gstreamer-rs/docs/gstreamer-rtsp-server/docs.md
2021-04-20 18:18:02 +02:00

91 KiB

An address

Make a copy of self.

Returns

a copy of self.

Free self and releasing it back into the pool when owned by a pool.

Flags used to control allocation of addresses

no flags

an IPv4 address

and IPv6 address

address with an even port

a multicast address

a unicast address

An address pool, all member are private

Implements

RTSPAddressPoolExt, [trait@glib::object::ObjectExt], RTSPAddressPoolExtManual

Trait containing all RTSPAddressPool methods.

Implementors

RTSPAddressPool

Make a new RTSPAddressPool.

Returns

a new RTSPAddressPool

Take an address and ports from self. flags can be used to control the allocation. n_ports consecutive ports will be allocated of which the first one can be found in port.

flags

flags

n_ports

the amount of ports

Returns

a RTSPAddress that should be freed with gst_rtsp_address_free after use or None when no address could be acquired.

Adds the addresses from min_addess to max_address (inclusive) to self. The valid port range for the addresses will be from min_port to max_port inclusive.

When ttl is 0, min_address and max_address should be unicast addresses. min_address and max_address can be set to GST_RTSP_ADDRESS_POOL_ANY_IPV4 or GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind to all available IPv4 or IPv6 addresses.

When ttl > 0, min_address and max_address should be multicast addresses.

min_address

a minimum address to add

max_address

a maximum address to add

min_port

the minimum port

max_port

the maximum port

ttl

a TTL or 0 for unicast addresses

Returns

true if the addresses could be added.

Clear all addresses in self. There should be no outstanding allocations.

Dump the free and allocated addresses to stdout.

Used to know if the pool includes any unicast addresses.

Returns

true if the pool includes any unicast addresses, false otherwise

Take a specific address and ports from self. n_ports consecutive ports will be allocated of which the first one can be found in port.

If ttl is 0, address should be a unicast address. If ttl > 0, address should be a valid multicast address.

ip_address

The IP address to reserve

port

The first port to reserve

n_ports

The number of ports

ttl

The requested ttl

address

storage for a RTSPAddress

Returns

RTSPAddressPoolResult::Ok if an address was reserved. The address is returned in address and should be freed with gst_rtsp_address_free after use.

Result codes from RTSP address pool functions.

no error

invalid arguments were provided to a function

the addres has already been reserved

the address is not in the pool

last error

The authentication structure.

Implements

RTSPAuthExt, [trait@glib::object::ObjectExt], RTSPAuthExtManual

Trait containing all RTSPAuth methods.

Implementors

RTSPAuth

Create a new RTSPAuth instance.

Returns

a new RTSPAuth

Check if check is allowed in the current context.

check

the item to check

Returns

FALSE if check failed.

Construct a Basic authorisation token from user and pass.

user

a userid

pass

a password

Returns

the base64 encoding of the string user:pass. g_free after usage.

Add a basic token for the default authentication algorithm that enables the client with privileges listed in token.

basic

the basic token

token

authorisation token

Add a digest user and pass for the default authentication algorithm that enables the client with privileges listed in token.

Feature: v1_12

user

the digest user name

pass

the digest password

token

authorisation token

Get the default token for self. This token will be used for unauthenticated users.

Returns

the RTSPToken of self. gst_rtsp_token_unref after usage.

Feature: v1_16

Returns

the realm of self

Gets the supported authentication methods of self.

Feature: v1_12

Returns

The supported authentication methods

Get the gio::TlsAuthenticationMode.

Returns

the gio::TlsAuthenticationMode.

Get the gio::TlsCertificate used for negotiating TLS self.

Returns

the gio::TlsCertificate of self. glib::object::ObjectExt::unref after usage.

Get the gio::TlsDatabase used for verifying client certificate.

Returns

the gio::TlsDatabase of self. glib::object::ObjectExt::unref after usage.

Parse the contents of the file at path and enable the privileges listed in token for the users it describes.

The format of the file is expected to match the format described by https://en.wikipedia.org/wiki/Digest_access_authentication`The_.htdigest_file`, as output by the htdigest command.

Feature: v1_16

path

Path to the htdigest file

token

authorisation token

Returns

true if the file was successfully parsed, false otherwise.

Removes basic authentication token.

basic

the basic token

Removes a digest user.

Feature: v1_12

user

the digest user name

Set the default RTSPToken to token in self. The default token will be used for unauthenticated users.

token

a RTSPToken

Set the realm of self

Feature: v1_16

Sets the supported authentication methods for self.

Feature: v1_12

methods

supported methods

The gio::TlsAuthenticationMode to set on the underlying GTlsServerConnection. When set to another value than gio::TlsAuthenticationMode::None, RTSPAuth::accept-certificate signal will be emitted and must be handled.

mode

a gio::TlsAuthenticationMode

Set the TLS certificate for the auth. Client connections will only be accepted when TLS is negotiated.

cert

a gio::TlsCertificate

Sets the certificate database that is used to verify peer certificates. If set to None (the default), then peer certificate validation will always set the gio::TlsCertificateFlags::UnknownCa error.

database

a gio::TlsDatabase

Emitted during the TLS handshake after the client certificate has been received. See also RTSPAuthExt::set_tls_authentication_mode.

connection

a gio::TlsConnection

peer_cert

the peer's gio::TlsCertificate

errors

the problems with peer_cert.

Returns

true to accept peer_cert (which will also immediately end the signal emission). false to allow the signal emission to continue, which will cause the handshake to fail if no one else overrides it.

The client object represents the connection and its state with a client.

Implements

RTSPClientExt, [trait@glib::object::ObjectExt], RTSPClientExtManual

Trait containing all RTSPClient methods.

Implementors

RTSPClient

Create a new RTSPClient instance.

Returns

a new RTSPClient

Attaches self to context. When the mainloop for context is run, the client will be dispatched. When context is None, the default context will be used).

