@relaymessenger/livekit 0.1.0-staging.16 → 0.1.0-staging.17

package/dist/transport.d.ts

@@ -1,327 +0,0 @@
-import type { CallRoomEndedFrame, CallRoomOptions, CallRoomStateFrame, Relay } from "@relaymessenger/sdk";
-import { type LocalVideoTrack, type RelayVideoReceiverStats, type RelayVideoSenderStats, RemoteVideoTrack, type TrackPublishOptions } from "./video.js";
-export { LocalVideoTrack, RemoteVideoTrack, VideoBufferType, VideoCodec, VideoFrame, VideoRotation, VideoSource, VideoStream, type RelayVideoReceiverStats, type RelayVideoSenderStats, type TrackPublishOptions, type VideoEncoding, type VideoFrameEvent, type VideoStreamOptions, } from "./video.js";
-export interface RelayAudioFrame {
-    /** Interleaved signed PCM16 samples. */
-    samples: Int16Array;
-    sampleRate: number;
-    channelCount: number;
-}
-/** Standard `RTCIceServer` shape: STUN or TURN URLs with optional credentials. */
-export interface RelayIceServer {
-    urls: string | string[];
-    username?: string;
-    credential?: string;
-}
-export type RelayIceTransportPolicy = "all" | "relay";
-/**
- * Returns the ICE servers for one peer connection. Called before the first
- * peer and again before every restart, so short-lived TURN credentials can be
- * minted per attempt (PartyTracks: a reconnect "will trigger new sessionId,
- * new ice server credentials and a new peerConnection").
- */
-export type RelayIceServersProvider = (attempt: {
-    /** 0 for the first peer, then 1, 2, ... for each restart. */
-    restarts: number;
-}) => RelayIceServer[] | Promise<RelayIceServer[]>;
-/** Why the transport replaced its peer connection with a new SFU session. */
-export type RelayCallRestartReason = "timeout" | "failed" | "disconnected" | "error";
-export interface RelayCallRestartEvent {
-    reason: RelayCallRestartReason;
-    /** `diagnostics().summary` of the session being replaced, taken as it was given up. */
-    summary: string;
-    /** Restarts since `connect()`, this one included. */
-    restarts: number;
-    /** Backoff waited before the new peer was built. */
-    delayMs: number;
-}
-/**
- * PCM format the engine decodes the remote participant's Opus into. The values
- * are the ones libopus decodes to (`@evan/opus` lib.d.ts: `channels?: 1 | 2`,
- * `sample_rate?: 8000 | 12000 | 16000 | 24000 | 48000`); the decoder itself
- * resamples and downmixes, so no PCM is converted by hand.
- */
-export interface RelayInboundAudioFormat {
-    sampleRate: 8000 | 12000 | 16000 | 24000 | 48000;
-    channelCount: 1 | 2;
-}
-/** Opus's native rate and Relay's wire channel count. */
-export declare const DEFAULT_INBOUND_AUDIO: Readonly<RelayInboundAudioFormat>;
-/** Counters for both video directions; `undefined` for a direction that has no track. */
-export interface RelayCallVideoStats {
-    outbound: RelayVideoSenderStats | undefined;
-    inbound: RelayVideoReceiverStats | undefined;
-}
-export type RelayCallEngine = "werift" | "wrtc";
-export interface RelayCallTransportOptions {
-    relay: Relay;
-    callId: string;
-    room?: CallRoomOptions;
-    /**
-     * WebRTC engine. `"werift"` (default) is pure TypeScript plus prebuilt Opus
-     * and loads no native binding. `"wrtc"` loads `@roamhq/wrtc` on demand.
-     */
-    engine?: RelayCallEngine;
-    /**
-     * STUN and TURN servers handed to the engine's `RTCPeerConnection`. Defaults
-     * to the servers the Call room sends (its `iceServers` frame: Cloudflare STUN
-     * plus TURN credentials Relay mints), read again before every restart; when
-     * the room sends none, Cloudflare's STUN server, the configuration of
-     * Cloudflare's own Realtime echo example. Set this only to replace Relay's
-     * servers. A function is called before every peer connection, restarts
-     * included, so it can mint fresh TURN credentials each time.
