forked from mirrors/gstreamer-rs
3321 lines
90 KiB
Markdown
3321 lines
90 KiB
Markdown
<!-- file * -->
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<!-- struct RTSPAddress -->
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An address
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<!-- impl RTSPAddress::fn copy -->
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Make a copy of `self`.
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# Returns
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a copy of `self`.
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<!-- impl RTSPAddress::fn free -->
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Free `self` and releasing it back into the pool when owned by a
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pool.
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<!-- struct RTSPAddressPool -->
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An address pool, all member are private
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# Implements
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[`RTSPAddressPoolExt`](trait.RTSPAddressPoolExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPAddressPoolExtManual`](prelude/trait.RTSPAddressPoolExtManual.html)
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<!-- trait RTSPAddressPoolExt -->
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Trait containing all `RTSPAddressPool` methods.
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# Implementors
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[`RTSPAddressPool`](struct.RTSPAddressPool.html)
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<!-- impl RTSPAddressPool::fn new -->
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Make a new `RTSPAddressPool`.
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# Returns
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a new `RTSPAddressPool`
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<!-- trait RTSPAddressPoolExt::fn acquire_address -->
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Take an address and ports from `self`. `flags` can be used to control the
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allocation. `n_ports` consecutive ports will be allocated of which the first
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one can be found in `port`.
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## `flags`
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flags
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## `n_ports`
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the amount of ports
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# Returns
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a `RTSPAddress` that should be freed with
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gst_rtsp_address_free after use or `None` when no address could be
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acquired.
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<!-- trait RTSPAddressPoolExt::fn add_range -->
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Adds the addresses from `min_addess` to `max_address` (inclusive)
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to `self`. The valid port range for the addresses will be from `min_port` to
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`max_port` inclusive.
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When `ttl` is 0, `min_address` and `max_address` should be unicast addresses.
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`min_address` and `max_address` can be set to
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`GST_RTSP_ADDRESS_POOL_ANY_IPV4` or `GST_RTSP_ADDRESS_POOL_ANY_IPV6` to bind
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to all available IPv4 or IPv6 addresses.
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When `ttl` > 0, `min_address` and `max_address` should be multicast addresses.
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## `min_address`
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a minimum address to add
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## `max_address`
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a maximum address to add
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## `min_port`
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the minimum port
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## `max_port`
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the maximum port
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## `ttl`
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a TTL or 0 for unicast addresses
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# Returns
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`true` if the addresses could be added.
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<!-- trait RTSPAddressPoolExt::fn clear -->
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Clear all addresses in `self`. There should be no outstanding
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allocations.
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<!-- trait RTSPAddressPoolExt::fn dump -->
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Dump the free and allocated addresses to stdout.
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<!-- trait RTSPAddressPoolExt::fn has_unicast_addresses -->
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Used to know if the pool includes any unicast addresses.
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# Returns
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`true` if the pool includes any unicast addresses, `false` otherwise
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<!-- trait RTSPAddressPoolExt::fn reserve_address -->
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Take a specific address and ports from `self`. `n_ports` consecutive
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ports will be allocated of which the first one can be found in
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`port`.
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If `ttl` is 0, `address` should be a unicast address. If `ttl` > 0, `address`
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should be a valid multicast address.
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## `ip_address`
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The IP address to reserve
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## `port`
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The first port to reserve
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## `n_ports`
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The number of ports
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## `ttl`
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The requested ttl
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## `address`
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storage for a `RTSPAddress`
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# Returns
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`RTSPAddressPoolResult::Ok` if an address was reserved. The address
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is returned in `address` and should be freed with gst_rtsp_address_free
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after use.
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<!-- enum RTSPAddressPoolResult -->
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Result codes from RTSP address pool functions.
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<!-- enum RTSPAddressPoolResult::variant Ok -->
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no error
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<!-- enum RTSPAddressPoolResult::variant Einval -->
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invalid arguments were provided to a function
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<!-- enum RTSPAddressPoolResult::variant Ereserved -->
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the addres has already been reserved
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<!-- enum RTSPAddressPoolResult::variant Erange -->
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the address is not in the pool
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<!-- enum RTSPAddressPoolResult::variant Elast -->
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last error
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<!-- struct RTSPAuth -->
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The authentication structure.
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# Implements
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[`RTSPAuthExt`](trait.RTSPAuthExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPAuthExtManual`](prelude/trait.RTSPAuthExtManual.html)
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<!-- trait RTSPAuthExt -->
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Trait containing all `RTSPAuth` methods.
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# Implementors
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[`RTSPAuth`](struct.RTSPAuth.html)
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<!-- impl RTSPAuth::fn new -->
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Create a new `RTSPAuth` instance.
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# Returns
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a new `RTSPAuth`
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<!-- impl RTSPAuth::fn check -->
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Check if `check` is allowed in the current context.
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## `check`
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the item to check
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# Returns
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FALSE if check failed.
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<!-- impl RTSPAuth::fn make_basic -->
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Construct a Basic authorisation token from `user` and `pass`.
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## `user`
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a userid
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## `pass`
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a password
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# Returns
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the base64 encoding of the string `user`:`pass`.
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`g_free` after usage.
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<!-- trait RTSPAuthExt::fn add_basic -->
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Add a basic token for the default authentication algorithm that
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enables the client with privileges listed in `token`.
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## `basic`
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the basic token
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## `token`
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authorisation token
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<!-- trait RTSPAuthExt::fn add_digest -->
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Add a digest `user` and `pass` for the default authentication algorithm that
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enables the client with privileges listed in `token`.
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Feature: `v1_12`
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## `user`
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the digest user name
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## `pass`
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the digest password
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## `token`
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authorisation token
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<!-- trait RTSPAuthExt::fn get_default_token -->
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Get the default token for `self`. This token will be used for unauthenticated
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users.
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# Returns
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the `RTSPToken` of `self`. `gst_rtsp_token_unref` after
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usage.
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<!-- trait RTSPAuthExt::fn get_realm -->
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Feature: `v1_16`
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# Returns
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the `realm` of `self`
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<!-- trait RTSPAuthExt::fn get_supported_methods -->
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Gets the supported authentication methods of `self`.
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Feature: `v1_12`
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# Returns
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The supported authentication methods
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<!-- trait RTSPAuthExt::fn get_tls_authentication_mode -->
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Get the `gio::TlsAuthenticationMode`.
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# Returns
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the `gio::TlsAuthenticationMode`.
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<!-- trait RTSPAuthExt::fn get_tls_certificate -->
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Get the `gio::TlsCertificate` used for negotiating TLS `self`.
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# Returns
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the `gio::TlsCertificate` of `self`. `gobject::ObjectExt::unref` after
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usage.
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<!-- trait RTSPAuthExt::fn get_tls_database -->
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Get the `gio::TlsDatabase` used for verifying client certificate.
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# Returns
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the `gio::TlsDatabase` of `self`. `gobject::ObjectExt::unref` after
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usage.
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<!-- trait RTSPAuthExt::fn parse_htdigest -->
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Parse the contents of the file at `path` and enable the privileges
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listed in `token` for the users it describes.
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The format of the file is expected to match the format described by
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<https://en.wikipedia.org/wiki/Digest_access_authentication`The_.htdigest_file`>,
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as output by the `htdigest` command.
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Feature: `v1_16`
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## `path`
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Path to the htdigest file
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## `token`
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authorisation token
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# Returns
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`true` if the file was successfully parsed, `false` otherwise.
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<!-- trait RTSPAuthExt::fn remove_basic -->
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Removes `basic` authentication token.
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## `basic`
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the basic token
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<!-- trait RTSPAuthExt::fn remove_digest -->
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Removes a digest user.
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Feature: `v1_12`
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## `user`
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the digest user name
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<!-- trait RTSPAuthExt::fn set_default_token -->
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Set the default `RTSPToken` to `token` in `self`. The default token will
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be used for unauthenticated users.
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## `token`
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a `RTSPToken`
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<!-- trait RTSPAuthExt::fn set_realm -->
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Set the `realm` of `self`
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Feature: `v1_16`
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<!-- trait RTSPAuthExt::fn set_supported_methods -->
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Sets the supported authentication `methods` for `self`.
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Feature: `v1_12`
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## `methods`
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supported methods
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<!-- trait RTSPAuthExt::fn set_tls_authentication_mode -->
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The `gio::TlsAuthenticationMode` to set on the underlying GTlsServerConnection.
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When set to another value than `gio::TlsAuthenticationMode::None`,
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`RTSPAuth::accept-certificate` signal will be emitted and must be handled.
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## `mode`
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a `gio::TlsAuthenticationMode`
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<!-- trait RTSPAuthExt::fn set_tls_certificate -->
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Set the TLS certificate for the auth. Client connections will only
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be accepted when TLS is negotiated.
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## `cert`
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a `gio::TlsCertificate`
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<!-- trait RTSPAuthExt::fn set_tls_database -->
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Sets the certificate database that is used to verify peer certificates.
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If set to `None` (the default), then peer certificate validation will always
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set the `gio::TlsCertificateFlags::UnknownCa` error.
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## `database`
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a `gio::TlsDatabase`
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<!-- trait RTSPAuthExt::fn connect_accept_certificate -->
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Emitted during the TLS handshake after the client certificate has
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been received. See also `RTSPAuthExt::set_tls_authentication_mode`.
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## `connection`
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a `gio::TlsConnection`
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## `peer_cert`
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the peer's `gio::TlsCertificate`
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## `errors`
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the problems with `peer_cert`.