This function should be called when the client properties and urls are fully configured and the client is ready to start.

context

a glib::MainContext

Returns

the ID (greater than 0) for the source within the GMainContext.

Close the connection of self and remove all media it was managing.

Get the RTSPAuth used as the authentication manager of self.

Returns

the RTSPAuth of self. glib::object::ObjectExt::unref after usage.

Get the gst_rtsp::RTSPConnection of self.

Returns

the gst_rtsp::RTSPConnection of self. The connection object returned remains valid until the client is freed.

Get the Content-Length limit of self.

Feature: v1_18

Returns

the Content-Length limit.

Get the RTSPMountPoints object that self uses to manage its sessions.

Returns

a RTSPMountPoints, unref after usage.

Get the RTSPSessionPool object that self uses to manage its sessions.

Returns

a RTSPSessionPool, unref after usage.

This is useful when providing a send function through RTSPClientExt::set_send_func when doing RTSP over TCP: the send function must call gst_rtsp_stream_transport_message_sent () on the appropriate transport when data has been received for streaming to continue.

Feature: v1_18

Returns

the RTSPStreamTransport associated with channel.

Get the RTSPThreadPool used as the thread pool of self.

Returns

the RTSPThreadPool of self. glib::object::ObjectExt::unref after usage.

Let the client handle message.

message

an gst_rtsp::RTSPMessage

Returns

a gst_rtsp::RTSPResult.

Send a message message to the remote end. message must be a gst_rtsp::RTSPMsgType::Request or a gst_rtsp::RTSPMsgType::Response.

session

a RTSPSession to send the message to or None

message

The gst_rtsp::RTSPMessage to send

Call func for each session managed by self. The result value of func determines what happens to the session. func will be called with self locked so no further actions on self can be performed from func.

If func returns RTSPFilterResult::Remove, the session will be removed from self.

If func returns RTSPFilterResult::Keep, the session will remain in self.

If func returns RTSPFilterResult::Ref, the session will remain in self but will also be added with an additional ref to the result glib::List of this function..

When func is None, RTSPFilterResult::Ref will be assumed for each session.

func

a callback

user_data

user data passed to func

Returns

a glib::List with all sessions for which func returned RTSPFilterResult::Ref. After usage, each element in the glib::List should be unreffed before the list is freed.

configure auth to be used as the authentication manager of self.

auth

a RTSPAuth

Set the gst_rtsp::RTSPConnection of self. This function takes ownership of conn.

conn

a gst_rtsp::RTSPConnection

Returns

true on success.

Configure self to use the specified Content-Length limit.

Define an appropriate request size limit and reject requests exceeding the limit with response status 413 Request Entity Too Large

Feature: v1_18

limit

Content-Length limit

Set mounts as the mount points for self which it will use to map urls to media streams. These mount points are usually inherited from the server that created the client but can be overriden later.

mounts

a RTSPMountPoints

Set func as the callback that will be called when a new message needs to be sent to the client. user_data is passed to func and notify is called when user_data is no longer in use.

By default, the client will send the messages on the gst_rtsp::RTSPConnection that was configured with RTSPClient::attach was called.

It is only allowed to set either a send_func or a send_messages_func but not both at the same time.

func

a GstRTSPClientSendFunc

user_data

user data passed to func

notify

called when user_data is no longer in use

Set func as the callback that will be called when new messages needs to be sent to the client. user_data is passed to func and notify is called when user_data is no longer in use.

By default, the client will send the messages on the gst_rtsp::RTSPConnection that was configured with RTSPClient::attach was called.

It is only allowed to set either a send_func or a send_messages_func but not both at the same time.

Feature: v1_16

func

a GstRTSPClientSendMessagesFunc

user_data

user data passed to func

notify

called when user_data is no longer in use

Set pool as the sessionpool for self which it will use to find or allocate sessions. the sessionpool is usually inherited from the server that created the client but can be overridden later.

pool

a RTSPSessionPool

configure pool to be used as the thread pool of self.

pool

a RTSPThreadPool

ctx

a RTSPContext

ctx

a RTSPContext

arr

a NULL-terminated array of strings

Returns

a newly allocated string with comma-separated list of unsupported options. An empty string must be returned if all options are supported.

ctx

a RTSPContext

ctx

a RTSPContext

ctx

a RTSPContext

ctx

a RTSPContext

ctx

a RTSPContext

ctx

a RTSPContext

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

Feature: v1_12

ctx

a RTSPContext

Returns

a gst_rtsp::RTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code

ctx

a RTSPContext

session

The session

message

The message

ctx

a RTSPContext

ctx

a RTSPContext

ctx

a RTSPContext

Information passed around containing the context of a request.

Pops self off the context stack (verifying that self is on the top of the stack).

Pushes self onto the context stack. The current context can then be received using RTSPContext::get_current.

Get the current RTSPContext. This object is retrieved from the current thread that is handling the request for a client.

Returns

a RTSPContext

Possible return values for RTSPSessionPoolExt::filter.

Remove session

Keep session in the pool

Ref session in the result list

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

This object is usually created from a RTSPMediaFactory.

Implements

RTSPMediaExt, [trait@glib::object::ObjectExt], RTSPMediaExtManual

Trait containing all RTSPMedia methods.

Implementors

RTSPMedia

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

Ownership is taken of element.

element

a gst::Element

Returns

a new RTSPMedia object.

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

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

Feature: v1_14

transports

a list of gst_rtsp::RTSPTransport

Returns

true if the media pipeline has been sucessfully updated.

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

payloader

a gst::Element

pad

a gst::Pad

Returns

a new RTSPStream that remains valid for as long as self exists.

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

control

the control of the stream

Returns

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

Get the RTSPAddressPool used as the address pool of self.

Returns

the RTSPAddressPool of self. glib::object::ObjectExt::unref after usage.

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

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

Returns

the base_time used by self.

Get the kernel UDP buffer size.

Returns

the kernel UDP buffer size.

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

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

Returns

the gst::Clock used by self. unref after usage.

