created from a GstRtspServer.RTSPMediaFactory.

Members

parent (GObject.Object) –
No description available

Constructors

gst_rtsp_media_new

GstRTSPMedia *
gst_rtsp_media_new (GstElement * element)

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

Ownership is taken of element.

Parameters:

element ( [transfer: full])

a GstElement

Returns ( [transfer: full])

a new GstRTSPMedia object.


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.

Parameters:

element (Gst.Element)

a Gst.Element

Returns (GstRtspServer.RTSPMedia)

a new GstRtspServer.RTSPMedia object.


GstRtspServer.RTSPMedia.new

def GstRtspServer.RTSPMedia.new (element):
    #python 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.

Parameters:

element (Gst.Element)

a Gst.Element

Returns (GstRtspServer.RTSPMedia)

a new GstRtspServer.RTSPMedia object.


Methods

gst_rtsp_media_collect_streams

gst_rtsp_media_collect_streams (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia


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.collect_streams

def GstRtspServer.RTSPMedia.collect_streams (self):
    #python 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.


gst_rtsp_media_complete_pipeline

gboolean
gst_rtsp_media_complete_pipeline (GstRTSPMedia * media,
                                  GPtrArray * transports)

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

Parameters:

media

a GstRTSPMedia

transports ( [element-typeGstRTSPTransport])

a list of GstRTSPTransport

Returns

TRUE if the media pipeline has been sucessfully updated.

Since : 1.14


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.

Parameters:

transports ([ GstRtsp.RTSPTransport ])

a list of GstRtsp.RTSPTransport

Returns (Number)

true if the media pipeline has been sucessfully updated.

Since : 1.14


GstRtspServer.RTSPMedia.complete_pipeline

def GstRtspServer.RTSPMedia.complete_pipeline (self, transports):
    #python wrapper for 'gst_rtsp_media_complete_pipeline'

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

Parameters:

transports ([ GstRtsp.RTSPTransport ])

a list of GstRtsp.RTSPTransport

Returns (bool)

True if the media pipeline has been sucessfully updated.

Since : 1.14


gst_rtsp_media_create_stream

GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media,
                              GstElement * payloader,
                              GstPad * pad)

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

Parameters:

media

a GstRTSPMedia

payloader

a GstElement

pad

a GstPad

Returns ( [transfer: none])

a new GstRTSPStream that remains valid for as long as media exists.


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.

Parameters:

payloader (Gst.Element)

a Gst.Element

pad (Gst.Pad)

a Gst.Pad

Returns (GstRtspServer.RTSPStream)

a new GstRtspServer.RTSPStream that remains valid for as long as media exists.


GstRtspServer.RTSPMedia.create_stream

def GstRtspServer.RTSPMedia.create_stream (self, payloader, pad):
    #python 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.

Parameters:

payloader (Gst.Element)

a Gst.Element

pad (Gst.Pad)

a Gst.Pad

Returns (GstRtspServer.RTSPStream)

a new GstRtspServer.RTSPStream that remains valid for as long as media exists.


gst_rtsp_media_find_stream

GstRTSPStream *
gst_rtsp_media_find_stream (GstRTSPMedia * media,
                            const gchar * control)

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

Parameters:

media

a GstRTSPMedia

control

the control of the stream

Returns ( [nullable][transfer: none])

the GstRTSPStream with control uri control or NULL when a stream with that control did not exist.


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.

Parameters:

control (String)

the control of the stream

Returns (GstRtspServer.RTSPStream)

the GstRtspServer.RTSPStream with control uri control or null when a stream with that control did not exist.


GstRtspServer.RTSPMedia.find_stream

def GstRtspServer.RTSPMedia.find_stream (self, control):
    #python wrapper for 'gst_rtsp_media_find_stream'

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

Parameters:

control (str)

the control of the stream

Returns (GstRtspServer.RTSPStream)

the GstRtspServer.RTSPStream with control uri control or None when a stream with that control did not exist.


gst_rtsp_media_get_address_pool

GstRTSPAddressPool *
gst_rtsp_media_get_address_pool (GstRTSPMedia * media)

Get the GstRTSPAddressPool used as the address pool of media.

Parameters:

media

a GstRTSPMedia

Returns ( [transfer: full][nullable])

the GstRTSPAddressPool of media. g_object_unref after usage.


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.get_address_pool

def GstRtspServer.RTSPMedia.get_address_pool (self):
    #python wrapper for 'gst_rtsp_media_get_address_pool'

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


gst_rtsp_media_get_base_time

GstClockTime
gst_rtsp_media_get_base_time (GstRTSPMedia * media)

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

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

Parameters:

media

a GstRTSPMedia

Returns

the base_time used by 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.

Returns (Number)

the base_time used by media.


GstRtspServer.RTSPMedia.get_base_time

def GstRtspServer.RTSPMedia.get_base_time (self):
    #python 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.

Returns (int)

the base_time used by media.


gst_rtsp_media_get_buffer_size

guint
gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)

Get the kernel UDP buffer size.

Parameters:

media

a GstRTSPMedia

Returns

the kernel UDP buffer size.


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.

Returns (Number)

the kernel UDP buffer size.


GstRtspServer.RTSPMedia.get_buffer_size

def GstRtspServer.RTSPMedia.get_buffer_size (self):
    #python wrapper for 'gst_rtsp_media_get_buffer_size'

Get the kernel UDP buffer size.

Returns (int)

the kernel UDP buffer size.


gst_rtsp_media_get_clock

GstClock *
gst_rtsp_media_get_clock (GstRTSPMedia * media)

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

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

Parameters:

media

a GstRTSPMedia

Returns ( [transfer: full][nullable])

the GstClock used by media. unref after usage.


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.

Returns (Gst.Clock)

the Gst.Clock used by media. unref after usage.


GstRtspServer.RTSPMedia.get_clock

def GstRtspServer.RTSPMedia.get_clock (self):
    #python 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.

Returns (Gst.Clock)

the Gst.Clock used by media. unref after usage.


gst_rtsp_media_get_do_retransmission

gboolean
gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)

Parameters:

media
No description available
Returns

Whether retransmission requests will be sent

Since : 1.16


GstRtspServer.RTSPMedia.prototype.get_do_retransmission

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

Parameters:

No description available
Returns (Number)

Whether retransmission requests will be sent

Since : 1.16


GstRtspServer.RTSPMedia.get_do_retransmission

def GstRtspServer.RTSPMedia.get_do_retransmission (self):
    #python wrapper for 'gst_rtsp_media_get_do_retransmission'

Parameters:

No description available
Returns (bool)

Whether retransmission requests will be sent

Since : 1.16


gst_rtsp_media_get_dscp_qos

gint
gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media)

Get the configured DSCP QoS of attached media.

Parameters:

media

a GstRTSPMedia

Returns

the DSCP QoS value of attached streams or -1 if disabled.

Since : 1.18


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.

Returns (Number)

the DSCP QoS value of attached streams or -1 if disabled.

Since : 1.18


GstRtspServer.RTSPMedia.get_dscp_qos

def GstRtspServer.RTSPMedia.get_dscp_qos (self):
    #python wrapper for 'gst_rtsp_media_get_dscp_qos'

Get the configured DSCP QoS of attached media.

Returns (int)

the DSCP QoS value of attached streams or -1 if disabled.

Since : 1.18


gst_rtsp_media_get_element

GstElement *
gst_rtsp_media_get_element (GstRTSPMedia * media)

Get the element that was used when constructing media.

Parameters:

media

a GstRTSPMedia

Returns ( [transfer: full])

a GstElement. Unref after usage.


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.

Returns (Gst.Element)

a Gst.Element. Unref after usage.


GstRtspServer.RTSPMedia.get_element

def GstRtspServer.RTSPMedia.get_element (self):
    #python wrapper for 'gst_rtsp_media_get_element'

Get the element that was used when constructing media.

Returns (Gst.Element)

a Gst.Element. Unref after usage.


gst_rtsp_media_get_latency

guint
gst_rtsp_media_get_latency (GstRTSPMedia * media)

Get the latency that is used for receiving media.

Parameters:

media

a GstRTSPMedia

Returns

latency in milliseconds


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.

Returns (Number)

latency in milliseconds


GstRtspServer.RTSPMedia.get_latency

def GstRtspServer.RTSPMedia.get_latency (self):
    #python wrapper for 'gst_rtsp_media_get_latency'

Get the latency that is used for receiving media.

Returns (int)

latency in milliseconds


gst_rtsp_media_get_max_mcast_ttl

guint
gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

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

Since : 1.16


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.

Returns (Number)

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

Since : 1.16


GstRtspServer.RTSPMedia.get_max_mcast_ttl

def GstRtspServer.RTSPMedia.get_max_mcast_ttl (self):
    #python wrapper for 'gst_rtsp_media_get_max_mcast_ttl'

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

Returns (int)

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

Since : 1.16


gst_rtsp_media_get_multicast_iface

gchar *
gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)

Get the multicast interface used for media.

Parameters:

media

a GstRTSPMedia

Returns ( [transfer: full][nullable])

the multicast interface for media. g_free after usage.


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.

Returns (String)

the multicast interface for media. GLib.prototype.free after usage.


GstRtspServer.RTSPMedia.get_multicast_iface

def GstRtspServer.RTSPMedia.get_multicast_iface (self):
    #python wrapper for 'gst_rtsp_media_get_multicast_iface'

Get the multicast interface used for media.