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# Returns
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`true` to accept `peer_cert` (which will also
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immediately end the signal emission). `false` to allow the signal
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emission to continue, which will cause the handshake to fail if
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no one else overrides it.
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<!-- struct RTSPClient -->
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The client object represents the connection and its state with a client.
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# Implements
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[`RTSPClientExt`](trait.RTSPClientExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPClientExtManual`](prelude/trait.RTSPClientExtManual.html)
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<!-- trait RTSPClientExt -->
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Trait containing all `RTSPClient` methods.
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# Implementors
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[`RTSPClient`](struct.RTSPClient.html)
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<!-- impl RTSPClient::fn new -->
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Create a new `RTSPClient` instance.
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# Returns
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a new `RTSPClient`
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<!-- trait RTSPClientExt::fn attach -->
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Attaches `self` to `context`. When the mainloop for `context` is run, the
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client will be dispatched. When `context` is `None`, the default context will be
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used).
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This function should be called when the client properties and urls are fully
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configured and the client is ready to start.
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## `context`
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a `glib::MainContext`
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# Returns
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the ID (greater than 0) for the source within the GMainContext.
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<!-- trait RTSPClientExt::fn close -->
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Close the connection of `self` and remove all media it was managing.
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<!-- trait RTSPClientExt::fn get_auth -->
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Get the `RTSPAuth` used as the authentication manager of `self`.
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# Returns
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the `RTSPAuth` of `self`.
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`gobject::ObjectExt::unref` after usage.
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<!-- trait RTSPClientExt::fn get_connection -->
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Get the `gst_rtsp::RTSPConnection` of `self`.
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# Returns
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the `gst_rtsp::RTSPConnection` of `self`.
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The connection object returned remains valid until the client is freed.
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<!-- trait RTSPClientExt::fn get_content_length_limit -->
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Get the Content-Length limit of `self`.
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Feature: `v1_18`
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# Returns
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the Content-Length limit.
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<!-- trait RTSPClientExt::fn get_mount_points -->
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Get the `RTSPMountPoints` object that `self` uses to manage its sessions.
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# Returns
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a `RTSPMountPoints`, unref after usage.
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<!-- trait RTSPClientExt::fn get_session_pool -->
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Get the `RTSPSessionPool` object that `self` uses to manage its sessions.
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# Returns
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a `RTSPSessionPool`, unref after usage.
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<!-- trait RTSPClientExt::fn get_stream_transport -->
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This is useful when providing a send function through
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`RTSPClientExt::set_send_func` when doing RTSP over TCP:
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the send function must call gst_rtsp_stream_transport_message_sent ()
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on the appropriate transport when data has been received for streaming
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to continue.
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Feature: `v1_18`
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# Returns
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the `RTSPStreamTransport` associated with `channel`.
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<!-- trait RTSPClientExt::fn get_thread_pool -->
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Get the `RTSPThreadPool` used as the thread pool of `self`.
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# Returns
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the `RTSPThreadPool` of `self`. `gobject::ObjectExt::unref` after
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usage.
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<!-- trait RTSPClientExt::fn handle_message -->
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Let the client handle `message`.
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## `message`
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an `gst_rtsp::RTSPMessage`
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# Returns
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a `gst_rtsp::RTSPResult`.
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<!-- trait RTSPClientExt::fn send_message -->
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Send a message message to the remote end. `message` must be a
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`gst_rtsp::RTSPMsgType::Request` or a `gst_rtsp::RTSPMsgType::Response`.
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## `session`
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a `RTSPSession` to send
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the message to or `None`
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## `message`
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The `gst_rtsp::RTSPMessage` to send
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<!-- trait RTSPClientExt::fn session_filter -->
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Call `func` for each session managed by `self`. The result value of `func`
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determines what happens to the session. `func` will be called with `self`
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locked so no further actions on `self` can be performed from `func`.
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If `func` returns `RTSPFilterResult::Remove`, the session will be removed from
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`self`.
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If `func` returns `RTSPFilterResult::Keep`, the session will remain in `self`.
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If `func` returns `RTSPFilterResult::Ref`, the session will remain in `self` but
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will also be added with an additional ref to the result `glib::List` of this
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function..
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When `func` is `None`, `RTSPFilterResult::Ref` will be assumed for each session.
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## `func`
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a callback
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## `user_data`
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user data passed to `func`
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# Returns
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a `glib::List` with all
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sessions for which `func` returned `RTSPFilterResult::Ref`. After usage, each
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element in the `glib::List` should be unreffed before the list is freed.
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<!-- trait RTSPClientExt::fn set_auth -->
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configure `auth` to be used as the authentication manager of `self`.
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## `auth`
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a `RTSPAuth`
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<!-- trait RTSPClientExt::fn set_connection -->
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Set the `gst_rtsp::RTSPConnection` of `self`. This function takes ownership of
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`conn`.
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## `conn`
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a `gst_rtsp::RTSPConnection`
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# Returns
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`true` on success.
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<!-- trait RTSPClientExt::fn set_content_length_limit -->
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Configure `self` to use the specified Content-Length limit.
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Define an appropriate request size limit and reject requests exceeding the
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limit with response status 413 Request Entity Too Large
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Feature: `v1_18`
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## `limit`
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Content-Length limit
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<!-- trait RTSPClientExt::fn set_mount_points -->
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Set `mounts` as the mount points for `self` which it will use to map urls
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to media streams. These mount points are usually inherited from the server that
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created the client but can be overriden later.
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## `mounts`
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a `RTSPMountPoints`
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<!-- trait RTSPClientExt::fn set_send_func -->
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Set `func` as the callback that will be called when a new message needs to be
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sent to the client. `user_data` is passed to `func` and `notify` is called when
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`user_data` is no longer in use.
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By default, the client will send the messages on the `gst_rtsp::RTSPConnection` that
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was configured with `RTSPClient::attach` was called.
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It is only allowed to set either a `send_func` or a `send_messages_func`
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but not both at the same time.
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## `func`
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a `GstRTSPClientSendFunc`
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## `user_data`
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user data passed to `func`
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## `notify`
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called when `user_data` is no longer in use
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<!-- trait RTSPClientExt::fn set_send_messages_func -->
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Set `func` as the callback that will be called when new messages needs to be
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sent to the client. `user_data` is passed to `func` and `notify` is called when
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`user_data` is no longer in use.
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By default, the client will send the messages on the `gst_rtsp::RTSPConnection` that
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was configured with `RTSPClient::attach` was called.
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It is only allowed to set either a `send_func` or a `send_messages_func`
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but not both at the same time.
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Feature: `v1_16`
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## `func`
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a `GstRTSPClientSendMessagesFunc`
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## `user_data`
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user data passed to `func`
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## `notify`
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called when `user_data` is no longer in use
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<!-- trait RTSPClientExt::fn set_session_pool -->
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Set `pool` as the sessionpool for `self` which it will use to find
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or allocate sessions. the sessionpool is usually inherited from the server
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that created the client but can be overridden later.
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## `pool`
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a `RTSPSessionPool`
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<!-- trait RTSPClientExt::fn set_thread_pool -->
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configure `pool` to be used as the thread pool of `self`.
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## `pool`
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a `RTSPThreadPool`
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<!-- trait RTSPClientExt::fn connect_announce_request -->
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## `ctx`
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a `RTSPContext`
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<!-- trait RTSPClientExt::fn connect_check_requirements -->
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## `ctx`
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a `RTSPContext`
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## `arr`
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a NULL-terminated array of strings
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# Returns
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a newly allocated string with comma-separated list of
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unsupported options. An empty string must be returned if
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all options are supported.
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<!-- trait RTSPClientExt::fn connect_describe_request -->
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## `ctx`
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a `RTSPContext`
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<!-- trait RTSPClientExt::fn connect_get_parameter_request -->
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## `ctx`
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a `RTSPContext`
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<!-- trait RTSPClientExt::fn connect_handle_response -->
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## `ctx`
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a `RTSPContext`
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<!-- trait RTSPClientExt::fn connect_options_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_pause_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_play_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_pre_announce_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_describe_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_get_parameter_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_options_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_pause_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_play_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_record_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_set_parameter_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_setup_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_pre_teardown_request -->
|
|
|
|
Feature: `v1_12`
|
|
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPStatusCode`, GST_RTSP_STS_OK in case of success,
|
|
otherwise an appropriate return code
|
|
<!-- trait RTSPClientExt::fn connect_record_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_send_message -->
|
|
## `session`
|
|
The session
|
|
## `message`
|
|
The message
|
|
<!-- trait RTSPClientExt::fn connect_set_parameter_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_setup_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- trait RTSPClientExt::fn connect_teardown_request -->
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
<!-- enum RTSPFilterResult -->
|
|
Possible return values for `RTSPSessionPoolExt::filter`.
|
|
<!-- enum RTSPFilterResult::variant Remove -->
|
|
Remove session
|
|
<!-- enum RTSPFilterResult::variant Keep -->
|
|
Keep session in the pool
|
|
<!-- enum RTSPFilterResult::variant Ref -->
|
|
Ref session in the result list
|
|
<!-- struct RTSPMedia -->
|
|
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.RTSPMediaExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPMediaExtManual`](prelude/trait.RTSPMediaExtManual.html)
|
|
<!-- trait RTSPMediaExt -->
|
|
Trait containing all `RTSPMedia` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPMedia`](struct.RTSPMedia.html)
|
|
<!-- impl RTSPMedia::fn new -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn collect_streams -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn complete_pipeline -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn create_stream -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn find_stream -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_address_pool -->
|
|
Get the `RTSPAddressPool` used as the address pool of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPAddressPool` of `self`.
|
|
`gobject::ObjectExt::unref` after usage.
|
|
<!-- trait RTSPMediaExt::fn get_base_time -->
|
|
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`.
|
|
<!-- trait RTSPMediaExt::fn get_buffer_size -->
|
|
Get the kernel UDP buffer size.
|
|
|
|
# Returns
|
|
|
|
the kernel UDP buffer size.
|
|
<!-- trait RTSPMediaExt::fn get_clock -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_do_retransmission -->
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
Whether retransmission requests will be sent
|
|
<!-- trait RTSPMediaExt::fn get_dscp_qos -->
|
|
Get the configured DSCP QoS of attached media.
