@framers/agentos 0.1.113 → 0.1.114

package/dist/voice-pipeline/VoicePipelineOrchestrator.js

@@ -22,7 +22,7 @@
  * - The orchestrator does NOT resolve providers from ExtensionManager yet.
  *   All components must be injected via {@link VoicePipelineOverrides}.
  *   ExtensionManager integration is a planned future task.
- * - Event wiring is done once during {@link startSession} and never rewired.
+ * - Event wiring is done once during `startSession()` and never rewired.
  *   The transport/STT/TTS sessions are immutable for the session's lifetime.
  * - A watchdog timer prevents the pipeline from staying in LISTENING forever
  *   if the user walks away (default 30 s). The watchdog resets after each
@@ -46,7 +46,7 @@ import { randomUUID } from 'node:crypto';
  * | `'state_changed'` | `{ from: PipelineState, to: PipelineState }`  |
  * | `'turn_complete'` | {@link TurnCompleteEvent}                     |
  *
- * @see {@link VoicePipelineSession} for the public session interface returned by {@link startSession}.
+ * @see {@link VoicePipelineSession} for the public session interface returned by `startSession()`.
  */
 export class VoicePipelineOrchestrator extends EventEmitter {
     // --------------------------------------------------------------------------
@@ -71,7 +71,7 @@ export class VoicePipelineOrchestrator extends EventEmitter {
     /**
      * Create a new orchestrator with the given pipeline configuration.
      *
-     * The orchestrator starts in `'idle'` state. Call {@link startSession}
+     * The orchestrator starts in `'idle'` state. Call `startSession()`
      * to wire up components and transition to `'listening'`.
      *
      * @param config - Top-level pipeline configuration specifying providers and options.
@@ -82,13 +82,13 @@ export class VoicePipelineOrchestrator extends EventEmitter {
         // --------------------------------------------------------------------------
         // Private state
         // --------------------------------------------------------------------------
-        /** Current pipeline state. Transitions are managed exclusively by {@link _setState}. */
+        /** Current pipeline state. Transitions are managed exclusively by the internal state setter. */
         this._state = 'idle';
-        /** Active STT session created during {@link startSession}. Null when idle or closed. */
+        /** Active STT session created during `startSession()`. Null when idle or closed. */
         this._sttSession = null;
-        /** Active TTS session created during {@link startSession}. Null when idle or closed. */
+        /** Active TTS session created during `startSession()`. Null when idle or closed. */
         this._ttsSession = null;
-        /** The endpoint detector wired during {@link startSession}. Null when idle or closed. */
+        /** The endpoint detector wired during `startSession()`. Null when idle or closed. */
         this._endpointDetector = null;
         /** The barge-in handler consulted when speech is detected during SPEAKING. Null when idle or closed. */
         this._bargeinHandler = null;
@@ -539,7 +539,7 @@ export class VoicePipelineOrchestrator extends EventEmitter {
      * Transition to a new pipeline state, emitting a `'state_changed'` event.
      *
      * No-ops if the target state equals the current state (idempotent).
-     * This is the ONLY method that mutates {@link _state}, ensuring all
+     * This is the ONLY method that mutates the internal `_state`, ensuring all
      * transitions are observable via the `'state_changed'` event.
      *
      * @param state - The target pipeline state.

package/dist/voice-pipeline/VoicePipelineOrchestrator.js.map

@@ -1 +1 @@
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package/dist/voice-pipeline/WebRTCStreamTransport.d.ts

@@ -0,0 +1,421 @@
+/**
+ * @module voice-pipeline/WebRTCStreamTransport
+ *
+ * A concrete {@link IStreamTransport} implementation that wraps a WebRTC
+ * `RTCPeerConnection` and its DataChannels for bidirectional audio/text
+ * streaming. Provides lower latency than WebSocket by using unreliable/
+ * unordered DataChannels (UDP-like semantics) for audio frames and a
+ * reliable/ordered channel for control messages.
