Trait gstreamer_rtsp_server::prelude::RTSPMediaExt

source ·

pub trait RTSPMediaExt:
    IsA<RTSPMedia>
    + Sealed
    + 'static {
Show 94 methods    // Provided methods
    fn can_be_shared(&self) -> bool { ... }
    fn collect_streams(&self) { ... }
    fn create_stream(
        &self,
        payloader: &impl IsA<Element>,
        pad: &impl IsA<Pad>,
    ) -> RTSPStream { ... }
    fn find_stream(&self, control: &str) -> Option<RTSPStream> { ... }
    fn address_pool(&self) -> Option<RTSPAddressPool> { ... }
    fn base_time(&self) -> Option<ClockTime> { ... }
    fn buffer_size(&self) -> u32 { ... }
    fn clock(&self) -> Option<Clock> { ... }
    fn does_retransmission(&self) -> bool { ... }
    fn dscp_qos(&self) -> i32 { ... }
    fn element(&self) -> Element { ... }
    fn is_ensure_keyunit_on_start(&self) -> bool { ... }
    fn ensure_keyunit_on_start_timeout(&self) -> u32 { ... }
    fn latency(&self) -> u32 { ... }
    fn max_mcast_ttl(&self) -> u32 { ... }
    fn multicast_iface(&self) -> Option<GString> { ... }
    fn profiles(&self) -> RTSPProfile { ... }
    fn protocols(&self) -> RTSPLowerTrans { ... }
    fn publish_clock_mode(&self) -> RTSPPublishClockMode { ... }
    fn range_string(&self, play: bool, unit: RTSPRangeUnit) -> Option<GString> { ... }
    fn is_rate_control(&self) -> bool { ... }
    fn rates(&self) -> Option<(f64, f64)> { ... }
    fn retransmission_time(&self) -> Option<ClockTime> { ... }
    fn status(&self) -> RTSPMediaStatus { ... }
    fn stream(&self, idx: u32) -> Option<RTSPStream> { ... }
    fn suspend_mode(&self) -> RTSPSuspendMode { ... }
    fn time_provider(
        &self,
        address: Option<&str>,
        port: u16,
    ) -> Option<NetTimeProvider> { ... }
    fn transport_mode(&self) -> RTSPTransportMode { ... }
    fn has_completed_sender(&self) -> bool { ... }
    fn is_bind_mcast_address(&self) -> bool { ... }
    fn is_eos_shutdown(&self) -> bool { ... }
    fn is_receive_only(&self) -> bool { ... }
    fn is_reusable(&self) -> bool { ... }
    fn is_shared(&self) -> bool { ... }
    fn is_stop_on_disconnect(&self) -> bool { ... }
    fn is_time_provider(&self) -> bool { ... }
    fn lock(&self) { ... }
    fn n_streams(&self) -> u32 { ... }
    fn prepare(&self, thread: Option<RTSPThread>) -> Result<(), BoolError> { ... }
    fn set_address_pool(&self, pool: Option<&impl IsA<RTSPAddressPool>>) { ... }
    fn set_bind_mcast_address(&self, bind_mcast_addr: bool) { ... }
    fn set_buffer_size(&self, size: u32) { ... }
    fn set_clock(&self, clock: Option<&impl IsA<Clock>>) { ... }
    fn set_do_retransmission(&self, do_retransmission: bool) { ... }
    fn set_dscp_qos(&self, dscp_qos: i32) { ... }
    fn set_ensure_keyunit_on_start(&self, ensure_keyunit_on_start: bool) { ... }
    fn set_ensure_keyunit_on_start_timeout(&self, timeout: u32) { ... }
    fn set_eos_shutdown(&self, eos_shutdown: bool) { ... }
    fn set_latency(&self, latency: u32) { ... }
    fn set_max_mcast_ttl(&self, ttl: u32) -> bool { ... }
    fn set_multicast_iface(&self, multicast_iface: Option<&str>) { ... }
    fn set_pipeline_state(&self, state: State) { ... }
    fn set_profiles(&self, profiles: RTSPProfile) { ... }
    fn set_protocols(&self, protocols: RTSPLowerTrans) { ... }
    fn set_publish_clock_mode(&self, mode: RTSPPublishClockMode) { ... }
    fn set_rate_control(&self, enabled: bool) { ... }
    fn set_retransmission_time(&self, time: impl Into<Option<ClockTime>>) { ... }
    fn set_reusable(&self, reusable: bool) { ... }
    fn set_shared(&self, shared: bool) { ... }
    fn set_state(
        &self,
        state: State,
        transports: &[RTSPStreamTransport],
    ) -> bool { ... }
    fn set_stop_on_disconnect(&self, stop_on_disconnect: bool) { ... }
    fn set_suspend_mode(&self, mode: RTSPSuspendMode) { ... }
    fn set_transport_mode(&self, mode: RTSPTransportMode) { ... }
    fn suspend(&self) -> Result<(), BoolError> { ... }
    fn unlock(&self) { ... }
    fn unprepare(&self) -> Result<(), BoolError> { ... }
    fn unsuspend(&self) -> Result<(), BoolError> { ... }
    fn use_time_provider(&self, time_provider: bool) { ... }
    fn get_property_bind_mcast_address(&self) -> bool { ... }
    fn set_time_provider(&self, time_provider: bool) { ... }
    fn connect_handle_message<F: Fn(&Self, &Message) -> bool + Send + Sync + 'static>(
        &self,
        detail: Option<&str>,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_new_state<F: Fn(&Self, i32) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_new_stream<F: Fn(&Self, &RTSPStream) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_prepared<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_removed_stream<F: Fn(&Self, &RTSPStream) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_target_state<F: Fn(&Self, i32) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_unprepared<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_bind_mcast_address_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_buffer_size_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_clock_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_dscp_qos_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_ensure_keyunit_on_start_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_ensure_keyunit_on_start_timeout_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_eos_shutdown_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_latency_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_max_mcast_ttl_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_profiles_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_protocols_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_reusable_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_shared_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_stop_on_disconnect_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_suspend_mode_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_time_provider_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
    fn connect_transport_mode_notify<F: Fn(&Self) + Send + Sync + 'static>(
        &self,
        f: F,
    ) -> SignalHandlerId { ... }
}