Returns (str)

the multicast interface for media. GLib.free after usage.


gst_rtsp_media_get_permissions

GstRTSPPermissions *
gst_rtsp_media_get_permissions (GstRTSPMedia * media)

Get the permissions object from media.

Parameters:

media

a GstRTSPMedia

Returns ( [transfer: full][nullable])

a GstRTSPPermissions object, unref after usage.


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.

a GstRtspServer.RTSPPermissions object, unref after usage.


GstRtspServer.RTSPMedia.get_permissions

def GstRtspServer.RTSPMedia.get_permissions (self):
    #python wrapper for 'gst_rtsp_media_get_permissions'

Get the permissions object from media.

a GstRtspServer.RTSPPermissions object, unref after usage.


gst_rtsp_media_get_profiles

GstRTSPProfile
gst_rtsp_media_get_profiles (GstRTSPMedia * media)

Get the allowed profiles of media.

Parameters:

media

a GstRTSPMedia

Returns

a GstRTSPProfile


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.get_profiles

def GstRtspServer.RTSPMedia.get_profiles (self):
    #python wrapper for 'gst_rtsp_media_get_profiles'

Get the allowed profiles of media.


gst_rtsp_media_get_protocols

GstRTSPLowerTrans
gst_rtsp_media_get_protocols (GstRTSPMedia * media)

Get the allowed protocols of media.

Parameters:

media

a GstRTSPMedia

Returns

a GstRTSPLowerTrans


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.get_protocols

def GstRtspServer.RTSPMedia.get_protocols (self):
    #python wrapper for 'gst_rtsp_media_get_protocols'

Get the allowed protocols of media.


gst_rtsp_media_get_publish_clock_mode

GstRTSPPublishClockMode
gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

The GstRTSPPublishClockMode

Since : 1.8


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.

The GstRTSPPublishClockMode

Since : 1.8


GstRtspServer.RTSPMedia.get_publish_clock_mode

def GstRtspServer.RTSPMedia.get_publish_clock_mode (self):
    #python wrapper for 'gst_rtsp_media_get_publish_clock_mode'

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

The GstRTSPPublishClockMode

Since : 1.8


gst_rtsp_media_get_range_string

gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media,
                                 gboolean play,
                                 GstRTSPRangeUnit unit)

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

Parameters:

media

a GstRTSPMedia

play

for the PLAY request

unit

the unit to use for the string

Returns ( [transfer: full][nullable])

The range as a string, g_free after usage.


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 ().

Parameters:

play (Number)

for the PLAY request

the unit to use for the string

Returns (String)

The range as a string, GLib.prototype.free after usage.


GstRtspServer.RTSPMedia.get_range_string

def GstRtspServer.RTSPMedia.get_range_string (self, play, unit):
    #python wrapper for 'gst_rtsp_media_get_range_string'

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

Parameters:

play (bool)

for the PLAY request

the unit to use for the string

Returns (str)

The range as a string, GLib.free after usage.


gst_rtsp_media_get_rate_control

gboolean
gst_rtsp_media_get_rate_control (GstRTSPMedia * media)

Parameters:

media
No description available
Returns

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

Since : 1.18


GstRtspServer.RTSPMedia.prototype.get_rate_control

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

Parameters:

No description available
Returns (Number)

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

Since : 1.18


GstRtspServer.RTSPMedia.get_rate_control

def GstRtspServer.RTSPMedia.get_rate_control (self):
    #python wrapper for 'gst_rtsp_media_get_rate_control'

Parameters:

No description available
Returns (bool)

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

Since : 1.18


gst_rtsp_media_get_rates

gboolean
gst_rtsp_media_get_rates (GstRTSPMedia * media,
                          gdouble * rate,
                          gdouble * applied_rate)

Get the rate and applied_rate of the current segment.

Parameters:

media

a GstRTSPMedia

rate ( [optional][out])

the rate of the current segment

applied_rate ( [optional][out])

the applied_rate of the current segment

Returns

FALSE if looking up the rate and applied rate failed. Otherwise TRUE is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

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.

Returns a tuple made of:

(Number )

false if looking up the rate and applied rate failed. Otherwise true is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

rate (Number )

false if looking up the rate and applied rate failed. Otherwise true is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

applied_rate (Number )

false if looking up the rate and applied rate failed. Otherwise true is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

Since : 1.18


GstRtspServer.RTSPMedia.get_rates

def GstRtspServer.RTSPMedia.get_rates (self):
    #python wrapper for 'gst_rtsp_media_get_rates'

Get the rate and applied_rate of the current segment.

Returns a tuple made of:

(bool )

False if looking up the rate and applied rate failed. Otherwise True is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

rate (float )

False if looking up the rate and applied rate failed. Otherwise True is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

applied_rate (float )

False if looking up the rate and applied rate failed. Otherwise True is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

Since : 1.18


gst_rtsp_media_get_retransmission_time

GstClockTime
gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)

Get the amount of time to store retransmission data.

Parameters:

media

a GstRTSPMedia

Returns

the amount of time to store retransmission data.


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.

Returns (Number)

the amount of time to store retransmission data.


GstRtspServer.RTSPMedia.get_retransmission_time

def GstRtspServer.RTSPMedia.get_retransmission_time (self):
    #python wrapper for 'gst_rtsp_media_get_retransmission_time'

Get the amount of time to store retransmission data.

Returns (int)

the amount of time to store retransmission data.


gst_rtsp_media_get_status

GstRTSPMediaStatus
gst_rtsp_media_get_status (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

the status of media.


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.

the status of media.


GstRtspServer.RTSPMedia.get_status

def GstRtspServer.RTSPMedia.get_status (self):
    #python 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.

the status of media.


gst_rtsp_media_get_stream

GstRTSPStream *
gst_rtsp_media_get_stream (GstRTSPMedia * media,
                           guint idx)

Retrieve the stream with index idx from media.

Parameters:

media

a GstRTSPMedia

idx

the stream index

Returns ( [nullable][transfer: none])

the GstRTSPStream at index idx or NULL when a stream with that index did not exist.


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.

Parameters:

idx (Number)

the stream index

Returns (GstRtspServer.RTSPStream)

the GstRtspServer.RTSPStream at index idx or null when a stream with that index did not exist.


GstRtspServer.RTSPMedia.get_stream

def GstRtspServer.RTSPMedia.get_stream (self, idx):
    #python wrapper for 'gst_rtsp_media_get_stream'

Retrieve the stream with index idx from media.

Parameters:

idx (int)

the stream index

Returns (GstRtspServer.RTSPStream)

the GstRtspServer.RTSPStream at index idx or None when a stream with that index did not exist.


gst_rtsp_media_get_suspend_mode

GstRTSPSuspendMode
gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)

Get how media will be suspended.

Parameters:

media

a GstRTSPMedia

Returns

GstRTSPSuspendMode.


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.get_suspend_mode

def GstRtspServer.RTSPMedia.get_suspend_mode (self):
    #python wrapper for 'gst_rtsp_media_get_suspend_mode'

Get how media will be suspended.


gst_rtsp_media_get_time_provider

GstNetTimeProvider *
gst_rtsp_media_get_time_provider (GstRTSPMedia * media,
                                  const gchar * address,
                                  guint16 port)

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

Parameters:

media

a GstRTSPMedia

address ( [allow-none])

an address or NULL

port

a port or 0

Returns ( [transfer: full])

the GstNetTimeProvider of media.


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.

Parameters:

address (String)

an address or null

port (Number)

a port or 0

Returns (GstNet.NetTimeProvider)

the GstNet.NetTimeProvider of media.


GstRtspServer.RTSPMedia.get_time_provider

def GstRtspServer.RTSPMedia.get_time_provider (self, address, port):
    #python 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.

Parameters:

address (str)

an address or None

port (int)

a port or 0

Returns (GstNet.NetTimeProvider)

the GstNet.NetTimeProvider of media.


gst_rtsp_media_get_transport_mode

GstRTSPTransportMode
gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

The transport mode.


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.

The transport mode.


GstRtspServer.RTSPMedia.get_transport_mode

def GstRtspServer.RTSPMedia.get_transport_mode (self):
    #python wrapper for 'gst_rtsp_media_get_transport_mode'

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

The transport mode.


gst_rtsp_media_handle_sdp

gboolean
gst_rtsp_media_handle_sdp (GstRTSPMedia * media,
                           GstSDPMessage * sdp)

Configure an SDP on media for receiving streams

Parameters:

media

a GstRTSPMedia

sdp ( [transfer: none])

a GstSDPMessage

Returns

TRUE on success.


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

Returns (Number)

TRUE on success.


GstRtspServer.RTSPMedia.handle_sdp

def GstRtspServer.RTSPMedia.handle_sdp (self, sdp):
    #python wrapper for 'gst_rtsp_media_handle_sdp'

Configure an SDP on media for receiving streams

Returns (bool)

TRUE on success.


gst_rtsp_media_has_completed_sender

gboolean
gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)

See gst_rtsp_stream_is_complete, gst_rtsp_stream_is_sender.

Parameters:

media
No description available
Returns

whether media has at least one complete sender stream.

Since : 1.18


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.

Parameters:

No description available
Returns (Number)

whether media has at least one complete sender stream.

Since : 1.18


GstRtspServer.RTSPMedia.has_completed_sender

def GstRtspServer.RTSPMedia.has_completed_sender (self):
    #python wrapper for 'gst_rtsp_media_has_completed_sender'

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

Parameters:

No description available
Returns (bool)

whether media has at least one complete sender stream.