-     */
-    iceServers?: RelayIceServer[] | RelayIceServersProvider;
-    /** `"relay"` forces every candidate through TURN. Defaults to `"all"`. */
-    iceTransportPolicy?: RelayIceTransportPolicy;
-    /**
-     * How long one SFU session has, after its answer is applied, to reach
-     * `connected` before the transport restarts onto a new session. Defaults to
-     * 5 seconds. `connect()` itself has no deadline: it waits through restarts
-     * until media connects, the Call ends, `close()` is called, or its signal aborts.
-     */
-    sessionConnectTimeoutMs?: number;
-    /** @deprecated Use `sessionConnectTimeoutMs`; now the per-session wait, not a `connect()` deadline. */
-    mediaConnectTimeoutMs?: number;
-    /** @deprecated Use `sessionConnectTimeoutMs`. */
-    connectionTimeoutMs?: number;
-    /**
-     * PCM format of the `audio` events: the Opus decoder decodes the remote
-     * track straight to this rate and channel count. Defaults to 48 kHz stereo.
-     * The `"wrtc"` engine delivers libwebrtc's own format and accepts only the default.
-     */
-    inboundAudio?: RelayInboundAudioFormat;
-    /**
-     * Called once per call when outbound audio is queued but no RTP packet has
-     * been written for 2 s while media is connected. Receives the diagnostics
-     * summary. Defaults to a no-op; nothing is restarted.
-     */
-    onWarning?: (message: string) => void;
-}
-/** Packet counts for the other participant's track, as seen by this participant. */
-export interface RelayCallInboundDiagnostics {
-    /** RTP packets received on the subscribed track. */
-    rtpPackets: number;
-    /** Opus packets the decoder rejected. */
-    decodeFailures: number;
-    /** PCM frames handed to the `audio` listeners. */
-    frames: number;
-    /** ms since `connect()` for the first and last RTP packet; undefined until one arrives. */
-    firstPacketAtMs: number | undefined;
-    lastPacketAtMs: number | undefined;
-    /** RTP packets received in the 5 s before `diagnostics()` was called. */
-    recentRtpPackets: number;
-}
-/** Packet counts for this participant's published track. */
-export interface RelayCallOutboundDiagnostics {
-    /** PCM slices accepted from the caller and handed to the engine source. */
-    frames: number;
-    opusPackets: number;
-    /** RTP packets carrying the caller's audio. */
-    rtpPackets: number;
-    /** RTP packets carrying Opus silence, written while nothing was queued. */
-    silencePackets: number;
-    /** ms since `connect()` for the first and last RTP packet of either kind. */
-    firstPacketAtMs: number | undefined;
-    lastPacketAtMs: number | undefined;
-    /** RTP packets of either kind written in the 5 s before `diagnostics()` was called. */
-    recentRtpPackets: number;
-    /** Encoded packets waiting for the 20 ms pacer. */
-    queued: number;
-    /** `undefined` when the engine does not expose its pacer (`wrtc`). */
-    pacerAlive: boolean | undefined;
-}
-/** Room signaling frames counted since `connect()`. */
-export interface RelayCallRoomDiagnostics {
-    roomStates: number;
-    /** Subscription (pull) offers received from the room. */
-    offers: number;
-    endedReason: string | undefined;
-    /** `error` frame messages, in order. */
-    errors: string[];
-}
-/** ICE facts recorded for one call, for logs and for the connect timeout error. */
-export interface RelayCallIceDiagnostics {
-    /** Local candidates gathered, counted by type. */
-    local: {
-        host: number;
-        srflx: number;
-        relay: number;
-        other: number;
-    };
-    /** Remote candidates from the SFU answer: transport and port only, never the address. */
-    remote: Array<{
-        transport: string;
-        port: number;
-    }>;
-    /** ICE gathering, ICE connection and peer connection state changes since `connect()`. */
-    transitions: Array<{
-        kind: "gathering" | "ice" | "connection";
-        state: string;
-        atMs: number;
-    }>;
-    /**
-     * The local side of the candidate pair media flows on, as `"<type> <protocol>"`
-     * (for example `"relay udp"`), read from the W3C stats once the peer connects;
-     * undefined until then or when the engine has no `getStats()`.