+ *
+ * ## Channel architecture
+ *
+ * Two DataChannels are created on the connection:
+ *
+ * | Channel   | Label       | Ordered | Reliable | Purpose                          |
+ * |-----------|-------------|---------|----------|----------------------------------|
+ * | Audio     | `'audio'`   | No      | No       | Low-latency PCM frame transport  |
+ * | Control   | `'control'` | Yes     | Yes      | JSON control messages (lossless) |
+ *
+ * The audio channel uses `maxRetransmits: 0` (fire-and-forget) because
+ * real-time voice tolerates occasional packet loss better than added
+ * latency from retransmissions. The control channel uses default TCP-like
+ * reliability since protocol messages (mute/unmute/stop) must arrive.
+ *
+ * ## Wire protocol
+ *
+ * ### Audio channel (binary)
+ * Each message is a raw `ArrayBuffer` containing Float32Array PCM samples.
+ * The transport reconstructs the Float32Array and wraps it in an
+ * {@link AudioFrame} with the configured sample rate.
+ *
+ * ### Control channel (text)
+ * Each message is a JSON string conforming to {@link ClientTextMessage}
+ * (inbound) or {@link ServerTextMessage} (outbound). Malformed JSON
+ * emits an `'error'` event without crashing the transport.
+ *
+ * ## Native dependency
+ * Node.js does not ship a built-in WebRTC stack. This transport requires
+ * the `wrtc` npm package (or a compatible polyfill) as an optional peer
+ * dependency. A helpful error is thrown at construction time if the
+ * package is not installed.
+ *
+ * @see {@link IStreamTransport} for the interface contract.
+ * @see {@link WebSocketStreamTransport} for the WebSocket-based sibling.
+ * @see {@link VoicePipelineOrchestrator} which consumes this transport.
+ */
+import { EventEmitter } from 'node:events';
+import type { IStreamTransport, AudioFrame, EncodedAudioChunk, TransportControlMessage, ServerTextMessage } from './types.js';
+/**
+ * Minimal subset of the W3C RTCDataChannel interface required by this
+ * transport. Defined locally to avoid a hard compile-time dependency on
+ * `wrtc` type definitions — the actual `wrtc` module is loaded at runtime
+ * via dynamic import.
+ *
+ * @see https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel
+ */
+export interface RTCDataChannelLike {
+    /** Human-readable label assigned when the channel was created. */
+    readonly label: string;
+    /**
+     * Channel readyState following the W3C spec:
+     * `'connecting'` | `'open'` | `'closing'` | `'closed'`.
+     */
+    readonly readyState: string;
+    /**
+     * Controls whether data is sent as `'arraybuffer'` or `'blob'`.
+     * Must be set to `'arraybuffer'` for binary audio frames.
+     */
+    binaryType: string;
+    /** Send a string or binary payload over the channel. */
+    send(data: string | ArrayBuffer | ArrayBufferView): void;
+    /** Close the DataChannel. */
+    close(): void;
+    onopen: ((event: Event) => void) | null;
+    onclose: ((event: Event) => void) | null;
+    onerror: ((event: Event) => void) | null;
+    onmessage: ((event: {
+        data: unknown;
+    }) => void) | null;
+}
+/**
+ * Minimal subset of the W3C RTCPeerConnection interface consumed by
+ * this transport. Only the DataChannel creation, ICE handling, and
+ * connection-state observation surface is needed.
+ *
+ * @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection
+ */
+export interface RTCPeerConnectionLike {
+    /**
+     * Aggregate connection state:
+     * `'new'` | `'connecting'` | `'connected'` | `'disconnected'` | `'failed'` | `'closed'`.
+     */
+    readonly connectionState: string;
+    /**
+     * Create a new DataChannel on this connection.
+     *
+     * @param label - Human-readable channel label.
+     * @param options - DataChannel configuration (ordered, maxRetransmits, etc.).
+     * @returns The newly created DataChannel.
+     */
+    createDataChannel(label: string, options?: Record<string, unknown>): RTCDataChannelLike;
+    /**
+     * Add an ICE candidate received from the remote peer via the signaling
+     * channel.
+     *
+     * @param candidate - The ICE candidate object or `null` to signal
+     *   end-of-candidates.