Expand description

Trait containing all RTSPMedia methods.

§Implementors

RTSPMedia, RTSPOnvifMedia

Provided Methods§

source

fn can_be_shared(&self) -> bool

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

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

The function must be called with lock().

§Returns

true if the media can be shared between clients.

source

fn collect_streams(&self)

Find all payloader elements, they should be named pay`din the element ofself, 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`din the element ofself, and create GstRTSPStreams` for them.

source

fn create_stream( &self, payloader: &impl IsA<Element>, pad: &impl IsA<Pad>, ) -> RTSPStream

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.

source

fn find_stream(&self, control: &str) -> Option<RTSPStream>

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.

source

fn address_pool(&self) -> Option<RTSPAddressPool>

Get the RTSPAddressPool used as the address pool of self.

§Returns

the RTSPAddressPool of self. g_object_unref() after usage.

source

fn base_time(&self) -> Option<ClockTime>

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.

source

fn buffer_size(&self) -> u32

Get the kernel UDP buffer size.

§Returns

the kernel UDP buffer size.

source

fn clock(&self) -> Option<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.

source

fn does_retransmission(&self) -> bool

§Returns

Whether retransmission requests will be sent

source

fn dscp_qos(&self) -> i32

Get the configured DSCP QoS of attached media.

§Returns

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

source

fn element(&self) -> Element

Get the element that was used when constructing self.

§Returns

a gst::Element. Unref after usage.

source

fn is_ensure_keyunit_on_start(&self) -> bool

Get ensure-keyunit-on-start flag.

§Returns

The ensure-keyunit-on-start flag.

source

fn ensure_keyunit_on_start_timeout(&self) -> u32

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

§Returns

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

source

fn latency(&self) -> u32

Get the latency that is used for receiving media.

§Returns

latency in milliseconds

source

fn max_mcast_ttl(&self) -> u32

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

§Returns

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

source

fn multicast_iface(&self) -> Option<GString>

Get the multicast interface used for self.

§Returns

the multicast interface for self. g_free() after usage.

source

fn profiles(&self) -> RTSPProfile

Get the allowed profiles of self.

§Returns

a gst_rtsp::RTSPProfile

source

fn protocols(&self) -> RTSPLowerTrans

Get the allowed protocols of self.

§Returns

a gst_rtsp::RTSPLowerTrans

source

fn publish_clock_mode(&self) -> RTSPPublishClockMode

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

§Returns

The GstRTSPPublishClockMode

source

fn range_string(&self, play: bool, unit: RTSPRangeUnit) -> Option<GString>

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.

source

fn is_rate_control(&self) -> bool

§Returns

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

source

fn rates(&self) -> Option<(f64, f64)>

Get the rate and 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.

§`rate`

the rate of the current segment

§`applied_rate`

the applied_rate of the current segment

source

fn retransmission_time(&self) -> Option<ClockTime>

Get the amount of time to store retransmission data.

§Returns

the amount of time to store retransmission data.

source

fn status(&self) -> RTSPMediaStatus

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.

source

fn stream(&self, idx: u32) -> Option<RTSPStream>

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.

source

fn suspend_mode(&self) -> RTSPSuspendMode

Get how self will be suspended.