Feature: v1_16

Returns

Whether retransmission requests will be sent

Get the configured DSCP QoS of attached media.

Feature: v1_18

Returns

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

Get the element that was used when constructing self.

Returns

a gst::Element. Unref after usage.

Get the latency that is used for receiving media.

Returns

latency in milliseconds

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

Feature: v1_16

Returns

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

Get the multicast interface used for self.

Returns

the multicast interface for self. g_free after usage.

Get the permissions object from self.

Returns

a RTSPPermissions object, unref after usage.

Get the allowed profiles of self.

Returns

a gst_rtsp::RTSPProfile

Get the allowed protocols of self.

Returns

a gst_rtsp::RTSPLowerTrans

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

Returns

The GstRTSPPublishClockMode

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

play

for the PLAY request

unit

the unit to use for the string

Returns

The range as a string, g_free after usage.

Feature: v1_18

Returns

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

Get the rate and applied_rate of the current segment.

Feature: v1_18

rate

the rate of the current segment

applied_rate

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.

Get the amount of time to store retransmission data.

Returns

the amount of time to store retransmission data.

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

Returns

the status of self.

Retrieve the stream with index idx from self.

idx

the stream index

Returns

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

Get how self will be suspended.

Returns

RTSPSuspendMode.

Get the gst_net::NetTimeProvider for the clock used by self. The time provider will listen on address and port for client time requests.

address

an address or None

port

a port or 0

Returns

the gst_net::NetTimeProvider of self.

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

Returns

The transport mode.

Configure an SDP on self for receiving streams

sdp

a gst_sdp::SDPMessage

Returns

TRUE on success.

See RTSPStreamExt::is_complete, RTSPStreamExt::is_sender.

Feature: v1_18

Returns

whether self has at least one complete sender stream.

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

Feature: v1_16

Returns

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

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

Returns

true if the media will send EOS before unpreparing.

Feature: v1_18

Returns

true if self is receive-only, false otherwise.

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

Returns

true if the media can be reused

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

Returns

true if the media can be shared between clients.

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

Returns

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

Check if self can provide a gst_net::NetTimeProvider for its pipeline clock.

Use RTSPMediaExt::get_time_provider to get the network clock.

Returns

true if self can provide a gst_net::NetTimeProvider.

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.

Feature: v1_18

Get the number of streams in this media.

Returns

The number of streams.

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

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

thread

a RTSPThread to run the bus handler or None

Returns

true on success.

Seek the pipeline of self to range. self must be prepared with RTSPMediaExt::prepare.

range

a gst_rtsp::RTSPTimeRange

Returns

true on success.

Seek the pipeline of self to range with the given flags. self must be prepared with RTSPMediaExt::prepare.

Feature: v1_18

range

a gst_rtsp::RTSPTimeRange

flags

The minimal set of gst::SeekFlags to use

Returns

true on success.

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

Feature: v1_18

range

a gst_rtsp::RTSPTimeRange

flags

The minimal set of gst::SeekFlags 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.

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

Feature: v1_14

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.

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

pool

a RTSPAddressPool

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

Feature: v1_16

bind_mcast_addr

the new value

Set the kernel UDP buffer size.

size

the new value

Configure the clock used for the media.

clock

gst::Clock to be used

Set whether retransmission requests will be sent

Feature: v1_16

Configure the dscp qos of attached streams to dscp_qos.

Feature: v1_18

dscp_qos

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

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

eos_shutdown

the new value

Configure the latency used for receiving media.

latency

latency in milliseconds

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

Feature: v1_16

ttl

the new multicast ttl value

Returns

true if the requested ttl has been set successfully.

configure multicast_iface to be used for self.

multicast_iface

a multicast interface name

Set permissions on self.

permissions

a RTSPPermissions

Set the state of the pipeline managed by self to state

state

the target state of the pipeline

Configure the allowed lower transport for self.

profiles

the new flags

Configure the allowed lower transport for self.

protocols

the new flags

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

mode

the clock publish mode

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

Feature: v1_18

Set the amount of time to store retransmission packets.

time

the new value

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

reusable

the new value

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

shared

the new value

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

self must be prepared with RTSPMediaExt::prepare;

state

the target state of the media

transports

a glib::PtrArray of RTSPStreamTransport pointers

Returns

true on success.

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

stop_on_disconnect

the new value

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.

mode

the new RTSPSuspendMode

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

mode

the new value

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

sdp

a gst_sdp::SDPMessage

info

a SDPInfo

Returns

TRUE on success.

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

self must be prepared with RTSPMediaExt::prepare;

Returns

true on success.

Set pipeline as the gst::Pipeline for self. Ownership is taken of pipeline.

pipeline

a gst::Pipeline

Unlock the media.

Feature: v1_18

Unprepare self. 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

true on success.

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

Returns

true on success.

Set self to provide a gst_net::NetTimeProvider.

time_provider

if a gst_net::NetTimeProvider should be used

The definition and logic for constructing the pipeline for a media. The media can contain multiple streams like audio and video.

Implements

RTSPMediaFactoryExt, [trait@glib::object::ObjectExt], RTSPMediaFactoryExtManual

Trait containing all RTSPMediaFactory methods.

Implementors

RTSPMediaFactoryURI, RTSPMediaFactory

Create a new RTSPMediaFactory instance.

Returns

a new RTSPMediaFactory object.

A convenience method to add role with fieldname and additional arguments to the permissions of self. If self had no permissions, new permissions will be created and the role will be added to it.

role

a role

fieldname

the first field name

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

Feature: v1_14

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

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

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

url

the url used

Returns

a new RTSPMedia if the media could be prepared.

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

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

url

the url used

Returns

a new gst::Element.

Get the RTSPAddressPool used as the address pool of self.

Returns

the RTSPAddressPool of self. glib::object::ObjectExt::unref after usage.

Get the kernel UDP buffer size.

Returns

the kernel UDP buffer size.

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

Returns

The GstClock

Feature: v1_16

Returns

Whether retransmission requests will be sent for receiving media

Get the configured media DSCP QoS.