-     */
-    selectedPair: string | undefined;
-    connected: boolean;
-    inbound: RelayCallInboundDiagnostics;
-    outbound: RelayCallOutboundDiagnostics;
-    room: RelayCallRoomDiagnostics;
-    /** Peer connections replaced by a new SFU session since `connect()`. */
-    restarts: number;
-    /** One-line rendering of the fields above. */
-    summary: string;
-}
-export interface RelayCallTransportCloseEvent {
-    code: number;
-    reason: string;
-    wasClean: boolean;
-}
-type TransportEventMap = {
-    audio: [RelayAudioFrame];
-    connected: [];
-    /** A new peer connection published on a new SFU session; audio continues on it. */
-    restarted: [RelayCallRestartEvent];
-    roomState: [CallRoomStateFrame];
-    /** The person's camera started (`true`) or stopped (`false`) sending, from `roomState`. */
-    remoteVideo: [boolean];
-    /**
-     * The person's video track reached this peer (LiveKit `RoomEvent.TrackSubscribed`);
-     * read it with `new VideoStream(track)`. Once per call: restarts reuse the track.
-     */
-    trackSubscribed: [RemoteVideoTrack];
-    /** The call is over; every `VideoStream` of the track has ended. */
-    trackUnsubscribed: [RemoteVideoTrack];
-    /**
-     * Once per call: the person's audio has reached this peer (first inbound
-     * frame on the pulled track) and `roomState` shows the person connected. The
-     * room pulls both directions together, so from here the person hears what is
-     * written; start speaking after this, as LiveKit Agents start a session once
-     * the participant is in the room.
-     */
-    peerAudio: [];
-    ended: [CallRoomEndedFrame];
-    error: [Error];
-    close: [RelayCallTransportCloseEvent];
-};
-type TransportEvent = keyof TransportEventMap;
-type TransportListener<K extends TransportEvent> = (...args: TransportEventMap[K]) => void;
-/**
- * Restart rule, copied from PartyTracks and Cloudflare (PROTOCOL.md section 4):
- * not `connected` 5 s after the SFU answer (Cloudflare's echo example waits
- * 5000 ms; SFU operations block up to 5 s awaiting `connected`), `failed`, or
- * `disconnected` for 7 s (PartyTracks.ts `timeoutSeconds = 7`). Backoff 250 ms
- * x1.1 per attempt, capped at 10 s (PartyTracks `retryWithBackoff`
- * `backoffFactor: 1.1`, rxjs-helpers.ts defaults). werift never reports
- * `failed` on a session whose checks go unanswered (ice.js:983-986), so the
- * 5 s timer is the trigger that fires in practice.
- */
-export declare const RESTART_CONNECT_TIMEOUT_MS = 5000;
-export declare const RESTART_DISCONNECTED_MS = 7000;
-export declare const RESTART_INITIAL_DELAY_MS = 250;
-export declare const RESTART_BACKOFF_FACTOR = 1.1;
-export declare const RESTART_MAX_DELAY_MS = 10000;
-/**
- * Backoff before restart number `attempt` (1-based) counted since the last
- * session that connected. PartyTracks passes `resetOnSuccess: true` to rxjs
- * `retry` (rxjs-helpers.ts:18, :39); this transport reads "success" as a
- * session reaching `connected`.
- */
-export declare const restartDelayMs: (attempt: number) => number;
-export declare class RelayCallTransportError extends Error {
-    readonly code?: string;
-    constructor(message: string, code?: string);
-}
-/**
- * Provider-neutral Node WebRTC bridge for Relay Call rooms.
- *
- * The class owns Relay media negotiation internally. Higher-level adapters only
- * exchange PCM16 frames and call lifecycle events. When an SFU session never
- * connects or dies, the transport builds a new peer connection on a new
- * session (PROTOCOL.md section 4); the audio source and the `audio` events
- * carry on across the swap, so adapters only hear silence.