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
the DSCP QoS value of attached streams or -1 if disabled.
|
|
<!-- trait RTSPMediaExt::fn get_element -->
|
|
Get the element that was used when constructing `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst::Element`. Unref after usage.
|
|
<!-- trait RTSPMediaExt::fn get_latency -->
|
|
Get the latency that is used for receiving media.
|
|
|
|
# Returns
|
|
|
|
latency in milliseconds
|
|
<!-- trait RTSPMediaExt::fn get_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_multicast_iface -->
|
|
Get the multicast interface used for `self`.
|
|
|
|
# Returns
|
|
|
|
the multicast interface for `self`.
|
|
`g_free` after usage.
|
|
<!-- trait RTSPMediaExt::fn get_permissions -->
|
|
Get the permissions object from `self`.
|
|
|
|
# Returns
|
|
|
|
a `RTSPPermissions` object, unref after usage.
|
|
<!-- trait RTSPMediaExt::fn get_profiles -->
|
|
Get the allowed profiles of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPProfile`
|
|
<!-- trait RTSPMediaExt::fn get_protocols -->
|
|
Get the allowed protocols of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPLowerTrans`
|
|
<!-- trait RTSPMediaExt::fn get_publish_clock_mode -->
|
|
Gets if and how the media clock should be published according to RFC7273.
|
|
|
|
# Returns
|
|
|
|
The GstRTSPPublishClockMode
|
|
<!-- trait RTSPMediaExt::fn get_range_string -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_rate_control -->
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
whether `self` will follow the Rate-Control=no behaviour as specified
|
|
in the ONVIF replay spec.
|
|
<!-- trait RTSPMediaExt::fn get_rates -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_retransmission_time -->
|
|
Get the amount of time to store retransmission data.
|
|
|
|
# Returns
|
|
|
|
the amount of time to store retransmission data.
|
|
<!-- trait RTSPMediaExt::fn get_status -->
|
|
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`.
|
|
<!-- trait RTSPMediaExt::fn get_stream -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn get_suspend_mode -->
|
|
Get how `self` will be suspended.
|
|
|
|
# Returns
|
|
|
|
`RTSPSuspendMode`.
|
|
<!-- trait RTSPMediaExt::fn get_time_provider -->
|
|
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`.
|
|
<!-- trait RTSPMediaExt::fn get_transport_mode -->
|
|
Check if the pipeline for `self` can be used for PLAY or RECORD methods.
|
|
|
|
# Returns
|
|
|
|
The transport mode.
|
|
<!-- trait RTSPMediaExt::fn handle_sdp -->
|
|
Configure an SDP on `self` for receiving streams
|
|
## `sdp`
|
|
a `gst_sdp::SDPMessage`
|
|
|
|
# Returns
|
|
|
|
TRUE on success.
|
|
<!-- trait RTSPMediaExt::fn has_completed_sender -->
|
|
See `RTSPStreamExt::is_complete`, `RTSPStreamExt::is_sender`.
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
whether `self` has at least one complete sender stream.
|
|
<!-- trait RTSPMediaExt::fn is_bind_mcast_address -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn is_eos_shutdown -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn is_receive_only -->
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
`true` if `self` is receive-only, `false` otherwise.
|
|
<!-- trait RTSPMediaExt::fn is_reusable -->
|
|
Check if the pipeline for `self` can be reused after an unprepare.
|
|
|
|
# Returns
|
|
|
|
`true` if the media can be reused
|
|
<!-- trait RTSPMediaExt::fn is_shared -->
|
|
Check if the pipeline for `self` can be shared between multiple clients.
|
|
|
|
# Returns
|
|
|
|
`true` if the media can be shared between clients.
|
|
<!-- trait RTSPMediaExt::fn is_stop_on_disconnect -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn is_time_provider -->
|
|
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`.
|
|
<!-- trait RTSPMediaExt::fn lock -->
|
|
Lock the entire media. This is needed by callers such as rtsp_client to
|
|
protect the media when it is shared by many clients.
|
|
The lock prevents that concurrent clients alters the shared media,
|
|
while one client already is working with it.
|
|
Typically the lock is taken in external RTSP API calls that uses shared media
|
|
such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.
|
|
|
|
As best practice take the lock as soon as the function get hold of a shared
|
|
media object. Release the lock right before the function returns.
|
|
|
|
Feature: `v1_18`
|
|
|
|
<!-- trait RTSPMediaExt::fn n_streams -->
|
|
Get the number of streams in this media.
|
|
|
|
# Returns
|
|
|
|
The number of streams.
|
|
<!-- trait RTSPMediaExt::fn prepare -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn seek -->
|
|
Seek the pipeline of `self` to `range`. `self` must be prepared with
|
|
`RTSPMediaExt::prepare`.
|
|
## `range`
|
|
a `gst_rtsp::RTSPTimeRange`
|
|
|
|
# Returns
|
|
|
|
`true` on success.
|
|
<!-- trait RTSPMediaExt::fn seek_full -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn seek_trickmode -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn seekable -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn set_address_pool -->
|
|
configure `pool` to be used as the address pool of `self`.
|
|
## `pool`
|
|
a `RTSPAddressPool`
|
|
<!-- trait RTSPMediaExt::fn set_bind_mcast_address -->
|
|
Decide whether the multicast socket should be bound to a multicast address or
|
|
INADDR_ANY.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `bind_mcast_addr`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_buffer_size -->
|
|
Set the kernel UDP buffer size.
|
|
## `size`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_clock -->
|
|
Configure the clock used for the media.
|
|
## `clock`
|
|
`gst::Clock` to be used
|
|
<!-- trait RTSPMediaExt::fn set_do_retransmission -->
|
|
Set whether retransmission requests will be sent
|
|
|
|
Feature: `v1_16`
|
|
|
|
<!-- trait RTSPMediaExt::fn set_dscp_qos -->
|
|
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)
|
|
<!-- trait RTSPMediaExt::fn set_eos_shutdown -->
|
|
Set or unset if an EOS event will be sent to the pipeline for `self` before
|
|
it is unprepared.
|
|
## `eos_shutdown`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_latency -->
|
|
Configure the latency used for receiving media.
|
|
## `latency`
|
|
latency in milliseconds
|
|
<!-- trait RTSPMediaExt::fn set_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn set_multicast_iface -->
|
|
configure `multicast_iface` to be used for `self`.
|
|
## `multicast_iface`
|
|
a multicast interface name
|
|
<!-- trait RTSPMediaExt::fn set_permissions -->
|
|
Set `permissions` on `self`.
|
|
## `permissions`
|
|
a `RTSPPermissions`
|
|
<!-- trait RTSPMediaExt::fn set_pipeline_state -->
|
|
Set the state of the pipeline managed by `self` to `state`
|
|
## `state`
|
|
the target state of the pipeline
|
|
<!-- trait RTSPMediaExt::fn set_profiles -->
|
|
Configure the allowed lower transport for `self`.
|
|
## `profiles`
|
|
the new flags
|
|
<!-- trait RTSPMediaExt::fn set_protocols -->
|
|
Configure the allowed lower transport for `self`.
|
|
## `protocols`
|
|
the new flags
|
|
<!-- trait RTSPMediaExt::fn set_publish_clock_mode -->
|
|
Sets if and how the media clock should be published according to RFC7273.
|
|
## `mode`
|
|
the clock publish mode
|
|
<!-- trait RTSPMediaExt::fn set_rate_control -->
|
|
Define whether `self` will follow the Rate-Control=no behaviour as specified
|
|
in the ONVIF replay spec.
|
|
|
|
Feature: `v1_18`
|
|
|
|
<!-- trait RTSPMediaExt::fn set_retransmission_time -->
|
|
Set the amount of time to store retransmission packets.
|
|
## `time`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_reusable -->
|
|
Set or unset if the pipeline for `self` can be reused after the pipeline has
|
|
been unprepared.
|
|
## `reusable`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_shared -->
|
|
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
|
|
<!-- trait RTSPMediaExt::fn set_state -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn set_stop_on_disconnect -->
|
|
Set or unset if the pipeline for `self` should be stopped when a
|
|
client disconnects without sending TEARDOWN.
|
|
## `stop_on_disconnect`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn set_suspend_mode -->
|
|
Control how @ media will be suspended after the SDP has been generated and
|
|
after a PAUSE request has been performed.
|
|
|
|
Media must be unprepared when setting the suspend mode.
|
|
## `mode`
|
|
the new `RTSPSuspendMode`
|
|
<!-- trait RTSPMediaExt::fn set_transport_mode -->
|
|
Sets if the media pipeline can work in PLAY or RECORD mode
|
|
## `mode`
|
|
the new value
|
|
<!-- trait RTSPMediaExt::fn setup_sdp -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn suspend -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn take_pipeline -->
|
|
Set `pipeline` as the `gst::Pipeline` for `self`. Ownership is
|
|
taken of `pipeline`.