Since : 1.18


gst_rtsp_media_is_bind_mcast_address

gboolean
gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

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

Since : 1.16


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.

Returns (Number)

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

Since : 1.16


GstRtspServer.RTSPMedia.is_bind_mcast_address

def GstRtspServer.RTSPMedia.is_bind_mcast_address (self):
    #python wrapper for 'gst_rtsp_media_is_bind_mcast_address'

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

Returns (bool)

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

Since : 1.16


gst_rtsp_media_is_eos_shutdown

gboolean
gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

TRUE if the media will send EOS before unpreparing.


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.

Returns (Number)

true if the media will send EOS before unpreparing.


GstRtspServer.RTSPMedia.is_eos_shutdown

def GstRtspServer.RTSPMedia.is_eos_shutdown (self):
    #python wrapper for 'gst_rtsp_media_is_eos_shutdown'

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

Returns (bool)

True if the media will send EOS before unpreparing.


gst_rtsp_media_is_receive_only

gboolean
gst_rtsp_media_is_receive_only (GstRTSPMedia * media)

Parameters:

media
No description available
Returns

TRUE if media is receive-only, FALSE otherwise.

Since : 1.18


GstRtspServer.RTSPMedia.prototype.is_receive_only

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

Parameters:

No description available
Returns (Number)

true if media is receive-only, false otherwise.

Since : 1.18


GstRtspServer.RTSPMedia.is_receive_only

def GstRtspServer.RTSPMedia.is_receive_only (self):
    #python wrapper for 'gst_rtsp_media_is_receive_only'

Parameters:

No description available
Returns (bool)

True if media is receive-only, False otherwise.

Since : 1.18


gst_rtsp_media_is_reusable

gboolean
gst_rtsp_media_is_reusable (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

TRUE if the media can be reused


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.

Returns (Number)

true if the media can be reused


GstRtspServer.RTSPMedia.is_reusable

def GstRtspServer.RTSPMedia.is_reusable (self):
    #python wrapper for 'gst_rtsp_media_is_reusable'

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

Returns (bool)

True if the media can be reused


gst_rtsp_media_is_shared

gboolean
gst_rtsp_media_is_shared (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

TRUE if the media can be shared between clients.


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.

Returns (Number)

true if the media can be shared between clients.


GstRtspServer.RTSPMedia.is_shared

def GstRtspServer.RTSPMedia.is_shared (self):
    #python wrapper for 'gst_rtsp_media_is_shared'

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

Returns (bool)

True if the media can be shared between clients.


gst_rtsp_media_is_stop_on_disconnect

gboolean
gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

TRUE if the media will be stopped when a client disconnects without sending TEARDOWN.


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.

Returns (Number)

true if the media will be stopped when a client disconnects without sending TEARDOWN.


GstRtspServer.RTSPMedia.is_stop_on_disconnect

def GstRtspServer.RTSPMedia.is_stop_on_disconnect (self):
    #python 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.

Returns (bool)

True if the media will be stopped when a client disconnects without sending TEARDOWN.


gst_rtsp_media_is_time_provider

gboolean
gst_rtsp_media_is_time_provider (GstRTSPMedia * media)

Check if media can provide a GstNetTimeProvider for its pipeline clock.

Use gst_rtsp_media_get_time_provider to get the network clock.

Parameters:

media

a GstRTSPMedia

Returns

TRUE if media can provide a GstNetTimeProvider.


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.

Returns (Number)

true if media can provide a GstNet.NetTimeProvider.


GstRtspServer.RTSPMedia.is_time_provider

def GstRtspServer.RTSPMedia.is_time_provider (self):
    #python wrapper for 'gst_rtsp_media_is_time_provider'

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

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

Returns (bool)

True if media can provide a GstNet.NetTimeProvider.


gst_rtsp_media_lock

gst_rtsp_media_lock (GstRTSPMedia * media)

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.

Parameters:

media

a GstRTSPMedia

Since : 1.18


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.lock

def GstRtspServer.RTSPMedia.lock (self):
    #python 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


gst_rtsp_media_n_streams

guint
gst_rtsp_media_n_streams (GstRTSPMedia * media)

Get the number of streams in this media.

Parameters:

media

a GstRTSPMedia

Returns

The number of streams.


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.

Returns (Number)

The number of streams.


GstRtspServer.RTSPMedia.n_streams

def GstRtspServer.RTSPMedia.n_streams (self):
    #python wrapper for 'gst_rtsp_media_n_streams'

Get the number of streams in this media.

Returns (int)

The number of streams.


gst_rtsp_media_prepare

gboolean
gst_rtsp_media_prepare (GstRTSPMedia * media,
                        GstRTSPThread * thread)

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

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

Parameters:

media

a GstRTSPMedia

thread ( [transfer: full][allow-none])

a GstRTSPThread to run the bus handler or NULL

Returns

TRUE on success.


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.

Parameters:

a GstRtspServer.RTSPThread to run the bus handler or null

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.prepare

def GstRtspServer.RTSPMedia.prepare (self, thread):
    #python 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.take_pipeline.

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

Parameters:

a GstRtspServer.RTSPThread to run the bus handler or None

Returns (bool)

True on success.


gst_rtsp_media_seek

gboolean
gst_rtsp_media_seek (GstRTSPMedia * media,
                     GstRTSPTimeRange * range)

Seek the pipeline of media to range. media must be prepared with gst_rtsp_media_prepare.

Parameters:

media

a GstRTSPMedia

range ( [transfer: none])

a GstRTSPTimeRange

Returns

TRUE on success.


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.

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.seek

def GstRtspServer.RTSPMedia.seek (self, range):
    #python wrapper for 'gst_rtsp_media_seek'

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

Returns (bool)

True on success.


gst_rtsp_media_seek_full

gboolean
gst_rtsp_media_seek_full (GstRTSPMedia * media,
                          GstRTSPTimeRange * range,
                          GstSeekFlags flags)

Seek the pipeline of media to range with the given flags. media must be prepared with gst_rtsp_media_prepare.

Parameters:

media

a GstRTSPMedia

range ( [transfer: none])

a GstRTSPTimeRange

flags

The minimal set of GstSeekFlags to use

Returns

TRUE on success.

Since : 1.18


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.

Parameters:

flags (Gst.SeekFlags)

The minimal set of Gst.SeekFlags to use

Returns (Number)

true on success.

Since : 1.18


GstRtspServer.RTSPMedia.seek_full

def GstRtspServer.RTSPMedia.seek_full (self, range, flags):
    #python 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.prepare.

Parameters:

flags (Gst.SeekFlags)

The minimal set of Gst.SeekFlags to use

Returns (bool)

True on success.

Since : 1.18


gst_rtsp_media_seek_trickmode

gboolean
gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
                               GstRTSPTimeRange * range,
                               GstSeekFlags flags,
                               gdouble rate,
                               GstClockTime trickmode_interval)

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

Parameters:

media

a GstRTSPMedia

range ( [transfer: none])

a GstRTSPTimeRange

flags

The minimal set of GstSeekFlags to use

rate

the rate to use in the seek

trickmode_interval

The trickmode interval to use for KEY_UNITS trick mode

Returns

TRUE on success.

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).

Parameters:

flags (Gst.SeekFlags)

The minimal set of Gst.SeekFlags to use

rate (Number)

the rate to use in the seek

trickmode_interval (Number)

The trickmode interval to use for KEY_UNITS trick mode

Returns (Number)

true on success.

Since : 1.18


GstRtspServer.RTSPMedia.seek_trickmode

def GstRtspServer.RTSPMedia.seek_trickmode (self, range, flags, rate, trickmode_interval):
    #python 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.prepare. In order to perform the seek operation, the pipeline must contain all needed transport parts (transport sinks).

Parameters:

flags (Gst.SeekFlags)

The minimal set of Gst.SeekFlags to use

rate (float)

the rate to use in the seek

trickmode_interval (int)

The trickmode interval to use for KEY_UNITS trick mode

Returns (bool)

True on success.

Since : 1.18


gst_rtsp_media_seekable

GstClockTimeDiff
gst_rtsp_media_seekable (GstRTSPMedia * media)

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

Parameters:

media

a GstRTSPMedia

Returns

-1 if the stream is not seekable, 0 if seekable only to the beginning and > 0 to indicate the longest duration between any two random access points. G_MAXINT64 means any value is possible.

Since : 1.14


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.

Returns (Number)

-1 if the stream is not seekable, 0 if seekable only to the beginning and > 0 to indicate the longest duration between any two random access points. GLib.MAXINT64 means any value is possible.

Since : 1.14


GstRtspServer.RTSPMedia.seekable

def GstRtspServer.RTSPMedia.seekable (self):
    #python wrapper for 'gst_rtsp_media_seekable'

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

Returns (int)

-1 if the stream is not seekable, 0 if seekable only to the beginning and > 0 to indicate the longest duration between any two random access points. GLib.MAXINT64 means any value is possible.

Since : 1.14


gst_rtsp_media_set_address_pool

gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
                                 GstRTSPAddressPool * pool)

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

Parameters:

media

a GstRTSPMedia

pool ( [transfer: none][nullable])

a GstRTSPAddressPool


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.set_address_pool

def GstRtspServer.RTSPMedia.set_address_pool (self, pool):
    #python wrapper for 'gst_rtsp_media_set_address_pool'

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


gst_rtsp_media_set_bind_mcast_address

gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
                                       gboolean bind_mcast_addr)

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

Parameters:

media

a GstRTSPMedia

bind_mcast_addr

the new value

Since : 1.16


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.