+     */
+    addIceCandidate(candidate: unknown): Promise<void>;
+    /** Close the peer connection and all associated DataChannels. */
+    close(): void;
+    onicecandidate: ((event: {
+        candidate: unknown;
+    }) => void) | null;
+    onconnectionstatechange: (() => void) | null;
+    ondatachannel: ((event: {
+        channel: RTCDataChannelLike;
+    }) => void) | null;
+}
+/**
+ * Constructor options for {@link WebRTCStreamTransport}.
+ *
+ * @example
+ * ```typescript
+ * const config: WebRTCStreamTransportConfig = {
+ *   sampleRate: 16000,
+ *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
+ * };
+ * ```
+ */
+export interface WebRTCStreamTransportConfig {
+    /**
+     * Sample rate (in Hz) used to populate `AudioFrame.sampleRate` on
+     * inbound audio messages. Must match the rate the remote peer is sending.
+     *
+     * Common values: 16000 (telephony/STT), 24000 (TTS output), 48000 (high-fidelity).
+     *
+     * @example 16000
+     */
+    sampleRate: number;
+    /**
+     * ICE server configuration for NAT traversal. Forwarded to the
+     * underlying RTCPeerConnection when one is created internally.
+     *
+     * If the consumer provides a pre-existing RTCPeerConnection (via
+     * the constructor), this field is ignored — ICE servers are already
+     * baked into the connection at creation time.
+     *
+     * @example [{ urls: 'stun:stun.l.google.com:19302' }]
+     */
+    iceServers?: Array<{
+        urls: string | string[];
+        username?: string;
+        credential?: string;
+    }>;
+}
+/**
+ * Bidirectional voice pipeline transport backed by WebRTC DataChannels.
+ *
+ * Uses two DataChannels for optimal voice streaming:
+ * - An **unreliable, unordered** `audio` channel for PCM frames (minimal
+ *   latency, tolerates packet loss).
+ * - A **reliable, ordered** `control` channel for JSON control messages
+ *   (mute/unmute/stop/config).
+ *
+ * ## Lifecycle
+ *
+ * 1. Construct with an existing {@link RTCPeerConnectionLike}.
+ * 2. Call `initialize` to create DataChannels and wire up handlers.
+ * 3. The transport is `'open'` once both DataChannels reach `'open'` state.
+ * 4. Call `close` to tear down channels and the peer connection.
+ *
+ * ## Events emitted
+ *
+ * | Event            | Payload                      | Description                           |
+ * |------------------|------------------------------|---------------------------------------|
+ * | `'audio_frame'`  | {@link AudioFrame}           | Inbound audio from the remote peer.   |
+ * | `'control'`      | {@link ClientTextMessage}    | Inbound JSON control message.         |
+ * | `'connected'`    | *(none)*                     | Both DataChannels are open.           |
+ * | `'disconnected'` | *(none)*                     | Connection has been closed.           |
+ * | `'error'`        | `Error`                      | Transport-level error.                |
+ * | `'ice_candidate'`| ICE candidate object         | Local ICE candidate for signaling.    |
+ *
+ * @fires audio_frame - `(frame: AudioFrame)` for every inbound audio message.
+ * @fires control - `(msg: ClientTextMessage)` for every inbound control message.
+ * @fires connected - Both DataChannels have reached 'open' state.
+ * @fires disconnected - Peer connection has been closed.
+ * @fires error - Transport-level error occurred.
+ * @fires ice_candidate - Local ICE candidate available for the signaling layer.
+ *
+ * @see {@link IStreamTransport} for the interface contract.
+ * @see {@link WebSocketStreamTransport} for the WebSocket-based sibling.
+ *
+ * @example
+ * ```typescript
+ * import { WebRTCStreamTransport } from '../voice-pipeline';
+ *
+ * // Receive RTCPeerConnection from your signaling server
+ * const transport = new WebRTCStreamTransport(peerConnection, { sampleRate: 16000 });
+ * await transport.initialize();
+ *
+ * transport.on('audio_frame', (frame) => sttSession.pushAudio(frame));
+ * transport.on('control', (msg) => handleClientMessage(msg));
+ * transport.on('ice_candidate', (candidate) => signalingChannel.send(candidate));
+ * ```
+ */
+export declare class WebRTCStreamTransport extends EventEmitter implements IStreamTransport {
+    /**
+     * Stable UUID assigned at construction time.