Feature: v1_18

Returns

the media DSCP QoS value or -1 if disabled.

Get the latency that is used for receiving media

Returns

latency in milliseconds

Get the gst_parse_launch pipeline description that will be used in the default prepare vmethod.

Returns

the configured launch description. g_free after usage.

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

Feature: v1_16

Returns

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

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

Get the multicast interface used for self.

Returns

the multicast interface for self. g_free after usage.

Get the permissions object from self.

Returns

a RTSPPermissions object, unref after usage.

Get the allowed profiles of self.

Returns

a gst_rtsp::RTSPProfile

Get the allowed protocols of self.

Returns

a gst_rtsp::RTSPLowerTrans

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

Returns

The GstRTSPPublishClockMode

Get the time that is stored for retransmission purposes

Returns

a gst::ClockTime

Get how media created from this factory will be suspended.

Returns

a RTSPSuspendMode.

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

Returns

The transport mode.

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

Feature: v1_16

Returns

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

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

Returns

true if the media will receive EOS before shutdown.

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

Returns

true if the media will be shared between clients.

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

pool

a RTSPAddressPool

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

Feature: v1_16

bind_mcast_addr

the new value

Set the kernel UDP buffer size.

size

the new value

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

clock

the clock to be used by the media factory

Set whether retransmission requests will be sent for receiving media

Feature: v1_16

Configure the media dscp qos to dscp_qos.

Feature: v1_18

dscp_qos

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

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

eos_shutdown

the new value

Configure the latency used for receiving media

latency

latency in milliseconds

The gst_parse_launch line to use for constructing the pipeline in the default prepare vmethod.

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

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

launch

the launch description

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

Feature: v1_16

ttl

the new multicast ttl value

Returns

true if the requested ttl has been set successfully.

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

media_gtype

the GType of the class to create

configure multicast_iface to be used for self.

multicast_iface

a multicast interface name

Set permissions on self.

permissions

a RTSPPermissions

Configure the allowed profiles for self.

profiles

the new flags

Configure the allowed lower transport for self.

protocols

the new flags

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

mode

the clock publish mode

Configure the time to store for possible retransmission

time

a gst::ClockTime

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

shared

the new value

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

stop_on_disconnect

the new value

Configure how media created from this factory will be suspended.

mode

the new RTSPSuspendMode

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

mode

the new value

A media factory that creates a pipeline to play any uri.

Implements

RTSPMediaFactoryURIExt, RTSPMediaFactoryExt, [trait@glib::object::ObjectExt], RTSPMediaFactoryExtManual

Trait containing all RTSPMediaFactoryURI methods.

Implementors

RTSPMediaFactoryURI

Create a new RTSPMediaFactoryURI instance.

Returns

a new RTSPMediaFactoryURI object.

Get the URI that will provide media for this factory.

Returns

the configured URI. g_free after usage.

Set the URI of the resource that will be streamed by this factory.

uri

the uri the stream

The state of the media pipeline.

media pipeline not prerolled

media pipeline is busy doing a clean shutdown.

media pipeline is prerolling

media pipeline is prerolled

media is suspended

media pipeline is in error

Creates a RTSPMediaFactory object for a given url.

Implements

RTSPMountPointsExt, [trait@glib::object::ObjectExt]

Trait containing all RTSPMountPoints methods.

Implementors

RTSPMountPoints

Make a new mount points object.

Returns

a new RTSPMountPoints

Attach factory to the mount point path in self.

path is of the form (/node)+. Any previous mount point will be freed.

Ownership is taken of the reference on factory so that factory should not be used after calling this function.

path

a mount point

factory

a RTSPMediaFactory

Make a path string from url.

url

a gst_rtsp::RTSPUrl

Returns

a path string for url, g_free after usage.

Find the factory in self that has the longest match with path.

If matched is None, path will match the factory exactly otherwise the amount of characters that matched is returned in matched.

path

a mount point

matched

the amount of path matched

Returns

the RTSPMediaFactory for path. glib::object::ObjectExt::unref after usage.

Remove the RTSPMediaFactory associated with path in self.

path

a mount point

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

Publish nothing

Publish the clock but not the offset

Publish the clock and offset

This object listens on a port, creates and manages the clients connected to it.

Implements

RTSPServerExt, [trait@glib::object::ObjectExt], RTSPServerExtManual

Trait containing all RTSPServer methods.

Implementors

RTSPServer

Create a new RTSPServer instance.

Returns

a new RTSPServer

A default GSocketSourceFunc that creates a new RTSPClient to accept and handle a new connection on socket or server.

socket

a gio::Socket

condition

the condition on source

server

a RTSPServer

Returns

TRUE if the source could be connected, FALSE if an error occurred.

Attaches self to context. When the mainloop for context is run, the server will be dispatched. When context is None, the default context will be used).

This function should be called when the server properties and urls are fully configured and the server is ready to start.

This takes a reference on self until the source is destroyed. Note that if context is not the default main context as returned by glib::MainContext::default (or None), glib::Source::remove cannot be used to destroy the source. In that case it is recommended to use RTSPServerExt::create_source and attach it to context manually.

context

a glib::MainContext

Returns

the ID (greater than 0) for the source within the GMainContext.

Call func for each client managed by self. The result value of func determines what happens to the client. func will be called with self locked so no further actions on self can be performed from func.

If func returns RTSPFilterResult::Remove, the client will be removed from self.

If func returns RTSPFilterResult::Keep, the client will remain in self.

If func returns RTSPFilterResult::Ref, the client will remain in self but will also be added with an additional ref to the result glib::List of this function..

When func is None, RTSPFilterResult::Ref will be assumed for each client.

func

a callback

user_data

user data passed to func

Returns

a glib::List with all clients for which func returned RTSPFilterResult::Ref. After usage, each element in the glib::List should be unreffed before the list is freed.

Create a gio::Socket for self. The socket will listen on the configured service.

cancellable

a gio::Cancellable

Returns

the gio::Socket for self or None when an error occurred.