Parameters:

bind_mcast_addr (Number)

the new value

Since : 1.16


GstRtspServer.RTSPMedia.set_bind_mcast_address

def GstRtspServer.RTSPMedia.set_bind_mcast_address (self, bind_mcast_addr):
    #python wrapper for 'gst_rtsp_media_set_bind_mcast_address'

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

Parameters:

bind_mcast_addr (bool)

the new value

Since : 1.16


gst_rtsp_media_set_buffer_size

gst_rtsp_media_set_buffer_size (GstRTSPMedia * media,
                                guint size)

Set the kernel UDP buffer size.

Parameters:

media

a GstRTSPMedia

size

the new value


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.

Parameters:

size (Number)

the new value


GstRtspServer.RTSPMedia.set_buffer_size

def GstRtspServer.RTSPMedia.set_buffer_size (self, size):
    #python wrapper for 'gst_rtsp_media_set_buffer_size'

Set the kernel UDP buffer size.

Parameters:

size (int)

the new value


gst_rtsp_media_set_clock

gst_rtsp_media_set_clock (GstRTSPMedia * media,
                          GstClock * clock)

Configure the clock used for the media.

Parameters:

media

a GstRTSPMedia

clock ( [nullable])

GstClock to be used


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.

Parameters:

clock (Gst.Clock)

Gst.Clock to be used


GstRtspServer.RTSPMedia.set_clock

def GstRtspServer.RTSPMedia.set_clock (self, clock):
    #python wrapper for 'gst_rtsp_media_set_clock'

Configure the clock used for the media.

Parameters:

clock (Gst.Clock)

Gst.Clock to be used


gst_rtsp_media_set_do_retransmission

gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
                                      gboolean do_retransmission)

Set whether retransmission requests will be sent

Parameters:

media
No description available
do_retransmission
No description available

Since : 1.16


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

Parameters:

No description available
do_retransmission (Number)
No description available

Since : 1.16


GstRtspServer.RTSPMedia.set_do_retransmission

def GstRtspServer.RTSPMedia.set_do_retransmission (self, do_retransmission):
    #python wrapper for 'gst_rtsp_media_set_do_retransmission'

Set whether retransmission requests will be sent

Parameters:

No description available
do_retransmission (bool)
No description available

Since : 1.16


gst_rtsp_media_set_dscp_qos

gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media,
                             gint dscp_qos)

Configure the dscp qos of attached streams to dscp_qos.

Parameters:

media

a GstRTSPMedia

dscp_qos

a new dscp qos value (0-63, or -1 to disable)

Since : 1.18


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.

Parameters:

dscp_qos (Number)

a new dscp qos value (0-63, or -1 to disable)

Since : 1.18


GstRtspServer.RTSPMedia.set_dscp_qos

def GstRtspServer.RTSPMedia.set_dscp_qos (self, dscp_qos):
    #python wrapper for 'gst_rtsp_media_set_dscp_qos'

Configure the dscp qos of attached streams to dscp_qos.

Parameters:

dscp_qos (int)

a new dscp qos value (0-63, or -1 to disable)

Since : 1.18


gst_rtsp_media_set_eos_shutdown

gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media,
                                 gboolean eos_shutdown)

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

Parameters:

media

a GstRTSPMedia

eos_shutdown

the new value


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.

Parameters:

eos_shutdown (Number)

the new value


GstRtspServer.RTSPMedia.set_eos_shutdown

def GstRtspServer.RTSPMedia.set_eos_shutdown (self, eos_shutdown):
    #python 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.

Parameters:

eos_shutdown (bool)

the new value


gst_rtsp_media_set_latency

gst_rtsp_media_set_latency (GstRTSPMedia * media,
                            guint latency)

Configure the latency used for receiving media.

Parameters:

media

a GstRTSPMedia

latency

latency in milliseconds


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.

Parameters:

latency (Number)

latency in milliseconds


GstRtspServer.RTSPMedia.set_latency

def GstRtspServer.RTSPMedia.set_latency (self, latency):
    #python wrapper for 'gst_rtsp_media_set_latency'

Configure the latency used for receiving media.

Parameters:

latency (int)

latency in milliseconds


gst_rtsp_media_set_max_mcast_ttl

gboolean
gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media,
                                  guint ttl)

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

Parameters:

media

a GstRTSPMedia

ttl

the new multicast ttl value

Returns

TRUE if the requested ttl has been set successfully.

Since : 1.16


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.

Parameters:

ttl (Number)

the new multicast ttl value

Returns (Number)

true if the requested ttl has been set successfully.

Since : 1.16


GstRtspServer.RTSPMedia.set_max_mcast_ttl

def GstRtspServer.RTSPMedia.set_max_mcast_ttl (self, ttl):
    #python wrapper for 'gst_rtsp_media_set_max_mcast_ttl'

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

Parameters:

ttl (int)

the new multicast ttl value

Returns (bool)

True if the requested ttl has been set successfully.

Since : 1.16


gst_rtsp_media_set_multicast_iface

gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
                                    const gchar * multicast_iface)

configure multicast_iface to be used for media.

Parameters:

media

a GstRTSPMedia

multicast_iface ( [transfer: none][nullable])

a multicast interface name


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.

Parameters:

multicast_iface (String)

a multicast interface name


GstRtspServer.RTSPMedia.set_multicast_iface

def GstRtspServer.RTSPMedia.set_multicast_iface (self, multicast_iface):
    #python wrapper for 'gst_rtsp_media_set_multicast_iface'

configure multicast_iface to be used for media.

Parameters:

multicast_iface (str)

a multicast interface name


gst_rtsp_media_set_permissions

gst_rtsp_media_set_permissions (GstRTSPMedia * media,
                                GstRTSPPermissions * permissions)

Set permissions on media.

Parameters:

media

a GstRTSPMedia

permissions ( [transfer: none][nullable])

a GstRTSPPermissions


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.set_permissions

def GstRtspServer.RTSPMedia.set_permissions (self, permissions):
    #python wrapper for 'gst_rtsp_media_set_permissions'

Set permissions on media.


gst_rtsp_media_set_pipeline_state

gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
                                   GstState state)

Set the state of the pipeline managed by media to state

Parameters:

media

a GstRTSPMedia

state

the target state of the pipeline


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

Parameters:

state (Gst.State)

the target state of the pipeline


GstRtspServer.RTSPMedia.set_pipeline_state

def GstRtspServer.RTSPMedia.set_pipeline_state (self, state):
    #python wrapper for 'gst_rtsp_media_set_pipeline_state'

Set the state of the pipeline managed by media to state

Parameters:

state (Gst.State)

the target state of the pipeline


gst_rtsp_media_set_profiles

gst_rtsp_media_set_profiles (GstRTSPMedia * media,
                             GstRTSPProfile profiles)

Configure the allowed lower transport for media.

Parameters:

media

a GstRTSPMedia

profiles

the new flags


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.

Parameters:

profiles (GstRtsp.RTSPProfile)

the new flags


GstRtspServer.RTSPMedia.set_profiles

def GstRtspServer.RTSPMedia.set_profiles (self, profiles):
    #python wrapper for 'gst_rtsp_media_set_profiles'

Configure the allowed lower transport for media.

Parameters:

profiles (GstRtsp.RTSPProfile)

the new flags


gst_rtsp_media_set_protocols

gst_rtsp_media_set_protocols (GstRTSPMedia * media,
                              GstRTSPLowerTrans protocols)

Configure the allowed lower transport for media.

Parameters:

media

a GstRTSPMedia

protocols

the new flags


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.

Parameters:

protocols (GstRtsp.RTSPLowerTrans)

the new flags


GstRtspServer.RTSPMedia.set_protocols

def GstRtspServer.RTSPMedia.set_protocols (self, protocols):
    #python wrapper for 'gst_rtsp_media_set_protocols'

Configure the allowed lower transport for media.

Parameters:

protocols (GstRtsp.RTSPLowerTrans)

the new flags


gst_rtsp_media_set_publish_clock_mode

gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
                                       GstRTSPPublishClockMode mode)

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

Parameters:

media

a GstRTSPMedia

mode

the clock publish mode

Since : 1.8


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.

Parameters:

the clock publish mode

Since : 1.8


GstRtspServer.RTSPMedia.set_publish_clock_mode

def GstRtspServer.RTSPMedia.set_publish_clock_mode (self, mode):
    #python wrapper for 'gst_rtsp_media_set_publish_clock_mode'

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

Parameters:

the clock publish mode

Since : 1.8


gst_rtsp_media_set_rate_control

gst_rtsp_media_set_rate_control (GstRTSPMedia * media,
                                 gboolean enabled)

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

Parameters:

media
No description available
enabled
No description available

Since : 1.18


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.

Parameters:

No description available
enabled (Number)
No description available

Since : 1.18


GstRtspServer.RTSPMedia.set_rate_control

def GstRtspServer.RTSPMedia.set_rate_control (self, enabled):
    #python 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.

Parameters:

No description available
enabled (bool)
No description available

Since : 1.18


gst_rtsp_media_set_retransmission_time

gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media,
                                        GstClockTime time)

Set the amount of time to store retransmission packets.