+     * Used as a correlation key in logs and metrics.
+     */
+    readonly id: string;
+    /**
+     * Current connection state. Updated internally by DataChannel event
+     * handlers. Read externally via the `state` getter.
+     */
+    private _state;
+    /** The underlying WebRTC peer connection. */
+    private readonly _pc;
+    /**
+     * Audio sample rate propagated into every decoded {@link AudioFrame}.
+     * Configured once at construction and never changed.
+     */
+    private readonly _sampleRate;
+    /** Transport configuration (ICE servers, etc.). */
+    private readonly _config;
+    /**
+     * Unreliable/unordered DataChannel for audio frame transport.
+     * Created during `initialize`. `null` until then.
+     */
+    private _audioChannel;
+    /**
+     * Reliable/ordered DataChannel for JSON control messages.
+     * Created during `initialize`. `null` until then.
+     */
+    private _controlChannel;
+    /**
+     * Create a new WebRTC transport wrapping an existing peer connection.
+     *
+     * The transport starts in `'connecting'` state and transitions to
+     * `'open'` once both DataChannels are established and open.
+     *
+     * @param peerConnection - An existing RTCPeerConnection (from the `wrtc`
+     *   package or a browser-compatible polyfill). The connection should be
+     *   in `'new'` or `'connecting'` state — the transport will create
+     *   DataChannels on it.
+     * @param config - Transport-level configuration (must include sampleRate).
+     *
+     * @throws {Error} If `peerConnection` is null/undefined.
+     *
+     * @example
+     * ```typescript
+     * const pc = new RTCPeerConnection({ iceServers: [...] });
+     * const transport = new WebRTCStreamTransport(pc, { sampleRate: 16000 });
+     * ```
+     */
+    constructor(peerConnection: RTCPeerConnectionLike, config: WebRTCStreamTransportConfig);
+    /**
+     * Current connection state of the WebRTC transport.
+     *
+     * The transport is `'open'` only when both DataChannels (audio + control)
+     * have reached their `'open'` readyState. Any earlier state maps to
+     * `'connecting'`, and any teardown maps to `'closing'` or `'closed'`.
+     */
+    get state(): 'connecting' | 'open' | 'closing' | 'closed';
+    /**
+     * Create DataChannels and wire up all event handlers.
+     *
+     * This must be called after construction and before any send operations.
+     * It creates the `audio` (unreliable) and `control` (reliable)
+     * DataChannels on the peer connection and sets up message handlers.
+     *
+     * The transport transitions to `'open'` once both channels report
+     * readyState `'open'`.
+     *
+     * @returns Resolves when DataChannels are created (not necessarily open yet).
+     *
+     * @example
+     * ```typescript
+     * const transport = new WebRTCStreamTransport(pc, { sampleRate: 16000 });
+     * await transport.initialize();
+     * // transport.state may still be 'connecting' until ICE completes
+     * ```
+     */
+    initialize(): Promise<void>;
+    /**
+     * Send a synthesised audio chunk to the remote peer for playback.
+     *
+     * If the payload is an {@link EncodedAudioChunk} (has an `audio` Buffer
+     * property), that buffer's underlying ArrayBuffer is sent. If it is an
+     * {@link AudioFrame} (has a `samples` Float32Array), the samples'
+     * underlying ArrayBuffer is sent.
+     *
+     * @param chunk - Encoded audio chunk or raw PCM frame to deliver.
+     * @returns Resolves once the data has been handed to the DataChannel.
+     *
+     * @throws {Error} If the audio channel is not open or not initialized.
+     *
+     * @example
+     * ```typescript
+     * await transport.sendAudio({
+     *   audio: Buffer.from(opusBytes),
+     *   format: 'opus',
+     *   sampleRate: 24000,
+     *   durationMs: 20,
+     *   text: 'Hello!',
+     * });
+     * ```
+     */
+    sendAudio(chunk: EncodedAudioChunk | AudioFrame): Promise<void>;
+    /**
+     * Send a JSON control message to the remote peer.
+     *
+     * The message is JSON-stringified before transmission on the reliable
+     * control channel. Both {@link TransportControlMessage} and
+     * {@link ServerTextMessage} are accepted since they share the same
+     * serialisation path.