Create a glib::Source for self. The new source will have a default GSocketSourceFunc of RTSPServer::io_func.

cancellable if not None can be used to cancel the source, which will cause the source to trigger, reporting the current condition (which is likely 0 unless cancellation happened at the same time as a condition change). You can check for this in the callback using gio::CancellableExt::is_cancelled.

This takes a reference on self until source is destroyed.

cancellable

a gio::Cancellable or None.

Returns

the glib::Source for self or None when an error occurred. Free with g_source_unref ()

Get the address on which the server will accept connections.

Returns

the server address. g_free after usage.

Get the RTSPAuth used as the authentication manager of self.

Returns

the RTSPAuth of self. glib::object::ObjectExt::unref after usage.

The maximum amount of queued requests for the server.

Returns

the server backlog.

Get the port number where the server was bound to.

Returns

the port number

Get the Content-Length limit of self.

Feature: v1_18

Returns

the Content-Length limit.

Get the RTSPMountPoints used as the mount points of self.

Returns

the RTSPMountPoints of self. glib::object::ObjectExt::unref after usage.

Get the service on which the server will accept connections.

Returns

the service. use g_free after usage.

Get the RTSPSessionPool used as the session pool of self.

Returns

the RTSPSessionPool used for sessions. glib::object::ObjectExt::unref after usage.

Get the RTSPThreadPool used as the thread pool of self.

Returns

the RTSPThreadPool of self. glib::object::ObjectExt::unref after usage.

Configure self to accept connections on the given address.

This function must be called before the server is bound.

address

the address

configure auth to be used as the authentication manager of self.

auth

a RTSPAuth

configure the maximum amount of requests that may be queued for the server.

This function must be called before the server is bound.

backlog

the backlog

Define an appropriate request size limit and reject requests exceeding the limit.

Feature: v1_18

configure mounts to be used as the mount points of self.

mounts

a RTSPMountPoints

Configure self to accept connections on the given service. service should be a string containing the service name (see services(5)) or a string containing a port number between 1 and 65535.

When service is set to "0", the server will listen on a random free port. The actual used port can be retrieved with RTSPServerExt::get_bound_port.

This function must be called before the server is bound.

service

the service

configure pool to be used as the session pool of self.

pool

a RTSPSessionPool

configure pool to be used as the thread pool of self.

pool

a RTSPThreadPool

Take an existing network socket and use it for an RTSP connection. This is used when transferring a socket from an HTTP server which should be used as an RTSP over HTTP tunnel. The initial_buffer contains any remaining data that the HTTP server read from the socket while parsing the HTTP header.

socket

a network socket

ip

the IP address of the remote client

port

the port used by the other end

initial_buffer

any initial data that was already read from the socket

Returns

TRUE if all was ok, FALSE if an error occurred.

Session information kept by the server for a specific client. One client session, identified with a session id, can handle multiple medias identified with the url of a media.

Implements

RTSPSessionExt, [trait@glib::object::ObjectExt]

Trait containing all RTSPSession methods.

Implementors

RTSPSession

Create a new RTSPSession instance with sessionid.

sessionid

a session id

Returns

a new RTSPSession

Allow self to expire. This method must be called an equal amount of time as RTSPSessionExt::prevent_expire.

Call func for each media in self. The result value of func determines what happens to the media. func will be called with self locked so no further actions on self can be performed from func.

If func returns RTSPFilterResult::Remove, the media will be removed from self.

If func returns RTSPFilterResult::Keep, the media will remain in self.

If func returns RTSPFilterResult::Ref, the media will remain in self but will also be added with an additional ref to the result glib::List of this function..

When func is None, RTSPFilterResult::Ref will be assumed for all media.

func

a callback

user_data

user data passed to func

Returns

a GList with all media for which func returned RTSPFilterResult::Ref. After usage, each element in the glib::List should be unreffed before the list is freed.

Get the string that can be placed in the Session header field.

Returns

the Session header of self. g_free after usage.

Get the session media for path. matched will contain the number of matched characters of path.

path

the path for the media

matched

the amount of matched characters

Returns

the configuration for path in self.

Get the sessionid of self.

Returns

the sessionid of self. The value remains valid as long as self is alive.

Get the timeout value of self.

Returns

the timeout of self in seconds.

Check if self timeout out.

Deprecated

Use RTSPSessionExt::is_expired_usec instead.

now

the current system time

Returns

true if self timed out

Check if self timeout out.

now

the current monotonic time

Returns

true if self timed out

Manage the media object obj in self. path will be used to retrieve this media from the session with RTSPSessionExt::get_media.

Ownership is taken from media.

path

the path for the media

media

a RTSPMedia

Returns

a new [crate::RTSPSessionMedia] (XXX: @-reference does not belong to RTSPSessionExt!) object.

Get the amount of milliseconds till the session will expire.

Deprecated

Use RTSPSessionExt::next_timeout_usec instead.

now

the current system time

Returns

the amount of milliseconds since the session will time out.

Get the amount of milliseconds till the session will expire.

now

the current monotonic time

Returns

the amount of milliseconds since the session will time out.

Prevent self from expiring.

Release the managed media in self, freeing the memory allocated by it.

media

a RTSPMedia

Returns

true if there are more media session left in self.

Configure self for a timeout of timeout seconds. The session will be cleaned up when there is no activity for timeout seconds.

timeout

the new timeout

Update the last_access time of the session to the current time.

State of a client session regarding a specific media identified by path.

Implements

RTSPSessionMediaExt, [trait@glib::object::ObjectExt]

Trait containing all RTSPSessionMedia methods.

Implementors

RTSPSessionMedia

Create a new RTSPSessionMedia that manages the streams in media for path. media should be prepared.

Ownership is taken of media.

path

the path

media

the RTSPMedia

Returns

a new RTSPSessionMedia.

Fill range with the next available min and max channels for interleaved transport.

range

a gst_rtsp::RTSPRange

Returns

true on success.