|
|
## `pipeline`
|
|
a `gst::Pipeline`
|
|
<!-- trait RTSPMediaExt::fn unlock -->
|
|
Unlock the media.
|
|
|
|
Feature: `v1_18`
|
|
|
|
<!-- trait RTSPMediaExt::fn unprepare -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn unsuspend -->
|
|
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.
|
|
<!-- trait RTSPMediaExt::fn use_time_provider -->
|
|
Set `self` to provide a `gst_net::NetTimeProvider`.
|
|
## `time_provider`
|
|
if a `gst_net::NetTimeProvider` should be used
|
|
<!-- struct RTSPMediaFactory -->
|
|
The definition and logic for constructing the pipeline for a media. The media
|
|
can contain multiple streams like audio and video.
|
|
|
|
# Implements
|
|
|
|
[`RTSPMediaFactoryExt`](trait.RTSPMediaFactoryExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPMediaFactoryExtManual`](prelude/trait.RTSPMediaFactoryExtManual.html)
|
|
<!-- trait RTSPMediaFactoryExt -->
|
|
Trait containing all `RTSPMediaFactory` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPMediaFactoryURI`](struct.RTSPMediaFactoryURI.html), [`RTSPMediaFactory`](struct.RTSPMediaFactory.html)
|
|
<!-- impl RTSPMediaFactory::fn new -->
|
|
Create a new `RTSPMediaFactory` instance.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPMediaFactory` object.
|
|
<!-- trait RTSPMediaFactoryExt::fn add_role -->
|
|
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
|
|
<!-- trait RTSPMediaFactoryExt::fn add_role_from_structure -->
|
|
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`
|
|
|
|
<!-- trait RTSPMediaFactoryExt::fn construct -->
|
|
Construct the media object and create its streams. Implementations
|
|
should create the needed gstreamer elements and add them to the result
|
|
object. No state changes should be performed on them yet.
|
|
|
|
One or more GstRTSPStream objects should be created from the result
|
|
with gst_rtsp_media_create_stream ().
|
|
|
|
After the media is constructed, it can be configured and then prepared
|
|
with gst_rtsp_media_prepare ().
|
|
## `url`
|
|
the url used
|
|
|
|
# Returns
|
|
|
|
a new `RTSPMedia` if the media could be prepared.
|
|
<!-- trait RTSPMediaFactoryExt::fn create_element -->
|
|
Construct and return a `gst::Element` that is a `gst::Bin` containing
|
|
the elements to use for streaming the media.
|
|
|
|
The bin should contain payloaders pay\%d for each stream. The default
|
|
implementation of this function returns the bin created from the
|
|
launch parameter.
|
|
## `url`
|
|
the url used
|
|
|
|
# Returns
|
|
|
|
a new `gst::Element`.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_address_pool -->
|
|
Get the `RTSPAddressPool` used as the address pool of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPAddressPool` of `self`. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_buffer_size -->
|
|
Get the kernel UDP buffer size.
|
|
|
|
# Returns
|
|
|
|
the kernel UDP buffer size.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_clock -->
|
|
Returns the clock that is going to be used by the pipelines
|
|
of all medias created from this factory.
|
|
|
|
# Returns
|
|
|
|
The GstClock
|
|
<!-- trait RTSPMediaFactoryExt::fn get_do_retransmission -->
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
Whether retransmission requests will be sent for receiving media
|
|
<!-- trait RTSPMediaFactoryExt::fn get_dscp_qos -->
|
|
Get the configured media DSCP QoS.
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
the media DSCP QoS value or -1 if disabled.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_latency -->
|
|
Get the latency that is used for receiving media
|
|
|
|
# Returns
|
|
|
|
latency in milliseconds
|
|
<!-- trait RTSPMediaFactoryExt::fn get_launch -->
|
|
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.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_media_gtype -->
|
|
Return the GType of the GstRTSPMedia subclass this
|
|
factory will create.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_multicast_iface -->
|
|
Get the multicast interface used for `self`.
|
|
|
|
# Returns
|
|
|
|
the multicast interface for `self`. `g_free` after
|
|
usage.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_permissions -->
|
|
Get the permissions object from `self`.
|
|
|
|
# Returns
|
|
|
|
a `RTSPPermissions` object, unref after usage.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_profiles -->
|
|
Get the allowed profiles of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPProfile`
|
|
<!-- trait RTSPMediaFactoryExt::fn get_protocols -->
|
|
Get the allowed protocols of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPLowerTrans`
|
|
<!-- trait RTSPMediaFactoryExt::fn get_publish_clock_mode -->
|
|
Gets if and how the media clock should be published according to RFC7273.
|
|
|
|
# Returns
|
|
|
|
The GstRTSPPublishClockMode
|
|
<!-- trait RTSPMediaFactoryExt::fn get_retransmission_time -->
|
|
Get the time that is stored for retransmission purposes
|
|
|
|
# Returns
|
|
|
|
a `gst::ClockTime`
|
|
<!-- trait RTSPMediaFactoryExt::fn get_suspend_mode -->
|
|
Get how media created from this factory will be suspended.
|
|
|
|
# Returns
|
|
|
|
a `RTSPSuspendMode`.
|
|
<!-- trait RTSPMediaFactoryExt::fn get_transport_mode -->
|
|
Get if media created from this factory can be used for PLAY or RECORD
|
|
methods.
|
|
|
|
# Returns
|
|
|
|
The transport mode.
|
|
<!-- trait RTSPMediaFactoryExt::fn is_bind_mcast_address -->
|
|
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.
|
|
<!-- trait RTSPMediaFactoryExt::fn is_eos_shutdown -->
|
|
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.
|
|
<!-- trait RTSPMediaFactoryExt::fn is_shared -->
|
|
Get if media created from this factory can be shared between clients.
|
|
|
|
# Returns
|
|
|
|
`true` if the media will be shared between clients.
|
|
<!-- trait RTSPMediaFactoryExt::fn set_address_pool -->
|
|
configure `pool` to be used as the address pool of `self`.
|
|
## `pool`
|
|
a `RTSPAddressPool`
|
|
<!-- trait RTSPMediaFactoryExt::fn set_bind_mcast_address -->
|
|
Decide whether the multicast socket should be bound to a multicast address or
|
|
INADDR_ANY.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `bind_mcast_addr`
|
|
the new value
|
|
<!-- trait RTSPMediaFactoryExt::fn set_buffer_size -->
|
|
Set the kernel UDP buffer size.
|
|
## `size`
|
|
the new value
|
|
<!-- trait RTSPMediaFactoryExt::fn set_clock -->
|
|
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
|
|
<!-- trait RTSPMediaFactoryExt::fn set_do_retransmission -->
|
|
Set whether retransmission requests will be sent for
|
|
receiving media
|
|
|
|
Feature: `v1_16`
|
|
|
|
<!-- trait RTSPMediaFactoryExt::fn set_dscp_qos -->
|
|
Configure the media dscp qos to `dscp_qos`.
|
|
|
|
Feature: `v1_18`
|
|
|
|
## `dscp_qos`
|
|
a new dscp qos value (0-63, or -1 to disable)
|
|
<!-- trait RTSPMediaFactoryExt::fn set_eos_shutdown -->
|
|
Configure if media created from this factory will have an EOS sent to the
|
|
pipeline before shutdown.
|
|
## `eos_shutdown`
|
|
the new value
|
|
<!-- trait RTSPMediaFactoryExt::fn set_latency -->
|
|
Configure the latency used for receiving media
|
|
## `latency`
|
|
latency in milliseconds
|
|
<!-- trait RTSPMediaFactoryExt::fn set_launch -->
|
|
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
|
|
<!-- trait RTSPMediaFactoryExt::fn set_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPMediaFactoryExt::fn set_media_gtype -->
|
|
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
|
|
<!-- trait RTSPMediaFactoryExt::fn set_multicast_iface -->
|
|
configure `multicast_iface` to be used for `self`.
|
|
## `multicast_iface`
|
|
a multicast interface name
|
|
<!-- trait RTSPMediaFactoryExt::fn set_permissions -->
|
|
Set `permissions` on `self`.
|
|
## `permissions`
|
|
a `RTSPPermissions`
|
|
<!-- trait RTSPMediaFactoryExt::fn set_profiles -->
|
|
Configure the allowed profiles for `self`.
|
|
## `profiles`
|
|
the new flags
|
|
<!-- trait RTSPMediaFactoryExt::fn set_protocols -->
|
|
Configure the allowed lower transport for `self`.
|
|
## `protocols`
|
|
the new flags
|
|
<!-- trait RTSPMediaFactoryExt::fn set_publish_clock_mode -->
|
|
Sets if and how the media clock should be published according to RFC7273.
|
|
## `mode`
|
|
the clock publish mode
|
|
<!-- trait RTSPMediaFactoryExt::fn set_retransmission_time -->
|
|
Configure the time to store for possible retransmission
|
|
## `time`
|
|
a `gst::ClockTime`
|
|
<!-- trait RTSPMediaFactoryExt::fn set_shared -->
|
|
Configure if media created from this factory can be shared between clients.
|
|
## `shared`
|
|
the new value
|
|
<!-- trait RTSPMediaFactoryExt::fn set_stop_on_disconnect -->
|
|
Configure if media created from this factory should be stopped
|
|
when a client disconnects without sending TEARDOWN.
|
|
## `stop_on_disconnect`
|
|
the new value
|
|
<!-- trait RTSPMediaFactoryExt::fn set_suspend_mode -->
|
|
Configure how media created from this factory will be suspended.