Parameters:

media

a GstRTSPMedia

time

the new value


GstRtspServer.RTSPMedia.prototype.set_retransmission_time

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

Set the amount of time to store retransmission packets.

Parameters:

time (Number)

the new value


GstRtspServer.RTSPMedia.set_retransmission_time

def GstRtspServer.RTSPMedia.set_retransmission_time (self, time):
    #python wrapper for 'gst_rtsp_media_set_retransmission_time'

Set the amount of time to store retransmission packets.

Parameters:

time (int)

the new value


gst_rtsp_media_set_reusable

gst_rtsp_media_set_reusable (GstRTSPMedia * media,
                             gboolean reusable)

Set or unset if the pipeline for media can be reused after the pipeline has been unprepared.

Parameters:

media

a GstRTSPMedia

reusable

the new value


GstRtspServer.RTSPMedia.prototype.set_reusable

function GstRtspServer.RTSPMedia.prototype.set_reusable(reusable: Number): {
    // javascript wrapper for 'gst_rtsp_media_set_reusable'
}

Set or unset if the pipeline for media can be reused after the pipeline has been unprepared.

Parameters:

reusable (Number)

the new value


GstRtspServer.RTSPMedia.set_reusable

def GstRtspServer.RTSPMedia.set_reusable (self, reusable):
    #python wrapper for 'gst_rtsp_media_set_reusable'

Set or unset if the pipeline for media can be reused after the pipeline has been unprepared.

Parameters:

reusable (bool)

the new value


gst_rtsp_media_set_shared

gst_rtsp_media_set_shared (GstRTSPMedia * media,
                           gboolean shared)

Set or unset if the pipeline for media can be shared will multiple clients. When shared is TRUE, client requests for this media will share the media pipeline.

Parameters:

media

a GstRTSPMedia

shared

the new value


GstRtspServer.RTSPMedia.prototype.set_shared

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

Set or unset if the pipeline for media can be shared will multiple clients. When shared is true, client requests for this media will share the media pipeline.

Parameters:

shared (Number)

the new value


GstRtspServer.RTSPMedia.set_shared

def GstRtspServer.RTSPMedia.set_shared (self, shared):
    #python wrapper for 'gst_rtsp_media_set_shared'

Set or unset if the pipeline for media can be shared will multiple clients. When shared is True, client requests for this media will share the media pipeline.

Parameters:

shared (bool)

the new value


gst_rtsp_media_set_state

gboolean
gst_rtsp_media_set_state (GstRTSPMedia * media,
                          GstState state,
                          GPtrArray * transports)

Set the state of media to state and for the transports in transports.

media must be prepared with gst_rtsp_media_prepare;

Parameters:

media

a GstRTSPMedia

state

the target state of the media

transports

(transfer none) (element-type GstRtspServer.RTSPStreamTransport): a GPtrArray of GstRTSPStreamTransport pointers

Returns

TRUE on success.


GstRtspServer.RTSPMedia.prototype.set_state

function GstRtspServer.RTSPMedia.prototype.set_state(state: Gst.State, transports: [ GstRtspServer.RTSPStreamTransport ]): {
    // javascript wrapper for 'gst_rtsp_media_set_state'
}

Set the state of media to state and for the transports in transports.

media must be prepared with GstRtspServer.RTSPMedia.prototype.prepare;

Parameters:

state (Gst.State)

the target state of the media

transports ([ GstRtspServer.RTSPStreamTransport ])

(transfer none) (element-type GstRtspServer.RTSPStreamTransport): a GLib.PtrArray of GstRtspServer.RTSPStreamTransport pointers

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.set_state

def GstRtspServer.RTSPMedia.set_state (self, state, transports):
    #python wrapper for 'gst_rtsp_media_set_state'

Set the state of media to state and for the transports in transports.

media must be prepared with GstRtspServer.RTSPMedia.prepare;

Parameters:

state (Gst.State)

the target state of the media

transports ([ GstRtspServer.RTSPStreamTransport ])

(transfer none) (element-type GstRtspServer.RTSPStreamTransport): a GLib.PtrArray of GstRtspServer.RTSPStreamTransport pointers

Returns (bool)

True on success.


gst_rtsp_media_set_stop_on_disconnect

gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
                                       gboolean stop_on_disconnect)

Set or unset if the pipeline for media should be stopped when a client disconnects without sending TEARDOWN.

Parameters:

media

a GstRTSPMedia

stop_on_disconnect

the new value


GstRtspServer.RTSPMedia.prototype.set_stop_on_disconnect

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

Set or unset if the pipeline for media should be stopped when a client disconnects without sending TEARDOWN.

Parameters:

stop_on_disconnect (Number)

the new value


GstRtspServer.RTSPMedia.set_stop_on_disconnect

def GstRtspServer.RTSPMedia.set_stop_on_disconnect (self, stop_on_disconnect):
    #python wrapper for 'gst_rtsp_media_set_stop_on_disconnect'

Set or unset if the pipeline for media should be stopped when a client disconnects without sending TEARDOWN.

Parameters:

stop_on_disconnect (bool)

the new value


gst_rtsp_media_set_suspend_mode

gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media,
                                 GstRTSPSuspendMode mode)

Control how media will be suspended after the SDP has been generated and after a PAUSE request has been performed.

Media must be unprepared when setting the suspend mode.

Parameters:

media

a GstRTSPMedia

mode

the new GstRTSPSuspendMode


GstRtspServer.RTSPMedia.prototype.set_suspend_mode

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

Control how media will be suspended after the SDP has been generated and after a PAUSE request has been performed.

Media must be unprepared when setting the suspend mode.


GstRtspServer.RTSPMedia.set_suspend_mode

def GstRtspServer.RTSPMedia.set_suspend_mode (self, mode):
    #python wrapper for 'gst_rtsp_media_set_suspend_mode'

Control how media will be suspended after the SDP has been generated and after a PAUSE request has been performed.

Media must be unprepared when setting the suspend mode.


gst_rtsp_media_set_transport_mode

gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
                                   GstRTSPTransportMode mode)

Sets if the media pipeline can work in PLAY or RECORD mode

Parameters:

media

a GstRTSPMedia

mode

the new value


GstRtspServer.RTSPMedia.prototype.set_transport_mode

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

Sets if the media pipeline can work in PLAY or RECORD mode

Parameters:

the new value


GstRtspServer.RTSPMedia.set_transport_mode

def GstRtspServer.RTSPMedia.set_transport_mode (self, mode):
    #python wrapper for 'gst_rtsp_media_set_transport_mode'

Sets if the media pipeline can work in PLAY or RECORD mode

Parameters:

the new value


gst_rtsp_media_setup_sdp

gboolean
gst_rtsp_media_setup_sdp (GstRTSPMedia * media,
                          GstSDPMessage * sdp,
                          GstSDPInfo * info)

Add media specific info to sdp. info is used to configure the connection information in the SDP.

Parameters:

media

a GstRTSPMedia

sdp ( [transfer: none])

a GstSDPMessage

info ( [transfer: none])

a GstSDPInfo

Returns

TRUE on success.


GstRtspServer.RTSPMedia.prototype.setup_sdp

function GstRtspServer.RTSPMedia.prototype.setup_sdp(sdp: GstSdp.SDPMessage, info: GstRtspServer.SDPInfo): {
    // javascript wrapper for 'gst_rtsp_media_setup_sdp'
}

Add media specific info to sdp. info is used to configure the connection information in the SDP.

Returns (Number)

TRUE on success.


GstRtspServer.RTSPMedia.setup_sdp

def GstRtspServer.RTSPMedia.setup_sdp (self, sdp, info):
    #python wrapper for 'gst_rtsp_media_setup_sdp'

Add media specific info to sdp. info is used to configure the connection information in the SDP.

Returns (bool)

TRUE on success.


gst_rtsp_media_suspend

gboolean
gst_rtsp_media_suspend (GstRTSPMedia * media)

Suspend media. The state of the pipeline managed by media is set to GST_STATE_NULL but all streams are kept. media can be prepared again with gst_rtsp_media_unsuspend

media must be prepared with gst_rtsp_media_prepare;

Parameters:

media

a GstRTSPMedia

Returns

TRUE on success.


GstRtspServer.RTSPMedia.prototype.suspend

function GstRtspServer.RTSPMedia.prototype.suspend(): {
    // javascript wrapper for 'gst_rtsp_media_suspend'
}

Suspend media. The state of the pipeline managed by media is set to GST_STATE_NULL but all streams are kept. media can be prepared again with GstRtspServer.RTSPMedia.prototype.unsuspend

media must be prepared with GstRtspServer.RTSPMedia.prototype.prepare;

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.suspend

def GstRtspServer.RTSPMedia.suspend (self):
    #python wrapper for 'gst_rtsp_media_suspend'

Suspend media. The state of the pipeline managed by media is set to GST_STATE_NULL but all streams are kept. media can be prepared again with GstRtspServer.RTSPMedia.unsuspend

media must be prepared with GstRtspServer.RTSPMedia.prepare;

Returns (bool)

True on success.


gst_rtsp_media_take_pipeline

gst_rtsp_media_take_pipeline (GstRTSPMedia * media,
                              GstPipeline * pipeline)

Set pipeline as the GstPipeline for media. Ownership is taken of pipeline.