Get the base_time of the RTSPMedia in self

Returns

the base_time of the media.

Get the RTSPMedia that was used when constructing self

Returns

the RTSPMedia of self. Remains valid as long as self is valid.

Retrieve the RTP-Info header string for all streams in self with configured transports.

Returns

The RTP-Info as a string or None when no RTP-Info could be generated, g_free after usage.

Get the current RTSP state of self.

Returns

the current RTSP state of self.

Get a previously created RTSPStreamTransport for the stream at idx.

idx

the stream index

Returns

a RTSPStreamTransport that is valid until the session of self is unreffed.

Get a list of all available RTSPStreamTransport in this session.

Feature: v1_14

Returns

a list of RTSPStreamTransport, g_ptr_array_unref () after usage.

Check if the path of self matches path. It path matches, the amount of matched characters is returned in matched.

path

a path

matched

the amount of matched characters of path

Returns

true when path matches the path of self.

Set the RTSP state of self to state.

state

a gst_rtsp::RTSPState

Tell the media object self to change to state.

state

the new state

Returns

true on success.

Configure the transport for stream to tr in self.

stream

a RTSPStream

tr

a gst_rtsp::RTSPTransport

Returns

the new or updated RTSPStreamTransport for stream.

An object that keeps track of the active sessions. This object is usually attached to a RTSPServer object to manage the sessions in that server.

Implements

RTSPSessionPoolExt, [trait@glib::object::ObjectExt], RTSPSessionPoolExtManual

Trait containing all RTSPSessionPool methods.

Implementors

RTSPSessionPool

Create a new RTSPSessionPool instance.

Returns

A new RTSPSessionPool. glib::object::ObjectExt::unref after usage.

Inspect all the sessions in self and remove the sessions that are inactive for more than their timeout.

Returns

the amount of sessions that got removed.

Create a new RTSPSession object in self.

Returns

a new RTSPSession.

Create a glib::Source that will be dispatched when the session should be cleaned up.

Returns

a glib::Source

Call func for each session in self. The result value of func determines what happens to the session. func will be called with the session pool locked so no further actions on self can be performed from func.

If func returns RTSPFilterResult::Remove, the session will be set to the expired state and removed from self.

If func returns RTSPFilterResult::Keep, the session will remain in self.

If func returns RTSPFilterResult::Ref, the session will remain in self but will also be added with an additional ref to the result GList of this function..

When func is None, RTSPFilterResult::Ref will be assumed for all sessions.

func

a callback

user_data

user data passed to func

Returns

a GList with all sessions for which func returned RTSPFilterResult::Ref. After usage, each element in the GList should be unreffed before the list is freed.

Find the session with sessionid in self. The access time of the session will be updated with RTSPSessionExt::touch.

sessionid

the session id

Returns

the RTSPSession with sessionid or None when the session did not exist. glib::object::ObjectExt::unref after usage.

Get the maximum allowed number of sessions in self. 0 means an unlimited amount of sessions.

Returns

the maximum allowed number of sessions.

Get the amount of active sessions in self.

Returns

the amount of active sessions in self.

Remove sess from self, releasing the ref that the pool has on sess.

sess

a RTSPSession

Returns

true if the session was found and removed.

Configure the maximum allowed number of sessions in self to max. A value of 0 means an unlimited amount of sessions.

max

the maximum number of sessions

The definition of a media stream.

Implements

RTSPStreamExt, [trait@glib::object::ObjectExt], RTSPStreamExtManual

Trait containing all RTSPStream methods.

Implementors

RTSPStream

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

idx

an index

payloader

a gst::Element

pad

a gst::Pad

Returns

a new RTSPStream

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

Feature: v1_16

destination

a multicast address to add

rtp_port

RTP port

rtcp_port

RTCP port

family

socket family

Returns

true if destination can be addedd and handled by self.

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

self must be joined to a bin.

trans must contain a valid gst_rtsp::RTSPTransport.

trans

a RTSPStreamTransport

Returns

true if trans was added

Allocates RTP and RTCP ports.

family

protocol family

transport

transport method

use_client_settings

Whether to use client settings or not

Returns

true if the RTP and RTCP sockets have been succeccully allocated.

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

Feature: v1_14

transport

a gst_rtsp::RTSPTransport

Returns

true if the stream has been sucessfully updated.

Get the RTSPAddressPool used as the address pool of self.

Returns

the RTSPAddressPool of self. glib::object::ObjectExt::unref after usage.

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

Returns

the size of the UDP TX buffer

Retrieve the current caps of self.

Returns

the gst::Caps of self. use gst_caps_unref after usage.

Get the control string to identify this stream.

Returns

the control string. g_free after usage.

Get the configured DSCP QoS in of the outgoing sockets.

Returns

the DSCP QoS value of the outgoing sockets, or -1 if disbled.

Get the stream index.

Returns

the stream index.

Get the previous joined bin with RTSPStreamExt::join_bin or NULL.

Returns

the joined bin or NULL.

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

Feature: v1_16

Returns

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

Get the configured MTU in the payloader of self.

Returns

the MTU of the payloader.

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

family

the gio::SocketFamily

Returns

the RTSPAddress of self or None when no address could be allocated. RTSPAddress::free after usage.

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

Feature: v1_16

Returns

A comma separated list of host:port pairs with destinations

Get the multicast interface used for self.

Returns

the multicast interface for self. g_free after usage.

Get the allowed profiles of self.

Returns

a gst_rtsp::RTSPProfile

Get the allowed protocols of self.

Returns

a gst_rtsp::RTSPLowerTrans

Get the stream payload type.

Returns

the stream payload type.

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

Returns

The GstRTSPPublishClockMode

Feature: v1_18

Returns

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

Retrieve the current rate and/or applied_rate.

Feature: v1_18

rate

the configured rate

applied_rate

the configured applied_rate

Returns

true if rate and/or applied_rate could be determined.

Get the payload-type used for retransmission of this stream

Returns

The retransmission PT.

Get the amount of time to store retransmission data.