|
|
## `mode`
|
|
the new `RTSPSuspendMode`
|
|
<!-- trait RTSPMediaFactoryExt::fn set_transport_mode -->
|
|
Configure if this factory creates media for PLAY or RECORD modes.
|
|
## `mode`
|
|
the new value
|
|
<!-- struct RTSPMediaFactoryURI -->
|
|
A media factory that creates a pipeline to play any uri.
|
|
|
|
# Implements
|
|
|
|
[`RTSPMediaFactoryURIExt`](trait.RTSPMediaFactoryURIExt.html), [`RTSPMediaFactoryExt`](trait.RTSPMediaFactoryExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPMediaFactoryExtManual`](prelude/trait.RTSPMediaFactoryExtManual.html)
|
|
<!-- trait RTSPMediaFactoryURIExt -->
|
|
Trait containing all `RTSPMediaFactoryURI` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPMediaFactoryURI`](struct.RTSPMediaFactoryURI.html)
|
|
<!-- impl RTSPMediaFactoryURI::fn new -->
|
|
Create a new `RTSPMediaFactoryURI` instance.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPMediaFactoryURI` object.
|
|
<!-- trait RTSPMediaFactoryURIExt::fn get_uri -->
|
|
Get the URI that will provide media for this factory.
|
|
|
|
# Returns
|
|
|
|
the configured URI. `g_free` after usage.
|
|
<!-- trait RTSPMediaFactoryURIExt::fn set_uri -->
|
|
Set the URI of the resource that will be streamed by this factory.
|
|
## `uri`
|
|
the uri the stream
|
|
<!-- enum RTSPMediaStatus -->
|
|
The state of the media pipeline.
|
|
<!-- enum RTSPMediaStatus::variant Unprepared -->
|
|
media pipeline not prerolled
|
|
<!-- enum RTSPMediaStatus::variant Unpreparing -->
|
|
media pipeline is busy doing a clean
|
|
shutdown.
|
|
<!-- enum RTSPMediaStatus::variant Preparing -->
|
|
media pipeline is prerolling
|
|
<!-- enum RTSPMediaStatus::variant Prepared -->
|
|
media pipeline is prerolled
|
|
<!-- enum RTSPMediaStatus::variant Suspended -->
|
|
media is suspended
|
|
<!-- enum RTSPMediaStatus::variant Error -->
|
|
media pipeline is in error
|
|
<!-- struct RTSPMountPoints -->
|
|
Creates a `RTSPMediaFactory` object for a given url.
|
|
|
|
# Implements
|
|
|
|
[`RTSPMountPointsExt`](trait.RTSPMountPointsExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
|
|
<!-- trait RTSPMountPointsExt -->
|
|
Trait containing all `RTSPMountPoints` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPMountPoints`](struct.RTSPMountPoints.html)
|
|
<!-- impl RTSPMountPoints::fn new -->
|
|
Make a new mount points object.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPMountPoints`
|
|
<!-- trait RTSPMountPointsExt::fn add_factory -->
|
|
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`
|
|
<!-- trait RTSPMountPointsExt::fn make_path -->
|
|
Make a path string from `url`.
|
|
## `url`
|
|
a `gst_rtsp::RTSPUrl`
|
|
|
|
# Returns
|
|
|
|
a path string for `url`, `g_free` after usage.
|
|
<!-- trait RTSPMountPointsExt::fn match -->
|
|
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`.
|
|
`gobject::ObjectExt::unref` after usage.
|
|
<!-- trait RTSPMountPointsExt::fn remove_factory -->
|
|
Remove the `RTSPMediaFactory` associated with `path` in `self`.
|
|
## `path`
|
|
a mount point
|
|
<!-- enum RTSPPublishClockMode -->
|
|
Whether the clock and possibly RTP/clock offset should be published according to RFC7273.
|
|
<!-- enum RTSPPublishClockMode::variant None -->
|
|
Publish nothing
|
|
<!-- enum RTSPPublishClockMode::variant Clock -->
|
|
Publish the clock but not the offset
|
|
<!-- enum RTSPPublishClockMode::variant ClockAndOffset -->
|
|
Publish the clock and offset
|
|
<!-- struct RTSPServer -->
|
|
This object listens on a port, creates and manages the clients connected to
|
|
it.
|
|
|
|
# Implements
|
|
|
|
[`RTSPServerExt`](trait.RTSPServerExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPServerExtManual`](prelude/trait.RTSPServerExtManual.html)
|
|
<!-- trait RTSPServerExt -->
|
|
Trait containing all `RTSPServer` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPServer`](struct.RTSPServer.html)
|
|
<!-- impl RTSPServer::fn new -->
|
|
Create a new `RTSPServer` instance.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPServer`
|
|
<!-- impl RTSPServer::fn io_func -->
|
|
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.
|
|
<!-- trait RTSPServerExt::fn attach -->
|
|
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.
|
|
<!-- trait RTSPServerExt::fn client_filter -->
|
|
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.
|
|
<!-- trait RTSPServerExt::fn create_socket -->
|
|
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.
|
|
<!-- trait RTSPServerExt::fn create_source -->
|
|
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 ()
|
|
<!-- trait RTSPServerExt::fn get_address -->
|
|
Get the address on which the server will accept connections.
|
|
|
|
# Returns
|
|
|
|
the server address. `g_free` after usage.
|
|
<!-- trait RTSPServerExt::fn get_auth -->
|
|
Get the `RTSPAuth` used as the authentication manager of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPAuth` of `self`. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPServerExt::fn get_backlog -->
|
|
The maximum amount of queued requests for the server.
|
|
|
|
# Returns
|
|
|
|
the server backlog.
|
|
<!-- trait RTSPServerExt::fn get_bound_port -->
|
|
Get the port number where the server was bound to.
|
|
|
|
# Returns
|
|
|
|
the port number
|
|
<!-- trait RTSPServerExt::fn get_content_length_limit -->
|
|
Get the Content-Length limit of `self`.
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
the Content-Length limit.
|
|
<!-- trait RTSPServerExt::fn get_mount_points -->
|
|
Get the `RTSPMountPoints` used as the mount points of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPMountPoints` of `self`. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPServerExt::fn get_service -->
|
|
Get the service on which the server will accept connections.
|
|
|
|
# Returns
|
|
|
|
the service. use `g_free` after usage.
|
|
<!-- trait RTSPServerExt::fn get_session_pool -->
|
|
Get the `RTSPSessionPool` used as the session pool of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPSessionPool` used for sessions. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPServerExt::fn get_thread_pool -->
|
|
Get the `RTSPThreadPool` used as the thread pool of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPThreadPool` of `self`. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPServerExt::fn set_address -->
|
|
Configure `self` to accept connections on the given address.
|
|
|
|
This function must be called before the server is bound.
|
|
## `address`
|
|
the address
|
|
<!-- trait RTSPServerExt::fn set_auth -->
|
|
configure `auth` to be used as the authentication manager of `self`.
|
|
## `auth`
|
|
a `RTSPAuth`
|
|
<!-- trait RTSPServerExt::fn set_backlog -->
|
|
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
|
|
<!-- trait RTSPServerExt::fn set_content_length_limit -->
|
|
Define an appropriate request size limit and reject requests exceeding the
|
|
limit.
|
|
|
|
Feature: `v1_18`
|
|
|
|
<!-- trait RTSPServerExt::fn set_mount_points -->
|
|
configure `mounts` to be used as the mount points of `self`.
|
|
## `mounts`
|
|
a `RTSPMountPoints`
|
|
<!-- trait RTSPServerExt::fn set_service -->
|
|
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
|
|
<!-- trait RTSPServerExt::fn set_session_pool -->
|
|
configure `pool` to be used as the session pool of `self`.
|
|
## `pool`
|
|
a `RTSPSessionPool`
|
|
<!-- trait RTSPServerExt::fn set_thread_pool -->
|
|
configure `pool` to be used as the thread pool of `self`.
|
|
## `pool`
|
|
a `RTSPThreadPool`
|
|
<!-- trait RTSPServerExt::fn transfer_connection -->
|
|
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.
|
|
<!-- struct RTSPSession -->
|
|
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.RTSPSessionExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
|
|
<!-- trait RTSPSessionExt -->
|
|
Trait containing all `RTSPSession` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPSession`](struct.RTSPSession.html)
|
|
<!-- impl RTSPSession::fn new -->
|
|
Create a new `RTSPSession` instance with `sessionid`.
|
|
## `sessionid`
|
|
a session id
|
|
|
|
# Returns
|
|
|
|
a new `RTSPSession`
|
|
<!-- trait RTSPSessionExt::fn allow_expire -->
|
|
Allow `self` to expire. This method must be called an equal
|
|
amount of time as `RTSPSessionExt::prevent_expire`.
|
|
<!-- trait RTSPSessionExt::fn filter -->
|
|
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.
|
|
<!-- trait RTSPSessionExt::fn get_header -->
|
|
Get the string that can be placed in the Session header field.
|
|
|
|
# Returns
|
|
|
|
the Session header of `self`.
|
|
`g_free` after usage.
|
|
<!-- trait RTSPSessionExt::fn get_media -->
|
|
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`.
|
|
<!-- trait RTSPSessionExt::fn get_sessionid -->
|
|
Get the sessionid of `self`.
|
|
|
|
# Returns
|
|
|
|
the sessionid of `self`.
|
|
The value remains valid as long as `self` is alive.