Parameters:

media

a GstRTSPMedia

pipeline ( [transfer: floating])

a GstPipeline


GstRtspServer.RTSPMedia.prototype.take_pipeline

function GstRtspServer.RTSPMedia.prototype.take_pipeline(pipeline: Gst.Pipeline): {
    // javascript wrapper for 'gst_rtsp_media_take_pipeline'
}

Set pipeline as the Gst.Pipeline for media. Ownership is taken of pipeline.

Parameters:

pipeline (Gst.Pipeline)

a Gst.Pipeline


GstRtspServer.RTSPMedia.take_pipeline

def GstRtspServer.RTSPMedia.take_pipeline (self, pipeline):
    #python wrapper for 'gst_rtsp_media_take_pipeline'

Set pipeline as the Gst.Pipeline for media. Ownership is taken of pipeline.

Parameters:

pipeline (Gst.Pipeline)

a Gst.Pipeline


gst_rtsp_media_unlock

gst_rtsp_media_unlock (GstRTSPMedia * media)

Unlock the media.

Parameters:

media

a GstRTSPMedia

Since : 1.18


GstRtspServer.RTSPMedia.prototype.unlock

function GstRtspServer.RTSPMedia.prototype.unlock(): {
    // javascript wrapper for 'gst_rtsp_media_unlock'
}

Unlock the media.

Since : 1.18


GstRtspServer.RTSPMedia.unlock

def GstRtspServer.RTSPMedia.unlock (self):
    #python wrapper for 'gst_rtsp_media_unlock'

Unlock the media.

Since : 1.18


gst_rtsp_media_unprepare

gboolean
gst_rtsp_media_unprepare (GstRTSPMedia * media)

Unprepare media. After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

Parameters:

media

a GstRTSPMedia

Returns

TRUE on success.


GstRtspServer.RTSPMedia.prototype.unprepare

function GstRtspServer.RTSPMedia.prototype.unprepare(): {
    // javascript wrapper for 'gst_rtsp_media_unprepare'
}

Unprepare media. After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.unprepare

def GstRtspServer.RTSPMedia.unprepare (self):
    #python wrapper for 'gst_rtsp_media_unprepare'

Unprepare media. After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

Returns (bool)

True on success.


gst_rtsp_media_unsuspend

gboolean
gst_rtsp_media_unsuspend (GstRTSPMedia * media)

Unsuspend media if it was in a suspended state. This method does nothing when the media was not in the suspended state.

Parameters:

media

a GstRTSPMedia

Returns

TRUE on success.


GstRtspServer.RTSPMedia.prototype.unsuspend

function GstRtspServer.RTSPMedia.prototype.unsuspend(): {
    // javascript wrapper for 'gst_rtsp_media_unsuspend'
}

Unsuspend media if it was in a suspended state. This method does nothing when the media was not in the suspended state.

Returns (Number)

true on success.


GstRtspServer.RTSPMedia.unsuspend

def GstRtspServer.RTSPMedia.unsuspend (self):
    #python wrapper for 'gst_rtsp_media_unsuspend'

Unsuspend media if it was in a suspended state. This method does nothing when the media was not in the suspended state.

Returns (bool)

True on success.


gst_rtsp_media_use_time_provider

gst_rtsp_media_use_time_provider (GstRTSPMedia * media,
                                  gboolean time_provider)

Set media to provide a GstNetTimeProvider.

Parameters:

media

a GstRTSPMedia

time_provider

if a GstNetTimeProvider should be used


GstRtspServer.RTSPMedia.prototype.use_time_provider

function GstRtspServer.RTSPMedia.prototype.use_time_provider(time_provider: Number): {
    // javascript wrapper for 'gst_rtsp_media_use_time_provider'
}

Set media to provide a GstNet.NetTimeProvider.

Parameters:

time_provider (Number)

if a GstNet.NetTimeProvider should be used


GstRtspServer.RTSPMedia.use_time_provider

def GstRtspServer.RTSPMedia.use_time_provider (self, time_provider):
    #python wrapper for 'gst_rtsp_media_use_time_provider'

Set media to provide a GstNet.NetTimeProvider.

Parameters:

time_provider (bool)

if a GstNet.NetTimeProvider should be used


Signals

new-state

new_state_callback (GstRTSPMedia * self,
                    gint object,
                    gpointer user_data)

Parameters:

self
No description available
object
No description available
user_data
No description available

Flags: Run Last


new-state

function new_state_callback(self: GstRtspServer.RTSPMedia, object: Number, user_data: Object): {
    // javascript callback for the 'new-state' signal
}

Parameters:

No description available
object (Number)
No description available
user_data (Object)
No description available

Flags: Run Last


new-state

def new_state_callback (self, object, *user_data):
    #python callback for the 'new-state' signal

Parameters:

No description available
object (int)
No description available
user_data (variadic)
No description available

Flags: Run Last


new-stream

new_stream_callback (GstRTSPMedia * self,
                     GstRTSPStream * object,
                     gpointer user_data)

Parameters:

self
No description available
object
No description available
user_data
No description available

Flags: Run Last


new-stream

function new_stream_callback(self: GstRtspServer.RTSPMedia, object: GstRtspServer.RTSPStream, user_data: Object): {
    // javascript callback for the 'new-stream' signal
}

Parameters:

No description available
No description available
user_data (Object)
No description available

Flags: Run Last


new-stream

def new_stream_callback (self, object, *user_data):
    #python callback for the 'new-stream' signal

Parameters:

No description available
No description available
user_data (variadic)
No description available

Flags: Run Last


prepared

prepared_callback (GstRTSPMedia * self,
                   gpointer user_data)

Parameters:

self
No description available
user_data
No description available

Flags: Run Last


prepared

function prepared_callback(self: GstRtspServer.RTSPMedia, user_data: Object): {
    // javascript callback for the 'prepared' signal
}

Parameters:

No description available
user_data (Object)
No description available

Flags: Run Last


prepared

def prepared_callback (self, *user_data):
    #python callback for the 'prepared' signal

Parameters:

No description available
user_data (variadic)
No description available

Flags: Run Last


removed-stream

removed_stream_callback (GstRTSPMedia * self,
                         GstRTSPStream * object,
                         gpointer user_data)

Parameters:

self
No description available
object
No description available
user_data
No description available

Flags: Run Last


removed-stream

function removed_stream_callback(self: GstRtspServer.RTSPMedia, object: GstRtspServer.RTSPStream, user_data: Object): {
    // javascript callback for the 'removed-stream' signal
}

Parameters:

No description available
No description available
user_data (Object)
No description available

Flags: Run Last


removed-stream

def removed_stream_callback (self, object, *user_data):
    #python callback for the 'removed-stream' signal

Parameters:

No description available
No description available
user_data (variadic)
No description available

Flags: Run Last


target-state

target_state_callback (GstRTSPMedia * self,
                       gint object,
                       gpointer user_data)

Parameters:

self
No description available
object
No description available
user_data
No description available

Flags: Run Last


target-state

function target_state_callback(self: GstRtspServer.RTSPMedia, object: Number, user_data: Object): {
    // javascript callback for the 'target-state' signal
}

Parameters:

No description available
object (Number)
No description available
user_data (Object)
No description available

Flags: Run Last


target-state

def target_state_callback (self, object, *user_data):
    #python callback for the 'target-state' signal

Parameters:

No description available
object (int)
No description available
user_data (variadic)
No description available

Flags: Run Last


unprepared

unprepared_callback (GstRTSPMedia * self,
                     gpointer user_data)

Parameters:

self
No description available
user_data
No description available

Flags: Run Last


unprepared

function unprepared_callback(self: GstRtspServer.RTSPMedia, user_data: Object): {
    // javascript callback for the 'unprepared' signal
}

Parameters:

No description available
user_data (Object)
No description available

Flags: Run Last


unprepared

def unprepared_callback (self, *user_data):
    #python callback for the 'unprepared' signal

Parameters:

No description available
user_data (variadic)
No description available

Flags: Run Last


Properties

bind-mcast-address

“bind-mcast-address” gboolean

Flags : Read / Write


bind-mcast-address

“bind-mcast-address” Number

Flags : Read / Write


bind_mcast_address

“self.props.bind_mcast_address” bool

Flags : Read / Write


buffer-size

“buffer-size” guint

Flags : Read / Write


buffer-size

“buffer-size” Number

Flags : Read / Write


buffer_size

“self.props.buffer_size” int

Flags : Read / Write


clock

“clock” GstClock *

Flags : Read / Write


clock

“clock” Gst.Clock

Flags : Read / Write


clock

“self.props.clock” Gst.Clock

Flags : Read / Write


dscp-qos

“dscp-qos” gint

Flags : Read / Write


dscp-qos

“dscp-qos” Number

Flags : Read / Write


dscp_qos

“self.props.dscp_qos” int

Flags : Read / Write


element

“element” GstElement *

Flags : Read / Write / Construct Only


element

“element” Gst.Element

Flags : Read / Write / Construct Only


element

“self.props.element” Gst.Element

Flags : Read / Write / Construct Only


eos-shutdown

“eos-shutdown” gboolean

Flags : Read / Write


eos-shutdown

“eos-shutdown” Number

Flags : Read / Write


eos_shutdown

“self.props.eos_shutdown” bool

Flags : Read / Write


latency

“latency” guint

Flags : Read / Write


latency

“latency” Number

Flags : Read / Write


latency

“self.props.latency” int

Flags : Read / Write


max-mcast-ttl

“max-mcast-ttl” guint

Flags : Read / Write


max-mcast-ttl

“max-mcast-ttl” Number

Flags : Read / Write


max_mcast_ttl

“self.props.max_mcast_ttl” int

Flags : Read / Write


profiles

“profiles” GstRTSPProfile *

Flags : Read / Write


profiles

“profiles” GstRtsp.RTSPProfile

Flags : Read / Write


profiles

“self.props.profiles” GstRtsp.RTSPProfile

Flags : Read / Write


protocols

“protocols” GstRTSPLowerTrans *

Flags : Read / Write


protocols

“protocols” GstRtsp.RTSPLowerTrans

Flags : Read / Write


protocols

“self.props.protocols” GstRtsp.RTSPLowerTrans

Flags : Read / Write


reusable

“reusable” gboolean

Flags : Read / Write


reusable

“reusable” Number

Flags : Read / Write


reusable

“self.props.reusable” bool

Flags : Read / Write


shared

“shared” gboolean

Flags : Read / Write


shared

“shared” Number

Flags : Read / Write


shared

“self.props.shared” bool

Flags : Read / Write


stop-on-disconnect

“stop-on-disconnect” gboolean

Flags : Read / Write


stop-on-disconnect

“stop-on-disconnect” Number

Flags : Read / Write


stop_on_disconnect

“self.props.stop_on_disconnect” bool

Flags : Read / Write


suspend-mode

“suspend-mode” GstRTSPSuspendMode *

Flags : Read / Write


suspend-mode

“suspend-mode” GstRtspServer.RTSPSuspendMode

Flags : Read / Write


suspend_mode

“self.props.suspend_mode” GstRtspServer.RTSPSuspendMode

Flags : Read / Write


time-provider

“time-provider” gboolean

Flags : Read / Write


time-provider

“time-provider” Number

Flags : Read / Write


time_provider

“self.props.time_provider” bool

Flags : Read / Write


transport-mode

“transport-mode” GstRTSPTransportMode *

Flags : Read / Write


transport-mode

“transport-mode” GstRtspServer.RTSPTransportMode

Flags : Read / Write


transport_mode

“self.props.transport_mode” GstRtspServer.RTSPTransportMode

Flags : Read / Write


Virtual Methods

convert_range

gboolean
convert_range (GstRTSPMedia * media,
               GstRTSPTimeRange * range,
               GstRTSPRangeUnit unit)

convert a range to the given unit

Parameters:

media
No description available
range
No description available
unit
No description available
Returns
No description available

vfunc_convert_range

function vfunc_convert_range(media: GstRtspServer.RTSPMedia, range: GstRtsp.RTSPTimeRange, unit: GstRtsp.RTSPRangeUnit): {
    // javascript implementation of the 'convert_range' virtual method
}

convert a range to the given unit

Parameters:

No description available
range (GstRtsp.RTSPTimeRange)
No description available
No description available
Returns (Number)
No description available

do_convert_range

def do_convert_range (media, range, unit):
    #python implementation of the 'convert_range' virtual method

convert a range to the given unit

Parameters:

No description available
range (GstRtsp.RTSPTimeRange)
No description available
No description available
Returns (bool)
No description available

create_rtpbin

GstElement *
create_rtpbin (GstRTSPMedia * media)

Parameters:

media
No description available
Returns
No description available

handle_message

gboolean
handle_message (GstRTSPMedia * media,
                GstMessage * message)

handle a message

Parameters:

media
No description available
message
No description available
Returns
No description available

vfunc_handle_message

function vfunc_handle_message(media: GstRtspServer.RTSPMedia, message: Gst.Message): {
    // javascript implementation of the 'handle_message' virtual method
}

handle a message

Parameters:

No description available
message (Gst.Message)
No description available
Returns (Number)
No description available

do_handle_message

def do_handle_message (media, message):
    #python implementation of the 'handle_message' virtual method

handle a message

Parameters:

No description available
message (Gst.Message)
No description available
Returns (bool)
No description available

handle_sdp

gboolean
handle_sdp (GstRTSPMedia * media,
            GstSDPMessage * sdp)

Parameters:

media
No description available
sdp
No description available
Returns
No description available

vfunc_handle_sdp

function vfunc_handle_sdp(media: GstRtspServer.RTSPMedia, sdp: GstSdp.SDPMessage): {
    // javascript implementation of the 'handle_sdp' virtual method
}

Parameters:

No description available
sdp (GstSdp.SDPMessage)
No description available
Returns (Number)
No description available

do_handle_sdp

def do_handle_sdp (media, sdp):
    #python implementation of the 'handle_sdp' virtual method

Parameters:

No description available
sdp (GstSdp.SDPMessage)
No description available
Returns (bool)
No description available

new_state

new_state (GstRTSPMedia * media,
           GstState state)

Parameters:

media
No description available
state
No description available

vfunc_new_state

function vfunc_new_state(media: GstRtspServer.RTSPMedia, state: Gst.State): {
    // javascript implementation of the 'new_state' virtual method
}

Parameters:

No description available
state (Gst.State)
No description available

do_new_state

def do_new_state (media, state):
    #python implementation of the 'new_state' virtual method

Parameters:

No description available
state (Gst.State)
No description available

new_stream

new_stream (GstRTSPMedia * media,
            GstRTSPStream * stream)

Parameters:

media
No description available
stream
No description available

vfunc_new_stream

function vfunc_new_stream(media: GstRtspServer.RTSPMedia, stream: GstRtspServer.RTSPStream): {
    // javascript implementation of the 'new_stream' virtual method
}

Parameters:

No description available
No description available

do_new_stream

def do_new_stream (media, stream):
    #python implementation of the 'new_stream' virtual method

Parameters:

No description available
No description available

prepare

gboolean
prepare (GstRTSPMedia * media,
         GstRTSPThread * thread)

the default implementation adds all elements and sets the pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING in case of NO_PREROLL elements).

Parameters:

media
No description available
thread
No description available
Returns
No description available

vfunc_prepare

function vfunc_prepare(media: GstRtspServer.RTSPMedia, thread: GstRtspServer.RTSPThread): {
    // javascript implementation of the 'prepare' virtual method
}

the default implementation adds all elements and sets the pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING in case of NO_PREROLL elements).

Parameters:

No description available
No description available
Returns (Number)
No description available

do_prepare

def do_prepare (media, thread):
    #python implementation of the 'prepare' virtual method

the default implementation adds all elements and sets the pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING in case of NO_PREROLL elements).

Parameters:

No description available
No description available
Returns (bool)
No description available

prepared

prepared (GstRTSPMedia * media)

Parameters:

media
No description available

vfunc_prepared

function vfunc_prepared(media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'prepared' virtual method
}

Parameters:

No description available

do_prepared

def do_prepared (media):
    #python implementation of the 'prepared' virtual method

Parameters:

No description available

query_position

gboolean
query_position (GstRTSPMedia * media,
                gint64 * position)

query the current position in the pipeline

Parameters:

media
No description available
position
No description available
Returns
No description available

vfunc_query_position

function vfunc_query_position(media: GstRtspServer.RTSPMedia, position: Number): {
    // javascript implementation of the 'query_position' virtual method
}

query the current position in the pipeline

Parameters:

No description available
position (Number)
No description available
Returns (Number)
No description available

do_query_position

def do_query_position (media, position):
    #python implementation of the 'query_position' virtual method

query the current position in the pipeline

Parameters:

No description available
position (int)
No description available
Returns (bool)
No description available

query_stop

gboolean
query_stop (GstRTSPMedia * media,
            gint64 * stop)

query when playback will stop

Parameters:

media
No description available
stop
No description available
Returns
No description available

vfunc_query_stop

function vfunc_query_stop(media: GstRtspServer.RTSPMedia, stop: Number): {
    // javascript implementation of the 'query_stop' virtual method
}

query when playback will stop

Parameters:

No description available
stop (Number)
No description available
Returns (Number)
No description available

do_query_stop

def do_query_stop (media, stop):
    #python implementation of the 'query_stop' virtual method

query when playback will stop

Parameters:

No description available
stop (int)
No description available
Returns (bool)
No description available

removed_stream

removed_stream (GstRTSPMedia * media,
                GstRTSPStream * stream)

Parameters:

media
No description available
stream
No description available

vfunc_removed_stream

function vfunc_removed_stream(media: GstRtspServer.RTSPMedia, stream: GstRtspServer.RTSPStream): {
    // javascript implementation of the 'removed_stream' virtual method
}

Parameters:

No description available
No description available

do_removed_stream

def do_removed_stream (media, stream):
    #python implementation of the 'removed_stream' virtual method

Parameters:

No description available
No description available

setup_rtpbin

gboolean
setup_rtpbin (GstRTSPMedia * media,
              GstElement * rtpbin)

Parameters:

media
No description available
rtpbin
No description available
Returns
No description available

vfunc_setup_rtpbin

function vfunc_setup_rtpbin(media: GstRtspServer.RTSPMedia, rtpbin: Gst.Element): {
    // javascript implementation of the 'setup_rtpbin' virtual method
}

Parameters:

No description available
rtpbin (Gst.Element)
No description available
Returns (Number)
No description available

do_setup_rtpbin

def do_setup_rtpbin (media, rtpbin):
    #python implementation of the 'setup_rtpbin' virtual method

Parameters:

No description available
rtpbin (Gst.Element)
No description available
Returns (bool)
No description available

setup_sdp

gboolean
setup_sdp (GstRTSPMedia * media,
           GstSDPMessage * sdp,
           GstSDPInfo * info)

Parameters:

media
No description available
sdp
No description available
info
No description available
Returns
No description available

vfunc_setup_sdp

function vfunc_setup_sdp(media: GstRtspServer.RTSPMedia, sdp: GstSdp.SDPMessage, info: GstRtspServer.SDPInfo): {
    // javascript implementation of the 'setup_sdp' virtual method
}

Parameters:

No description available
sdp (GstSdp.SDPMessage)
No description available
No description available
Returns (Number)
No description available

do_setup_sdp

def do_setup_sdp (media, sdp, info):
    #python implementation of the 'setup_sdp' virtual method

Parameters:

No description available
sdp (GstSdp.SDPMessage)
No description available
No description available
Returns (bool)
No description available

suspend

gboolean
suspend (GstRTSPMedia * media)

the default implementation sets the pipeline's state to GST_STATE_NULL GST_STATE_PAUSED depending on the selected suspend mode.