Returns

the amount of time to store retransmission data.

Get the multicast RTCP socket from self for a family.

Feature: v1_14

family

the socket family

Returns

the multicast RTCP socket or None if no socket could be allocated for family. Unref after usage

Get the RTCP socket from self for a family.

self must be joined to a bin.

family

the socket family

Returns

the RTCP socket or None if no socket could be allocated for family. Unref after usage

Get the multicast RTP socket from self for a family.

family

the socket family

Returns

the multicast RTP socket or None if no

socket could be allocated for family. Unref after usage

Get the RTP socket from self for a family.

self must be joined to a bin.

family

the socket family

Returns

the RTP socket or None if no socket could be allocated for family. Unref after usage

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

rtptime

result RTP timestamp

seq

result RTP seqnum

clock_rate

the clock rate

running_time

result running-time

Returns

true when rtptime, seq and running-time could be determined.

Get the RTP session of this stream.

Returns

The RTP session of this stream. Unref after usage.

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

server_port

result server port

family

the port family to get

Get the sinkpad associated with self.

Returns

the sinkpad. Unref after usage.

Get the srcpad associated with self.

Returns

the srcpad. Unref after usage.

Get the SRTP encoder for this stream.

Returns

The SRTP encoder for this stream. Unref after usage.

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

ssrc

result ssrc

Feature: v1_16

Returns

the amount of redundancy applied when creating ULPFEC protection packets.

Feature: v1_16

Returns

the payload type used for ULPFEC protection packets

Parse and handle a KeyMgmt header.

Feature: v1_16

keymgmt

a keymgmt header

Check if self has the control string control.

control

a control string

Returns

true is self has control as the control string

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

Feature: v1_16

Returns

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

Check if self is blocking on a gst::Buffer.

Returns

true if self is blocking

See RTSPStreamExt::set_client_side

Returns

TRUE if this RTSPStream is client-side.

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

Feature: v1_14

Returns

true if the stream contains at least one sink element.

Checks whether the stream is a receiver.

Feature: v1_14

Returns

true if the stream is a receiver and false otherwise.

Checks whether the stream is a sender.

Feature: v1_14

Returns

true if the stream is a sender and false otherwise.

Check if transport can be handled by stream

transport

a gst_rtsp::RTSPTransport

Returns

true if transport can be handled by self.

Join the gst::Bin bin that contains the element rtpbin.

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

bin

a gst::Bin to join

rtpbin

a rtpbin element in bin

state

the target state of the new elements

Returns

true on success.

Remove the elements of self from bin.

bin

a gst::Bin

rtpbin

a rtpbin gst::Element

Returns

true on success.

Query the position of the stream in gst::Format::Time. This only considers the RTP parts of the pipeline and not the RTCP parts.

position

current position of a RTSPStream

Returns

true if the position could be queried

Query the stop of the stream in gst::Format::Time. This only considers the RTP parts of the pipeline and not the RTCP parts.

stop

current stop of a RTSPStream

Returns

true if the stop could be queried

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

This function takes ownership of buffer.

buffer

a gst::Buffer

Returns

a GstFlowReturn.

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

This function takes ownership of buffer.

buffer

a gst::Buffer

Returns

a GstFlowReturn.

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

self must be joined to a bin.

trans must contain a valid gst_rtsp::RTSPTransport.

trans

a RTSPStreamTransport

Returns

true if trans was removed

Creating a rtxreceive bin

Feature: v1_16

sessid

the session id

Returns

a gst::Element.

Creating a rtxsend bin

sessid

the session id

Returns

a gst::Element.

Creating a rtpulpfecdec element

Feature: v1_16

Returns

a gst::Element.

Creating a rtpulpfecenc element

Feature: v1_16

Returns

a gst::Element.

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

address

an address

port

a port

n_ports

n_ports

ttl

a TTL

Returns

the RTSPAddress of self or None when the address could not be reserved. RTSPAddress::free after usage.

Checks whether the individual self is seekable.

Feature: v1_14

Returns

true if self is seekable, else false.

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

pool

a RTSPAddressPool

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

Feature: v1_16

bind_mcast_addr

the new value

Blocks or unblocks the dataflow on self.

blocked

boolean indicating we should block or unblock

Returns

true on success

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

size

the buffer size

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

client_side

TRUE if this RTSPStream is running on the 'client' side of an RTSP connection.

Set the control string in self.

control

a control string

Configure the dscp qos of the outgoing sockets to dscp_qos.

dscp_qos

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

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

Feature: v1_16

ttl

the new multicast ttl value

Returns

true if the requested ttl has been set successfully.

Configure the mtu in the payloader of self to mtu.

mtu

a new MTU

configure multicast_iface to be used for self.

multicast_iface

a multicast interface name

Configure the allowed profiles for self.

profiles

the new profiles

Configure the allowed lower transport for self.

protocols

the new flags

Configure a pt map between pt and caps.

pt

the pt

caps

a gst::Caps

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

mode

the clock publish mode

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

Feature: v1_18

Set the payload type (pt) for retransmission of this stream.

rtx_pt

a guint

Set the amount of time to store retransmission packets.

time

a gst::ClockTime

Sets the amount of redundancy to apply when creating ULPFEC protection packets.

Feature: v1_16

Set the payload type to be used for ULPFEC protection packets

Feature: v1_16

Call func for each transport managed by self. The result value of func determines what happens to the transport. func will be called with self locked so no further actions on self can be performed from func.

If func returns RTSPFilterResult::Remove, the transport will be removed from self.

If func returns RTSPFilterResult::Keep, the transport will remain in self.

If func returns RTSPFilterResult::Ref, the transport will remain in self but will also be added with an additional ref to the result glib::List of this function..

When func is None, RTSPFilterResult::Ref will be assumed for each transport.

func

a callback

user_data

user data passed to func

Returns

a glib::List with all transports for which func returned RTSPFilterResult::Ref. After usage, each element in the glib::List should be unreffed before the list is freed.