|
|
<!-- trait RTSPSessionExt::fn get_timeout -->
|
|
Get the timeout value of `self`.
|
|
|
|
# Returns
|
|
|
|
the timeout of `self` in seconds.
|
|
<!-- trait RTSPSessionExt::fn is_expired -->
|
|
Check if `self` timeout out.
|
|
|
|
# Deprecated
|
|
|
|
Use `RTSPSessionExt::is_expired_usec` instead.
|
|
## `now`
|
|
the current system time
|
|
|
|
# Returns
|
|
|
|
`true` if `self` timed out
|
|
<!-- trait RTSPSessionExt::fn is_expired_usec -->
|
|
Check if `self` timeout out.
|
|
## `now`
|
|
the current monotonic time
|
|
|
|
# Returns
|
|
|
|
`true` if `self` timed out
|
|
<!-- trait RTSPSessionExt::fn manage_media -->
|
|
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 `RTSPSessionMedia` object.
|
|
<!-- trait RTSPSessionExt::fn next_timeout -->
|
|
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.
|
|
<!-- trait RTSPSessionExt::fn next_timeout_usec -->
|
|
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.
|
|
<!-- trait RTSPSessionExt::fn prevent_expire -->
|
|
Prevent `self` from expiring.
|
|
<!-- trait RTSPSessionExt::fn release_media -->
|
|
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`.
|
|
<!-- trait RTSPSessionExt::fn set_timeout -->
|
|
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
|
|
<!-- trait RTSPSessionExt::fn touch -->
|
|
Update the last_access time of the session to the current time.
|
|
<!-- struct RTSPSessionMedia -->
|
|
State of a client session regarding a specific media identified by path.
|
|
|
|
# Implements
|
|
|
|
[`RTSPSessionMediaExt`](trait.RTSPSessionMediaExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
|
|
<!-- trait RTSPSessionMediaExt -->
|
|
Trait containing all `RTSPSessionMedia` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPSessionMedia`](struct.RTSPSessionMedia.html)
|
|
<!-- impl RTSPSessionMedia::fn new -->
|
|
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`.
|
|
<!-- trait RTSPSessionMediaExt::fn alloc_channels -->
|
|
Fill `range` with the next available min and max channels for
|
|
interleaved transport.
|
|
## `range`
|
|
a `gst_rtsp::RTSPRange`
|
|
|
|
# Returns
|
|
|
|
`true` on success.
|
|
<!-- trait RTSPSessionMediaExt::fn get_base_time -->
|
|
Get the base_time of the `RTSPMedia` in `self`
|
|
|
|
# Returns
|
|
|
|
the base_time of the media.
|
|
<!-- trait RTSPSessionMediaExt::fn get_media -->
|
|
Get the `RTSPMedia` that was used when constructing `self`
|
|
|
|
# Returns
|
|
|
|
the `RTSPMedia` of `self`.
|
|
Remains valid as long as `self` is valid.
|
|
<!-- trait RTSPSessionMediaExt::fn get_rtpinfo -->
|
|
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.
|
|
<!-- trait RTSPSessionMediaExt::fn get_rtsp_state -->
|
|
Get the current RTSP state of `self`.
|
|
|
|
# Returns
|
|
|
|
the current RTSP state of `self`.
|
|
<!-- trait RTSPSessionMediaExt::fn get_transport -->
|
|
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.
|
|
<!-- trait RTSPSessionMediaExt::fn get_transports -->
|
|
Get a list of all available `RTSPStreamTransport` in this session.
|
|
|
|
Feature: `v1_14`
|
|
|
|
|
|
# Returns
|
|
|
|
a
|
|
list of `RTSPStreamTransport`, g_ptr_array_unref () after usage.
|
|
<!-- trait RTSPSessionMediaExt::fn matches -->
|
|
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`.
|
|
<!-- trait RTSPSessionMediaExt::fn set_rtsp_state -->
|
|
Set the RTSP state of `self` to `state`.
|
|
## `state`
|
|
a `gst_rtsp::RTSPState`
|
|
<!-- trait RTSPSessionMediaExt::fn set_state -->
|
|
Tell the media object `self` to change to `state`.
|
|
## `state`
|
|
the new state
|
|
|
|
# Returns
|
|
|
|
`true` on success.
|
|
<!-- trait RTSPSessionMediaExt::fn set_transport -->
|
|
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`.
|
|
<!-- struct RTSPSessionPool -->
|
|
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.RTSPSessionPoolExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPSessionPoolExtManual`](prelude/trait.RTSPSessionPoolExtManual.html)
|
|
<!-- trait RTSPSessionPoolExt -->
|
|
Trait containing all `RTSPSessionPool` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPSessionPool`](struct.RTSPSessionPool.html)
|
|
<!-- impl RTSPSessionPool::fn new -->
|
|
Create a new `RTSPSessionPool` instance.
|
|
|
|
# Returns
|
|
|
|
A new `RTSPSessionPool`. `gobject::ObjectExt::unref` after
|
|
usage.
|
|
<!-- trait RTSPSessionPoolExt::fn cleanup -->
|
|
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.
|
|
<!-- trait RTSPSessionPoolExt::fn create -->
|
|
Create a new `RTSPSession` object in `self`.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPSession`.
|
|
<!-- trait RTSPSessionPoolExt::fn create_watch -->
|
|
Create a `glib::Source` that will be dispatched when the session should be cleaned
|
|
up.
|
|
|
|
# Returns
|
|
|
|
a `glib::Source`
|
|
<!-- trait RTSPSessionPoolExt::fn filter -->
|
|
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.
|
|
<!-- trait RTSPSessionPoolExt::fn find -->
|
|
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. `gobject::ObjectExt::unref` after usage.
|
|
<!-- trait RTSPSessionPoolExt::fn get_max_sessions -->
|
|
Get the maximum allowed number of sessions in `self`. 0 means an unlimited
|
|
amount of sessions.
|
|
|
|
# Returns
|
|
|
|
the maximum allowed number of sessions.
|
|
<!-- trait RTSPSessionPoolExt::fn get_n_sessions -->
|
|
Get the amount of active sessions in `self`.
|
|
|
|
# Returns
|
|
|
|
the amount of active sessions in `self`.
|
|
<!-- trait RTSPSessionPoolExt::fn remove -->
|
|
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.
|
|
<!-- trait RTSPSessionPoolExt::fn set_max_sessions -->
|
|
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
|
|
<!-- struct RTSPStream -->
|
|
The definition of a media stream.
|
|
|
|
# Implements
|
|
|
|
[`RTSPStreamExt`](trait.RTSPStreamExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPStreamExtManual`](prelude/trait.RTSPStreamExtManual.html)
|
|
<!-- trait RTSPStreamExt -->
|
|
Trait containing all `RTSPStream` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPStream`](struct.RTSPStream.html)
|
|
<!-- impl RTSPStream::fn new -->
|
|
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`
|
|
<!-- trait RTSPStreamExt::fn add_multicast_client_address -->
|
|
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`.
|
|
<!-- trait RTSPStreamExt::fn add_transport -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn allocate_udp_sockets -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn complete_stream -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn get_address_pool -->
|
|
Get the `RTSPAddressPool` used as the address pool of `self`.
|
|
|
|
# Returns
|
|
|
|
the `RTSPAddressPool` of `self`.
|
|
`gobject::ObjectExt::unref` after usage.
|
|
<!-- trait RTSPStreamExt::fn get_buffer_size -->
|
|
Get the size of the UDP transmission buffer (in bytes)
|
|
|
|
# Returns
|
|
|
|
the size of the UDP TX buffer
|
|
<!-- trait RTSPStreamExt::fn get_caps -->
|
|
Retrieve the current caps of `self`.
|
|
|
|
# Returns
|
|
|
|
the `gst::Caps` of `self`.
|
|
use `gst::Caps::unref` after usage.
|
|
<!-- trait RTSPStreamExt::fn get_control -->
|
|
Get the control string to identify this stream.
|
|
|
|
# Returns
|
|
|
|
the control string. `g_free` after usage.
|
|
<!-- trait RTSPStreamExt::fn get_dscp_qos -->
|
|
Get the configured DSCP QoS in of the outgoing sockets.
|
|
|
|
# Returns
|
|
|
|
the DSCP QoS value of the outgoing sockets, or -1 if disbled.
|
|
<!-- trait RTSPStreamExt::fn get_index -->
|
|
Get the stream index.
|
|
|
|
# Returns
|
|
|
|
the stream index.
|
|
<!-- trait RTSPStreamExt::fn get_joined_bin -->
|
|
Get the previous joined bin with `RTSPStreamExt::join_bin` or NULL.
|
|
|
|
# Returns
|
|
|
|
the joined bin or NULL.
|
|
<!-- trait RTSPStreamExt::fn get_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn get_mtu -->
|
|
Get the configured MTU in the payloader of `self`.
|
|
|
|
# Returns
|
|
|
|
the MTU of the payloader.
|
|
<!-- trait RTSPStreamExt::fn get_multicast_address -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn get_multicast_client_addresses -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_multicast_iface -->
|
|
Get the multicast interface used for `self`.
|
|
|
|
# Returns
|
|
|
|
the multicast interface for `self`.
|
|
`g_free` after usage.
|
|
<!-- trait RTSPStreamExt::fn get_profiles -->
|
|
Get the allowed profiles of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPProfile`
|
|
<!-- trait RTSPStreamExt::fn get_protocols -->
|
|
Get the allowed protocols of `self`.
|
|
|
|
# Returns
|
|
|
|
a `gst_rtsp::RTSPLowerTrans`
|
|
<!-- trait RTSPStreamExt::fn get_pt -->
|
|
Get the stream payload type.