- */
-export declare class RelayCallTransport {
-    #private;
-    constructor(options: RelayCallTransportOptions);
-    on<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
-    off<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
-    /**
-     * Join the room, publish, and resolve on the first `connected`. Dead SFU
-     * sessions are replaced as they are found (PROTOCOL.md section 4), with no
-     * overall deadline: this rejects only when the Call ends, the room or
-     * transport closes, the room reports an error, or `signal` aborts (which
-     * also closes the transport).
-     */
-    connect(options?: {
-        signal?: AbortSignal;
-    }): Promise<void>;
-    /** Replace only the room signaling socket and replay the identical publication. */
-    reconnect(): Promise<void>;
-    /**
-     * ICE candidates, state transitions, packet counts in both directions,
-     * room frame counts and restarts recorded for this call, with a one-line summary.
-     */
-    diagnostics(): RelayCallIceDiagnostics;
-    /**
-     * Feed interleaved PCM16 audio into Relay. Frames are split into 10 ms WebRTC
-     * source slices and handed to the engine at once; the engine's 20 ms pump
-     * paces the wire, so adapters may push faster than real time (LiveKit's
-     * `AudioSource.captureFrame` shape). Resolves once the slices are queued;
-     * `waitForPlayout()` tells when they have left.
-     */
-    writeAudio(frame: RelayAudioFrame): Promise<void>;
-    /** Milliseconds of audio accepted by `writeAudio` but not yet written to RTP. */
-    queuedAudioMs(): number;
-    /**
-     * Resolves when every accepted slice has been written to RTP and the engine's
-     * pump is idle; immediately when nothing is queued; early on `clearAudio()`
-     * or `close()` (the caller reads `queuedAudioMs()` to learn what was dropped).
-     */
-    waitForPlayout(): Promise<void>;
-    /** Drop outgoing PCM that has not reached RTP and release `waitForPlayout()` callers. */
-    clearAudio(): void;
-    /**
-     * Resolves once `peerAudio` has fired (at once if it already has). Rejects
-     * after `timeoutMs`, or when the Call ends or the transport closes first.
-     */
-    waitForPeerAudio(timeoutMs: number): Promise<void>;
-    /**
-     * True once the person is receiving this transport's audio: a roomState
-     * that arrived after this peer's publish answer was applied lists `audio` in
-     * the person's `receiving` (PROTOCOL.md section 6b). False again from a
-     * restart until the new session is pulled.
-     */
-    get subscribed(): boolean;
-    /**
-     * Resolves once `subscribed` is true (at once if it is). Copy of LiveKit's
-     * `publication.waitForSubscription()`, which its room audio output awaits
-     * before playing a frame. Rejects when the Call ends or the transport closes.
-     */
-    waitForSubscription(): Promise<void>;
-    setMuted(muted: boolean): void;
-    /**
-     * Publish the camera (LiveKit `LocalParticipant.publishTrack`). The first
-     * call adds a `video` track to the SFU session with an add-track offer, then
-     * announces `userUpdate { video: true }`; later calls, after
-     * `unpublishTrack`, only resume sending and announce it again
-     * (PROTOCOL.md sections 1-2). Requires `connect()` and the werift engine.
-     */
-    publishTrack(track: LocalVideoTrack, options?: TrackPublishOptions): Promise<void>;
-    /**
-     * Stop sending the camera and announce `userUpdate { video: false }`. The
-     * track stays negotiated on the SFU (PROTOCOL.md section 1), so a later
-     * `publishTrack` with the same track resumes it without renegotiating.
-     */
-    unpublishTrack(track: LocalVideoTrack): Promise<void>;
-    /** The person's video track, once it has reached this peer. */
-    get remoteVideoTrack(): RemoteVideoTrack | undefined;
-    /** Frame, packet and keyframe counters for both video directions. */
-    videoStats(): RelayCallVideoStats;
-    end(): void;
-    close(): void;
-}
-//# sourceMappingURL=transport.d.ts.map

package/dist/transport.d.ts.map

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