Parameters:

media
No description available
Returns
No description available

vfunc_suspend

function vfunc_suspend(media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'suspend' virtual method
}

the default implementation sets the pipeline's state to GST_STATE_NULL GST_STATE_PAUSED depending on the selected suspend mode.

Parameters:

No description available
Returns (Number)
No description available

do_suspend

def do_suspend (media):
    #python implementation of the 'suspend' virtual method

the default implementation sets the pipeline's state to GST_STATE_NULL GST_STATE_PAUSED depending on the selected suspend mode.

Parameters:

No description available
Returns (bool)
No description available

target_state

target_state (GstRTSPMedia * media,
              GstState state)

Parameters:

media
No description available
state
No description available

vfunc_target_state

function vfunc_target_state(media: GstRtspServer.RTSPMedia, state: Gst.State): {
    // javascript implementation of the 'target_state' virtual method
}

Parameters:

No description available
state (Gst.State)
No description available

do_target_state

def do_target_state (media, state):
    #python implementation of the 'target_state' virtual method

Parameters:

No description available
state (Gst.State)
No description available

unprepare

gboolean
unprepare (GstRTSPMedia * media)

the default implementation sets the pipeline's state to GST_STATE_NULL and removes all elements.

Parameters:

media
No description available
Returns
No description available

vfunc_unprepare

function vfunc_unprepare(media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'unprepare' virtual method
}

the default implementation sets the pipeline's state to GST_STATE_NULL and removes all elements.

Parameters:

No description available
Returns (Number)
No description available

do_unprepare

def do_unprepare (media):
    #python implementation of the 'unprepare' virtual method

the default implementation sets the pipeline's state to GST_STATE_NULL and removes all elements.

Parameters:

No description available
Returns (bool)
No description available

unprepared

unprepared (GstRTSPMedia * media)

Parameters:

media
No description available

vfunc_unprepared

function vfunc_unprepared(media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'unprepared' virtual method
}

Parameters:

No description available

do_unprepared

def do_unprepared (media):
    #python implementation of the 'unprepared' virtual method

Parameters:

No description available

unsuspend

gboolean
unsuspend (GstRTSPMedia * media)

the default implementation reverts the suspend operation. The pipeline will be prerolled again if it's state was set to GST_STATE_NULL in suspend.

Parameters:

media
No description available
Returns
No description available

vfunc_unsuspend

function vfunc_unsuspend(media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'unsuspend' virtual method
}

the default implementation reverts the suspend operation. The pipeline will be prerolled again if it's state was set to GST_STATE_NULL in suspend.

Parameters:

No description available
Returns (Number)
No description available

do_unsuspend

def do_unsuspend (media):
    #python implementation of the 'unsuspend' virtual method

the default implementation reverts the suspend operation. The pipeline will be prerolled again if it's state was set to GST_STATE_NULL in suspend.

Parameters:

No description available
Returns (bool)
No description available

Function Macros

GST_RTSP_MEDIA_CAST

#define GST_RTSP_MEDIA_CAST(obj)         ((GstRTSPMedia*)(obj))

GST_RTSP_MEDIA_CLASS_CAST

#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))

Enumerations

GstRTSPMediaStatus

The state of the media pipeline.

Members
GST_RTSP_MEDIA_STATUS_UNPREPARED (0) –

media pipeline not prerolled

GST_RTSP_MEDIA_STATUS_UNPREPARING (1) –

media pipeline is busy doing a clean shutdown.

GST_RTSP_MEDIA_STATUS_PREPARING (2) –

media pipeline is prerolling

GST_RTSP_MEDIA_STATUS_PREPARED (3) –

media pipeline is prerolled

GST_RTSP_MEDIA_STATUS_SUSPENDED (4) –

media is suspended

GST_RTSP_MEDIA_STATUS_ERROR (5) –

media pipeline is in error


GstRtspServer.RTSPMediaStatus

The state of the media pipeline.

Members
GstRtspServer.RTSPMediaStatus.UNPREPARED (0) –

media pipeline not prerolled

GstRtspServer.RTSPMediaStatus.UNPREPARING (1) –

media pipeline is busy doing a clean shutdown.

GstRtspServer.RTSPMediaStatus.PREPARING (2) –

media pipeline is prerolling

GstRtspServer.RTSPMediaStatus.PREPARED (3) –

media pipeline is prerolled

GstRtspServer.RTSPMediaStatus.SUSPENDED (4) –

media is suspended

GstRtspServer.RTSPMediaStatus.ERROR (5) –

media pipeline is in error


GstRtspServer.RTSPMediaStatus

The state of the media pipeline.

Members
GstRtspServer.RTSPMediaStatus.UNPREPARED (0) –

media pipeline not prerolled

GstRtspServer.RTSPMediaStatus.UNPREPARING (1) –

media pipeline is busy doing a clean shutdown.

GstRtspServer.RTSPMediaStatus.PREPARING (2) –

media pipeline is prerolling

GstRtspServer.RTSPMediaStatus.PREPARED (3) –

media pipeline is prerolled

GstRtspServer.RTSPMediaStatus.SUSPENDED (4) –

media is suspended

GstRtspServer.RTSPMediaStatus.ERROR (5) –

media pipeline is in error


GstRTSPPublishClockMode

Whether the clock and possibly RTP/clock offset should be published according to RFC7273.

Members
GST_RTSP_PUBLISH_CLOCK_MODE_NONE (0) –

Publish nothing

GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK (1) –

Publish the clock but not the offset

GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET (2) –

Publish the clock and offset


GstRtspServer.RTSPPublishClockMode

Whether the clock and possibly RTP/clock offset should be published according to RFC7273.

Members
GstRtspServer.RTSPPublishClockMode.NONE (0) –

Publish nothing

GstRtspServer.RTSPPublishClockMode.CLOCK (1) –

Publish the clock but not the offset

GstRtspServer.RTSPPublishClockMode.CLOCK_AND_OFFSET (2) –

Publish the clock and offset


GstRtspServer.RTSPPublishClockMode

Whether the clock and possibly RTP/clock offset should be published according to RFC7273.

Members
GstRtspServer.RTSPPublishClockMode.NONE (0) –

Publish nothing

GstRtspServer.RTSPPublishClockMode.CLOCK (1) –

Publish the clock but not the offset

GstRtspServer.RTSPPublishClockMode.CLOCK_AND_OFFSET (2) –

Publish the clock and offset


GstRTSPSuspendMode

The suspend mode of the media pipeline. A media pipeline is suspended right after creating the SDP and when the client performs a PAUSED request.

Members
GST_RTSP_SUSPEND_MODE_NONE (0) –

Media is not suspended

GST_RTSP_SUSPEND_MODE_PAUSE (1) –

Media is PAUSED in suspend

GST_RTSP_SUSPEND_MODE_RESET (2) –

The media is set to NULL when suspended


GstRtspServer.RTSPSuspendMode

The suspend mode of the media pipeline. A media pipeline is suspended right after creating the SDP and when the client performs a PAUSED request.

Members
GstRtspServer.RTSPSuspendMode.NONE (0) –

Media is not suspended

GstRtspServer.RTSPSuspendMode.PAUSE (1) –

Media is PAUSED in suspend

GstRtspServer.RTSPSuspendMode.RESET (2) –

The media is set to NULL when suspended


GstRtspServer.RTSPSuspendMode

The suspend mode of the media pipeline. A media pipeline is suspended right after creating the SDP and when the client performs a PAUSED request.

Members
GstRtspServer.RTSPSuspendMode.NONE (0) –

Media is not suspended

GstRtspServer.RTSPSuspendMode.PAUSE (1) –

Media is PAUSED in suspend

GstRtspServer.RTSPSuspendMode.RESET (2) –

The media is set to NULL when suspended


GstRTSPTransportMode

The supported modes of the media.

Members
GST_RTSP_TRANSPORT_MODE_PLAY (1) –

Transport supports PLAY mode

GST_RTSP_TRANSPORT_MODE_RECORD (2) –

Transport supports RECORD mode


GstRtspServer.RTSPTransportMode

The supported modes of the media.

Members
GstRtspServer.RTSPTransportMode.PLAY (1) –

Transport supports PLAY mode

GstRtspServer.RTSPTransportMode.RECORD (2) –

Transport supports RECORD mode


GstRtspServer.RTSPTransportMode

The supported modes of the media.

Members
GstRtspServer.RTSPTransportMode.PLAY (1) –

Transport supports PLAY mode

GstRtspServer.RTSPTransportMode.RECORD (2) –

Transport supports RECORD mode


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