|
|
|
|
# Returns
|
|
|
|
the stream payload type.
|
|
<!-- trait RTSPStreamExt::fn get_publish_clock_mode -->
|
|
Gets if and how the stream clock should be published according to RFC7273.
|
|
|
|
# Returns
|
|
|
|
The GstRTSPPublishClockMode
|
|
<!-- trait RTSPStreamExt::fn get_rate_control -->
|
|
|
|
Feature: `v1_18`
|
|
|
|
|
|
# Returns
|
|
|
|
whether `self` will follow the Rate-Control=no behaviour as specified
|
|
in the ONVIF replay spec.
|
|
<!-- trait RTSPStreamExt::fn get_rates -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn get_retransmission_pt -->
|
|
Get the payload-type used for retransmission of this stream
|
|
|
|
# Returns
|
|
|
|
The retransmission PT.
|
|
<!-- trait RTSPStreamExt::fn get_retransmission_time -->
|
|
Get the amount of time to store retransmission data.
|
|
|
|
# Returns
|
|
|
|
the amount of time to store retransmission data.
|
|
<!-- trait RTSPStreamExt::fn get_rtcp_multicast_socket -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_rtcp_socket -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_rtp_multicast_socket -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_rtp_socket -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_rtpinfo -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn get_rtpsession -->
|
|
Get the RTP session of this stream.
|
|
|
|
# Returns
|
|
|
|
The RTP session of this stream. Unref after usage.
|
|
<!-- trait RTSPStreamExt::fn get_server_port -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_sinkpad -->
|
|
Get the sinkpad associated with `self`.
|
|
|
|
# Returns
|
|
|
|
the sinkpad. Unref after usage.
|
|
<!-- trait RTSPStreamExt::fn get_srcpad -->
|
|
Get the srcpad associated with `self`.
|
|
|
|
# Returns
|
|
|
|
the srcpad. Unref after usage.
|
|
<!-- trait RTSPStreamExt::fn get_srtp_encoder -->
|
|
Get the SRTP encoder for this stream.
|
|
|
|
# Returns
|
|
|
|
The SRTP encoder for this stream. Unref after usage.
|
|
<!-- trait RTSPStreamExt::fn get_ssrc -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn get_ulpfec_percentage -->
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
the amount of redundancy applied when creating ULPFEC
|
|
protection packets.
|
|
<!-- trait RTSPStreamExt::fn get_ulpfec_pt -->
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
the payload type used for ULPFEC protection packets
|
|
<!-- trait RTSPStreamExt::fn handle_keymgmt -->
|
|
Parse and handle a KeyMgmt header.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `keymgmt`
|
|
a keymgmt header
|
|
<!-- trait RTSPStreamExt::fn has_control -->
|
|
Check if `self` has the control string `control`.
|
|
## `control`
|
|
a control string
|
|
|
|
# Returns
|
|
|
|
`true` is `self` has `control` as the control string
|
|
<!-- trait RTSPStreamExt::fn is_bind_mcast_address -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn is_blocking -->
|
|
Check if `self` is blocking on a `gst::Buffer`.
|
|
|
|
# Returns
|
|
|
|
`true` if `self` is blocking
|
|
<!-- trait RTSPStreamExt::fn is_client_side -->
|
|
See `RTSPStreamExt::set_client_side`
|
|
|
|
# Returns
|
|
|
|
TRUE if this `RTSPStream` is client-side.
|
|
<!-- trait RTSPStreamExt::fn is_complete -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn is_receiver -->
|
|
Checks whether the stream is a receiver.
|
|
|
|
Feature: `v1_14`
|
|
|
|
|
|
# Returns
|
|
|
|
`true` if the stream is a receiver and `false` otherwise.
|
|
<!-- trait RTSPStreamExt::fn is_sender -->
|
|
Checks whether the stream is a sender.
|
|
|
|
Feature: `v1_14`
|
|
|
|
|
|
# Returns
|
|
|
|
`true` if the stream is a sender and `false` otherwise.
|
|
<!-- trait RTSPStreamExt::fn is_transport_supported -->
|
|
Check if `transport` can be handled by stream
|
|
## `transport`
|
|
a `gst_rtsp::RTSPTransport`
|
|
|
|
# Returns
|
|
|
|
`true` if `transport` can be handled by `self`.
|
|
<!-- trait RTSPStreamExt::fn join_bin -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn leave_bin -->
|
|
Remove the elements of `self` from `bin`.
|
|
## `bin`
|
|
a `gst::Bin`
|
|
## `rtpbin`
|
|
a rtpbin `gst::Element`
|
|
|
|
# Returns
|
|
|
|
`true` on success.
|
|
<!-- trait RTSPStreamExt::fn query_position -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn query_stop -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn recv_rtcp -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn recv_rtp -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn remove_transport -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn request_aux_receiver -->
|
|
Creating a rtxreceive bin
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `sessid`
|
|
the session id
|
|
|
|
# Returns
|
|
|
|
a `gst::Element`.
|
|
<!-- trait RTSPStreamExt::fn request_aux_sender -->
|
|
Creating a rtxsend bin
|
|
## `sessid`
|
|
the session id
|
|
|
|
# Returns
|
|
|
|
a `gst::Element`.
|
|
<!-- trait RTSPStreamExt::fn request_ulpfec_decoder -->
|
|
Creating a rtpulpfecdec element
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
a `gst::Element`.
|
|
<!-- trait RTSPStreamExt::fn request_ulpfec_encoder -->
|
|
Creating a rtpulpfecenc element
|
|
|
|
Feature: `v1_16`
|
|
|
|
|
|
# Returns
|
|
|
|
a `gst::Element`.
|
|
<!-- trait RTSPStreamExt::fn reserve_address -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn seekable -->
|
|
Checks whether the individual `self` is seekable.
|
|
|
|
Feature: `v1_14`
|
|
|
|
|
|
# Returns
|
|
|
|
`true` if `self` is seekable, else `false`.
|
|
<!-- trait RTSPStreamExt::fn set_address_pool -->
|
|
configure `pool` to be used as the address pool of `self`.
|
|
## `pool`
|
|
a `RTSPAddressPool`
|
|
<!-- trait RTSPStreamExt::fn set_bind_mcast_address -->
|
|
Decide whether the multicast socket should be bound to a multicast address or
|
|
INADDR_ANY.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `bind_mcast_addr`
|
|
the new value
|
|
<!-- trait RTSPStreamExt::fn set_blocked -->
|
|
Blocks or unblocks the dataflow on `self`.
|
|
## `blocked`
|
|
boolean indicating we should block or unblock
|
|
|
|
# Returns
|
|
|
|
`true` on success
|
|
<!-- trait RTSPStreamExt::fn set_buffer_size -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn set_client_side -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn set_control -->
|
|
Set the control string in `self`.
|
|
## `control`
|
|
a control string
|
|
<!-- trait RTSPStreamExt::fn set_dscp_qos -->
|
|
Configure the dscp qos of the outgoing sockets to `dscp_qos`.
|
|
## `dscp_qos`
|
|
a new dscp qos value (0-63, or -1 to disable)
|
|
<!-- trait RTSPStreamExt::fn set_max_mcast_ttl -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn set_mtu -->
|
|
Configure the mtu in the payloader of `self` to `mtu`.
|
|
## `mtu`
|
|
a new MTU
|
|
<!-- trait RTSPStreamExt::fn set_multicast_iface -->
|
|
configure `multicast_iface` to be used for `self`.
|
|
## `multicast_iface`
|
|
a multicast interface name
|
|
<!-- trait RTSPStreamExt::fn set_profiles -->
|
|
Configure the allowed profiles for `self`.
|
|
## `profiles`
|
|
the new profiles
|
|
<!-- trait RTSPStreamExt::fn set_protocols -->
|
|
Configure the allowed lower transport for `self`.
|
|
## `protocols`
|
|
the new flags
|
|
<!-- trait RTSPStreamExt::fn set_pt_map -->
|
|
Configure a pt map between `pt` and `caps`.
|
|
## `pt`
|
|
the pt
|
|
## `caps`
|
|
a `gst::Caps`
|
|
<!-- trait RTSPStreamExt::fn set_publish_clock_mode -->
|
|
Sets if and how the stream clock should be published according to RFC7273.
|
|
## `mode`
|
|
the clock publish mode
|
|
<!-- trait RTSPStreamExt::fn set_rate_control -->
|
|
Define whether `self` will follow the Rate-Control=no behaviour as specified
|
|
in the ONVIF replay spec.
|
|
|
|
Feature: `v1_18`
|
|
|
|
<!-- trait RTSPStreamExt::fn set_retransmission_pt -->
|
|
Set the payload type (pt) for retransmission of this stream.
|
|
## `rtx_pt`
|
|
a `guint`
|
|
<!-- trait RTSPStreamExt::fn set_retransmission_time -->
|
|
Set the amount of time to store retransmission packets.
|
|
## `time`
|
|
a `gst::ClockTime`
|
|
<!-- trait RTSPStreamExt::fn set_ulpfec_percentage -->
|
|
Sets the amount of redundancy to apply when creating ULPFEC
|
|
protection packets.
|
|
|
|
Feature: `v1_16`
|
|
|
|
<!-- trait RTSPStreamExt::fn set_ulpfec_pt -->
|
|
Set the payload type to be used for ULPFEC protection packets
|
|
|
|
Feature: `v1_16`
|
|
|
|
<!-- trait RTSPStreamExt::fn transport_filter -->
|
|
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.