Update the new crypto information for ssrc in self. If information for ssrc did not exist, it will be added. If information for ssrc existed, it will be replaced. If crypto is None, it will be removed from self.

ssrc

the SSRC

crypto

a gst::Caps with crypto info

Returns

true if crypto could be updated

Check if the requested multicast ttl value is allowed.

Feature: v1_16

ttl

a requested multicast ttl

Returns

TRUE if the requested ttl value is allowed.

A Transport description for a stream

Implements

RTSPStreamTransportExt, [trait@glib::object::ObjectExt], RTSPStreamTransportExtManual

Trait containing all RTSPStreamTransport methods.

Implementors

RTSPStreamTransport

Create a new RTSPStreamTransport that can be used to manage stream with transport tr.

stream

a RTSPStream

tr

a GstRTSPTransport

Returns

a new RTSPStreamTransport

Get the RTP-Info string for self and start_time.

start_time

a star time

Returns

the RTPInfo string for self and start_time or None when the RTP-Info could not be determined. g_free after usage.

Get the RTSPStream used when constructing self.

Returns

the stream used when constructing self.

Get the transport configured in self.

Returns

the transport configured in self. It remains valid for as long as self is valid.

Get the url configured in self.

Returns

the url configured in self. It remains valid for as long as self is valid.

Check if self is timed out.

Returns

true if self timed out.

Signal the installed keep_alive callback for self.

Signal the installed message_sent / message_sent_full callback for self.

Feature: v1_16

Receive buffer on channel self.

channel

a channel

buffer

a gst::Buffer

Returns

a gst::FlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of self.

Send buffer to the installed RTCP callback for self.

buffer

a gst::Buffer

Returns

true on success

Send buffer_list to the installed RTCP callback for self.

Feature: v1_16

buffer_list

a gst::Buffer

Returns

true on success

Send buffer to the installed RTP callback for self.

buffer

a gst::Buffer

Returns

true on success

Send buffer_list to the installed RTP callback for self.

Feature: v1_16

buffer_list

a gst::BufferList

Returns

true on success

Activate or deactivate datatransfer configured in self.

active

new state of self

Returns

true when the state was changed.

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

send_rtp

a callback called when RTP should be sent

send_rtcp

a callback called when RTCP should be sent

user_data

user data passed to callbacks

notify

called with the user_data when no longer needed.

Install callbacks that will be called when RTCP packets are received from the receiver of self.

keep_alive

a callback called when the receiver is active

user_data

user data passed to callback

notify

called with the user_data when no longer needed.

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Feature: v1_16

send_rtp_list

a callback called when RTP should be sent

send_rtcp_list

a callback called when RTCP should be sent

user_data

user data passed to callbacks

notify

called with the user_data when no longer needed.

Install a callback that will be called when a message has been sent on self.

message_sent

a callback called when a message has been sent

user_data

user data passed to callback

notify

called with the user_data when no longer needed

Install a callback that will be called when a message has been sent on self.

Feature: v1_18

message_sent

a callback called when a message has been sent

user_data

user data passed to callback

notify

called with the user_data when no longer needed

Set the timed out state of self to timedout

timedout

timed out value

Set tr as the client transport. This function takes ownership of the passed tr.

tr

a client gst_rtsp::RTSPTransport

Set url as the client url.

url

a client gst_rtsp::RTSPUrl

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.

Media is not suspended

Media is PAUSED in suspend

The media is set to NULL when suspended

Structure holding info about a mainloop running in a thread

Create a new thread object that can run a mainloop.

type_

the thread type

Returns

a RTSPThread.

Reuse the mainloop of self

Returns

true if the mainloop could be reused

Stop and unref self. When no threads are using the mainloop, the thread will be stopped and the final ref to self will be released.

The thread pool structure.

Implements

RTSPThreadPoolExt, [trait@glib::object::ObjectExt]

Trait containing all RTSPThreadPool methods.

Implementors

RTSPThreadPool

Create a new RTSPThreadPool instance.

Returns

a new RTSPThreadPool

Wait for all tasks to be stopped and free all allocated resources. This is mainly used in test suites to ensure proper cleanup of internal data structures.

Get the maximum number of threads used for client connections. See RTSPThreadPoolExt::set_max_threads.

Returns

the maximum number of threads.

Get a new RTSPThread for type_ and ctx.

type_

the RTSPThreadType

ctx

a RTSPContext

Returns

a new RTSPThread, RTSPThread::stop after usage

Set the maximum threads used by the pool to handle client requests. A value of 0 will use the pool mainloop, a value of -1 will use an unlimited number of threads.

max_threads

maximum threads

Different thread types

a thread to handle the client communication

a thread to handle media

An opaque object used for checking authorisations. It is generated after successful authentication.

Create a new Authorization token with the given fieldnames and values. Arguments are given similar to gst::Structure::new.

firstfield

the first fieldname

Returns

a new authorization token.

Create a new empty Authorization token.

Returns

a new empty authorization token.

Create a new Authorization token with the given fieldnames and values. Arguments are given similar to gst::Structure::new_valist.

firstfield

the first fieldname

var_args

additional arguments

Returns

a new authorization token.

Get the string value of field in self.

field

a field name

Returns

the string value of field in self or None when field is not defined in self. The string becomes invalid when you free self.

Access the structure of the token.

Returns

The structure of the token. The structure is still owned by the token, which means that you should not free it and that the pointer becomes invalid when you free the token.

MT safe.

Check if self has a boolean field and if it is set to true.

field

a field name

Returns

true if self has a boolean field named field set to true.

Sets a boolean value on self.

Feature: v1_14

field

field to set

bool_value

boolean value to set

Sets a string value on self.

Feature: v1_14

field

field to set

string_value

string value to set

Get a writable version of the structure.

Returns

The structure of the token. The structure is still owned by the token, which means that you should not free it and that the pointer becomes invalid when you free the token. This function checks if self is writable and will never return None.

MT safe.

The supported modes of the media.

Transport supports PLAY mode

Transport supports RECORD mode