SrtOutputSettingsCommon interface
The settings for an SRT output see: NorskOutput.srt()
Signature:
export interface SrtOutputSettingsCommon extends SinkNodeSettings<SrtOutputNode>, StreamStatisticsMixin
Extends: SinkNodeSettings<SrtOutputNode>, StreamStatisticsMixin
Events
| Property | Type | Description |
|---|---|---|
( streamId: string, streamIndex: number, remoteHost: string, parsed?: SrtStreamIdParsed) ⇒ void |
(Optional) On connect callback, notifying that a connection has been established. In caller mode this is raised for the single outbound caller, while in listener mode this means a new caller has connected (possibly one of many) and provides the stream_id that was set on the socket |
|
(rejection: { streamId: string; route: string; user?: string; remoteHost: string; reason: string; }) ⇒ void |
(Optional) A consumer refused at the CRYPTO layer by this egest’s (route, user) grant table — the handshake was rejected before SrtOutputSettingsCommon.onConnection could run. Reasons: "unknown_route" | "unknown_user". A WRONG passphrase fails later in SRT’s key-material exchange and produces no event. |
|
(stats: SrtOutputConnectionStats) ⇒ void |
(Optional) Each consumer’s SRT transport stats from the sending side — RTT, what was sent, lost and retransmitted — every SrtOutputSettingsCommon.statsPollIntervalMs (default 1 s). For a bonded consumer, each link’s figures are in |
Properties
| Property | Type | Description |
|---|---|---|
number |
(Optional) A/V delay in milliseconds - to allow inclusion of subtitles, metadata and other ancillary data. May be set to 0 if these are not present to reduce latency |
|
number |
(Optional) Input bandwidth (SRTO_INPUTBW) |
|
(Optional) Jitter buffer configuration |
||
number |
(Optional) Max bandwidth (SRTO_MAXBW) |
|
number |
(Optional) Maximum duration to hold any stream in the case that one or more are running behind. In the case the late stream is delayed rather than having a large gap, this will result in discarded input frames. |
|
(info: { streamIndex: number; }) ⇒ void |
(Optional) On disconnect callback, notifying that an individual client connection has disconnected. This will be followed by a retry in caller mode, while in listener mode is relevant to an individual client (per the provided metadata), while the remaining clients remain connected. |
|
number |
(Optional) Overhead bandwidth (SRTO_OHEADBW) |
|
number |
(Optional) The latency value provided by the sender side as a minimum value for the receiver (SRTO_PEERLATENCY) |
|
number |
(Optional) The latency value in the receiving direction of the socket (SRTO_RCVLATENCY) |
|
number |
(Optional) Interval in milliseconds at which to sample SRT socket statistics. Stats feed metrics, so tune this to the rate at which you want metric updates. When unspecified a default is used. |
|
number |
(Optional) Interval in milliseconds at which to sample the underlying SRT socket’s connection state for the per-connection Note this does not drive SrtOutputSettingsCommon.onConnection or disconnection events: those are driven by actual connection establishment / disconnection detection (a send/receive failure or the peer idle timeout), not by this poll. |
|
boolean |
(Optional) Whether to enable the too-late-packet-drop mechanism (SRTO_TLPKTDROP) - default true. On a sender this means dropping packets which have no chance to be delivered. Enabled by default when the receiver supports it. |
|
(Optional) |
||
(Optional) File to write through all data to (this will be read from in order to produce the final SRT stream, useful for compliance purposes) NB: This will rollover automatically, but no expiry will take place, it is assumed that infrastructure will take care of this |