|
|
<!-- trait RTSPStreamExt::fn update_crypto -->
|
|
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
|
|
<!-- trait RTSPStreamExt::fn verify_mcast_ttl -->
|
|
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.
|
|
<!-- struct RTSPStreamTransport -->
|
|
A Transport description for a stream
|
|
|
|
# Implements
|
|
|
|
[`RTSPStreamTransportExt`](trait.RTSPStreamTransportExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html), [`RTSPStreamTransportExtManual`](prelude/trait.RTSPStreamTransportExtManual.html)
|
|
<!-- trait RTSPStreamTransportExt -->
|
|
Trait containing all `RTSPStreamTransport` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPStreamTransport`](struct.RTSPStreamTransport.html)
|
|
<!-- impl RTSPStreamTransport::fn new -->
|
|
Create a new `RTSPStreamTransport` that can be used to manage
|
|
`stream` with transport `tr`.
|
|
## `stream`
|
|
a `RTSPStream`
|
|
## `tr`
|
|
a GstRTSPTransport
|
|
|
|
# Returns
|
|
|
|
a new `RTSPStreamTransport`
|
|
<!-- trait RTSPStreamTransportExt::fn get_rtpinfo -->
|
|
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.
|
|
<!-- trait RTSPStreamTransportExt::fn get_stream -->
|
|
Get the `RTSPStream` used when constructing `self`.
|
|
|
|
# Returns
|
|
|
|
the stream used when constructing `self`.
|
|
<!-- trait RTSPStreamTransportExt::fn get_transport -->
|
|
Get the transport configured in `self`.
|
|
|
|
# Returns
|
|
|
|
the transport configured in `self`. It remains
|
|
valid for as long as `self` is valid.
|
|
<!-- trait RTSPStreamTransportExt::fn get_url -->
|
|
Get the url configured in `self`.
|
|
|
|
# Returns
|
|
|
|
the url configured in `self`.
|
|
It remains valid for as long as `self` is valid.
|
|
<!-- trait RTSPStreamTransportExt::fn is_timed_out -->
|
|
Check if `self` is timed out.
|
|
|
|
# Returns
|
|
|
|
`true` if `self` timed out.
|
|
<!-- trait RTSPStreamTransportExt::fn keep_alive -->
|
|
Signal the installed keep_alive callback for `self`.
|
|
<!-- trait RTSPStreamTransportExt::fn message_sent -->
|
|
Signal the installed message_sent / message_sent_full callback for `self`.
|
|
|
|
Feature: `v1_16`
|
|
|
|
<!-- trait RTSPStreamTransportExt::fn recv_data -->
|
|
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`.
|
|
<!-- trait RTSPStreamTransportExt::fn send_rtcp -->
|
|
Send `buffer` to the installed RTCP callback for `self`.
|
|
## `buffer`
|
|
a `gst::Buffer`
|
|
|
|
# Returns
|
|
|
|
`true` on success
|
|
<!-- trait RTSPStreamTransportExt::fn send_rtcp_list -->
|
|
Send `buffer_list` to the installed RTCP callback for `self`.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `buffer_list`
|
|
a `gst::Buffer`
|
|
|
|
# Returns
|
|
|
|
`true` on success
|
|
<!-- trait RTSPStreamTransportExt::fn send_rtp -->
|
|
Send `buffer` to the installed RTP callback for `self`.
|
|
## `buffer`
|
|
a `gst::Buffer`
|
|
|
|
# Returns
|
|
|
|
`true` on success
|
|
<!-- trait RTSPStreamTransportExt::fn send_rtp_list -->
|
|
Send `buffer_list` to the installed RTP callback for `self`.
|
|
|
|
Feature: `v1_16`
|
|
|
|
## `buffer_list`
|
|
a `gst::BufferList`
|
|
|
|
# Returns
|
|
|
|
`true` on success
|
|
<!-- trait RTSPStreamTransportExt::fn set_active -->
|
|
Activate or deactivate datatransfer configured in `self`.
|
|
## `active`
|
|
new state of `self`
|
|
|
|
# Returns
|
|
|
|
`true` when the state was changed.
|
|
<!-- trait RTSPStreamTransportExt::fn set_callbacks -->
|
|
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.
|
|
<!-- trait RTSPStreamTransportExt::fn set_keepalive -->
|
|
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.
|
|
<!-- trait RTSPStreamTransportExt::fn set_list_callbacks -->
|
|
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.
|
|
<!-- trait RTSPStreamTransportExt::fn set_message_sent -->
|
|
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
|
|
<!-- trait RTSPStreamTransportExt::fn set_message_sent_full -->
|
|
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
|
|
<!-- trait RTSPStreamTransportExt::fn set_timed_out -->
|
|
Set the timed out state of `self` to `timedout`
|
|
## `timedout`
|
|
timed out value
|
|
<!-- trait RTSPStreamTransportExt::fn set_transport -->
|
|
Set `tr` as the client transport. This function takes ownership of the
|
|
passed `tr`.
|
|
## `tr`
|
|
a client `gst_rtsp::RTSPTransport`
|
|
<!-- trait RTSPStreamTransportExt::fn set_url -->
|
|
Set `url` as the client url.
|
|
## `url`
|
|
a client `gst_rtsp::RTSPUrl`
|
|
<!-- enum RTSPSuspendMode -->
|
|
The suspend mode of the media pipeline. A media pipeline is suspended right
|
|
after creating the SDP and when the client performs a PAUSED request.
|
|
<!-- enum RTSPSuspendMode::variant None -->
|
|
Media is not suspended
|
|
<!-- enum RTSPSuspendMode::variant Pause -->
|
|
Media is PAUSED in suspend
|
|
<!-- enum RTSPSuspendMode::variant Reset -->
|
|
The media is set to NULL when suspended
|
|
<!-- struct RTSPThread -->
|
|
Structure holding info about a mainloop running in a thread
|
|
<!-- impl RTSPThread::fn new -->
|
|
Create a new thread object that can run a mainloop.
|
|
## `type_`
|
|
the thread type
|
|
|
|
# Returns
|
|
|
|
a `RTSPThread`.
|
|
<!-- impl RTSPThread::fn reuse -->
|
|
Reuse the mainloop of `self`
|
|
|
|
# Returns
|
|
|
|
`true` if the mainloop could be reused
|
|
<!-- impl RTSPThread::fn stop -->
|
|
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.
|
|
<!-- struct RTSPThreadPool -->
|
|
The thread pool structure.
|
|
|
|
# Implements
|
|
|
|
[`RTSPThreadPoolExt`](trait.RTSPThreadPoolExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
|
|
<!-- trait RTSPThreadPoolExt -->
|
|
Trait containing all `RTSPThreadPool` methods.
|
|
|
|
# Implementors
|
|
|
|
[`RTSPThreadPool`](struct.RTSPThreadPool.html)
|
|
<!-- impl RTSPThreadPool::fn new -->
|
|
Create a new `RTSPThreadPool` instance.
|
|
|
|
# Returns
|
|
|
|
a new `RTSPThreadPool`
|
|
<!-- impl RTSPThreadPool::fn cleanup -->
|
|
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.
|
|
<!-- trait RTSPThreadPoolExt::fn get_max_threads -->
|
|
Get the maximum number of threads used for client connections.
|
|
See `RTSPThreadPoolExt::set_max_threads`.
|
|
|
|
# Returns
|
|
|
|
the maximum number of threads.
|
|
<!-- trait RTSPThreadPoolExt::fn get_thread -->
|
|
Get a new `RTSPThread` for `type_` and `ctx`.
|
|
## `type_`
|
|
the `RTSPThreadType`
|
|
## `ctx`
|
|
a `RTSPContext`
|
|
|
|
# Returns
|
|
|
|
a new `RTSPThread`,
|
|
`RTSPThread::stop` after usage
|
|
<!-- trait RTSPThreadPoolExt::fn set_max_threads -->
|
|
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
|
|
<!-- enum RTSPThreadType -->
|
|
Different thread types
|
|
<!-- enum RTSPThreadType::variant Client -->
|
|
a thread to handle the client communication
|
|
<!-- enum RTSPThreadType::variant Media -->
|
|
a thread to handle media
|
|
<!-- struct RTSPToken -->
|
|
An opaque object used for checking authorisations.
|
|
It is generated after successful authentication.
|
|
<!-- impl RTSPToken::fn new -->
|
|
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.
|
|
<!-- impl RTSPToken::fn new_empty -->
|
|
Create a new empty Authorization token.
|
|
|
|
# Returns
|
|
|
|
a new empty authorization token.
|
|
<!-- impl RTSPToken::fn new_valist -->
|
|
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.
|
|
<!-- impl RTSPToken::fn get_string -->
|
|
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`.
|
|
<!-- impl RTSPToken::fn get_structure -->
|
|
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.
|
|
<!-- impl RTSPToken::fn is_allowed -->
|
|
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`.
|
|
<!-- impl RTSPToken::fn set_bool -->
|
|
Sets a boolean value on `self`.
|
|
|
|
Feature: `v1_14`
|
|
|
|
## `field`
|
|
field to set
|
|
## `bool_value`
|
|
boolean value to set
|
|
<!-- impl RTSPToken::fn set_string -->
|
|
Sets a string value on `self`.
|
|
|
|
Feature: `v1_14`
|
|
|
|
## `field`
|
|
field to set
|
|
## `string_value`
|
|
string value to set
|
|
<!-- impl RTSPToken::fn writable_structure -->
|
|
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.
|