+     *
+     * @param msg - Server-side protocol message.
+     * @returns Resolves once the message has been handed to the DataChannel.
+     *
+     * @throws {Error} If the control channel is not open or not initialized.
+     *
+     * @example
+     * ```typescript
+     * await transport.sendControl({ type: 'mute' });
+     * ```
+     */
+    sendControl(msg: TransportControlMessage | ServerTextMessage): Promise<void>;
+    /**
+     * Initiate a graceful close of the transport.
+     *
+     * Closes both DataChannels and the underlying peer connection.
+     * The `'disconnected'` event fires once the connection has fully closed.
+     *
+     * @param _code - Ignored for WebRTC (included for IStreamTransport compat).
+     * @param _reason - Ignored for WebRTC (included for IStreamTransport compat).
+     */
+    close(_code?: number, _reason?: string): void;
+    /**
+     * Add a remote ICE candidate received through the signaling channel.
+     *
+     * WebRTC requires ICE candidate exchange for NAT traversal. The signaling
+     * server should forward remote candidates to this method.
+     *
+     * @param candidate - The ICE candidate object from the remote peer, or
+     *   `null` to signal end-of-candidates.
+     * @returns Resolves when the candidate has been added to the connection.
+     *
+     * @throws {Error} If the candidate is invalid or the connection is closed.
+     *
+     * @example
+     * ```typescript
+     * signalingChannel.on('ice_candidate', async (candidate) => {
+     *   await transport.addIceCandidate(candidate);
+     * });
+     * ```
+     */
+    addIceCandidate(candidate: unknown): Promise<void>;
+    /**
+     * Attach message and lifecycle handlers to the audio DataChannel.
+     *
+     * Inbound binary messages are decoded as Float32Array PCM samples and
+     * emitted as `'audio_frame'` events. This channel uses arraybuffer
+     * binary type for efficient binary transfer.
+     *
+     * @param channel - The audio DataChannel to attach handlers to.
+     */
+    private _attachAudioChannelHandlers;
+    /**
+     * Attach message and lifecycle handlers to the control DataChannel.
+     *
+     * Inbound text messages are JSON-parsed as {@link ClientTextMessage}
+     * and emitted as `'control'` events. Malformed JSON triggers an
+     * `'error'` event without crashing the transport.
+     *
+     * @param channel - The control DataChannel to attach handlers to.
+     */
+    private _attachControlChannelHandlers;
+    /**
+     * Attach handlers to the peer connection for ICE candidate exchange
+     * and connection state monitoring.
+     *
+     * ICE candidates are re-emitted as `'ice_candidate'` events so the
+     * consumer can relay them to the remote peer via a signaling channel.
+     * Connection state changes are mapped to transport state transitions.
+     */
+    private _attachPeerConnectionHandlers;
+    /**
+     * Check if both DataChannels are in the `'open'` readyState.
+     *
+     * Called from each channel's `onopen` handler. Only transitions the
+     * transport to `'open'` state once BOTH channels are ready, since
+     * sending on a non-open channel would throw.
+     */
+    private _checkBothChannelsOpen;
+}
+/**
+ * Create a WebRTC transport with a fresh RTCPeerConnection from the `wrtc`
+ * package. This is a convenience factory for server-side usage where a
+ * pre-existing connection is not available.
+ *
+ * The `wrtc` package is loaded via dynamic import to keep it as an
+ * optional peer dependency. If the package is not installed, a descriptive
+ * error is thrown.
+ *
+ * @param config - Transport configuration including sample rate and ICE servers.
+ * @returns A new {@link WebRTCStreamTransport} wrapping a fresh peer connection.
+ *
+ * @throws {Error} If the `wrtc` package is not installed or importable.
+ *
+ * @example
+ * ```typescript
+ * const transport = await createWebRTCTransport({
+ *   sampleRate: 16000,
+ *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
+ * });
+ * await transport.initialize();
+ * ```
+ */
+export declare function createWebRTCTransport(config: WebRTCStreamTransportConfig): Promise<WebRTCStreamTransport>;
+//# sourceMappingURL=WebRTCStreamTransport.d.ts.map

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