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