§Returns

RTSPSuspendMode.

source

fn time_provider( &self, address: Option<&str>, port: u16, ) -> Option<NetTimeProvider>

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.

source

fn transport_mode(&self) -> RTSPTransportMode

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

§Returns

The transport mode.

source

fn has_completed_sender(&self) -> bool

See RTSPStreamExt::is_complete(), RTSPStreamExt::is_sender().

§Returns

whether self has at least one complete sender stream.

source

fn is_bind_mcast_address(&self) -> bool

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

§Returns

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

source

fn is_eos_shutdown(&self) -> bool

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.

source

fn is_receive_only(&self) -> bool

§Returns

true if self is receive-only, false otherwise.

source

fn is_reusable(&self) -> bool

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

§Returns

true if the media can be reused

source

fn is_shared(&self) -> bool

Check if the pipeline for self can be shared between multiple clients in theory. This simply returns the value set via set_shared().

To know if a media can be shared in practice, i.e. if it’s shareable and either reusable or was never unprepared before, use can_be_shared().

§Returns

true if the media can be shared between clients.

source

fn is_stop_on_disconnect(&self) -> bool

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.

source

fn is_time_provider(&self) -> bool

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

Use time_provider() to get the network clock.

§Returns

true if self can provide a gst_net::NetTimeProvider.

source

fn lock(&self)

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.

source

fn n_streams(&self) -> u32

Get the number of streams in this media.

§Returns

The number of streams.

source

fn prepare(&self, thread: Option<RTSPThread>) -> Result<(), BoolError>

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

source

fn set_address_pool(&self, pool: Option<&impl IsA<RTSPAddressPool>>)

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

§`pool`

a RTSPAddressPool

source

fn set_bind_mcast_address(&self, bind_mcast_addr: bool)

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

§`bind_mcast_addr`

the new value

source

fn set_buffer_size(&self, size: u32)

Set the kernel UDP buffer size.

§`size`

the new value

source

fn set_clock(&self, clock: Option<&impl IsA<Clock>>)

Configure the clock used for the media.

§`clock`

gst::Clock to be used

source

fn set_do_retransmission(&self, do_retransmission: bool)

Set whether retransmission requests will be sent

source

fn set_dscp_qos(&self, dscp_qos: i32)

Configure the dscp qos of attached streams to dscp_qos.

§`dscp_qos`

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

source

fn set_ensure_keyunit_on_start(&self, ensure_keyunit_on_start: bool)

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

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

§`ensure_keyunit_on_start`

the new value

source

fn set_ensure_keyunit_on_start_timeout(&self, timeout: u32)

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

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

§`timeout`

the new value

source

fn set_eos_shutdown(&self, eos_shutdown: bool)

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

§`eos_shutdown`

the new value

source

fn set_latency(&self, latency: u32)

Configure the latency used for receiving media.

§`latency`

latency in milliseconds

source

fn set_max_mcast_ttl(&self, ttl: u32) -> bool

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

§`ttl`

the new multicast ttl value

§Returns

true if the requested ttl has been set successfully.

source

fn set_multicast_iface(&self, multicast_iface: Option<&str>)

configure multicast_iface to be used for self.

§`multicast_iface`

a multicast interface name

source

fn set_pipeline_state(&self, state: State)

Set the state of the pipeline managed by self to state

§`state`

the target state of the pipeline

source

fn set_profiles(&self, profiles: RTSPProfile)

Configure the allowed lower transport for self.

§`profiles`

the new flags

source

fn set_protocols(&self, protocols: RTSPLowerTrans)

Configure the allowed lower transport for self.

§`protocols`

the new flags

source

fn set_publish_clock_mode(&self, mode: RTSPPublishClockMode)

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

§`mode`

the clock publish mode

source

fn set_rate_control(&self, enabled: bool)

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

source

fn set_retransmission_time(&self, time: impl Into<Option<ClockTime>>)

Set the amount of time to store retransmission packets.

§`time`

the new value

source

fn set_reusable(&self, reusable: bool)

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

§`reusable`

the new value

source

fn set_shared(&self, shared: bool)

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

source

fn set_state(&self, state: State, transports: &[RTSPStreamTransport]) -> bool

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

self must be prepared with prepare();

§`state`

the target state of the media

§`transports`

a GPtrArray of RTSPStreamTransport pointers

§Returns

true on success.

source

fn set_stop_on_disconnect(&self, stop_on_disconnect: bool)

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

§`stop_on_disconnect`

the new value

source

fn set_suspend_mode(&self, mode: RTSPSuspendMode)

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

source

fn set_transport_mode(&self, mode: RTSPTransportMode)

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

§`mode`

the new value

source

fn suspend(&self) -> Result<(), BoolError>

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 unsuspend()

self must be prepared with prepare();

§Returns

true on success.

source

fn unlock(&self)

Unlock the media.

source

fn unprepare(&self) -> Result<(), BoolError>

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.