@framers/agentos 0.13.78 → 0.13.79

package/dist/io/voice-pipeline/AcousticEndpointDetector.d.ts

@@ -105,6 +105,8 @@ export declare class AcousticEndpointDetector extends EventEmitter implements IE
      * See `IEndpointDetector.mode`.
      */
     readonly mode: "acoustic";
+    /** Maximum silence wait plus one polling interval, used when draining a listen session. */
+    readonly turnCompletionTimeoutMs: number;
     /**
      * Underlying silence-duration tracker from `media/audio/`.
      * Handles the actual timer management and threshold comparison.

package/dist/io/voice-pipeline/AcousticEndpointDetector.d.ts.map

@@ -1 +1 @@
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package/dist/io/voice-pipeline/AcousticEndpointDetector.js

@@ -108,6 +108,7 @@ export class AcousticEndpointDetector extends EventEmitter {
             utteranceEndThresholdMs: config.utteranceEndThresholdMs ?? 3000,
         };
         this.silenceDetector = new SilenceDetector(sdConfig);
+        this.turnCompletionTimeoutMs = Math.max(500, sdConfig.utteranceEndThresholdMs || 3000) + 250;
         // When SilenceDetector decides the utterance has ended (silence exceeded
         // utteranceEndThresholdMs), translate that into a TurnCompleteEvent.
         this.silenceDetector.on('utterance_end_detected', (_silenceDurationMs) => {

package/dist/io/voice-pipeline/AcousticEndpointDetector.js.map

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

@@ -142,6 +142,8 @@ export declare class HeuristicEndpointDetector extends EventEmitter implements I
      * @param options - Optional configuration overrides.
      */
     constructor(options?: HeuristicEndpointDetectorOptions);
+    /** Maximum wait for a pending turn after speech ends, used when draining a listen session. */
+    get turnCompletionTimeoutMs(): number;
     /**
      * Ingest a transcript event from the upstream STT session.
      *

package/dist/io/voice-pipeline/HeuristicEndpointDetector.d.ts.map

@@ -1 +1 @@
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package/dist/io/voice-pipeline/HeuristicEndpointDetector.js

@@ -187,6 +187,10 @@ export class HeuristicEndpointDetector extends EventEmitter {
         this.speechEndWithoutTextAt = 0;
         this.silenceTimeoutMs = options.silenceTimeoutMs ?? DEFAULT_SILENCE_TIMEOUT_MS;
     }
+    /** Maximum wait for a pending turn after speech ends, used when draining a listen session. */
+    get turnCompletionTimeoutMs() {
+        return this.silenceTimeoutMs;
+    }
     // ---------------------------------------------------------------------------
     // IEndpointDetector -- pushTranscript
     // ---------------------------------------------------------------------------

package/dist/io/voice-pipeline/HeuristicEndpointDetector.js.map

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

@@ -0,0 +1,111 @@
+/**
+ * @module voice-pipeline/ListenSession
+ *
+ * A session that listens and writes: a transport's audio into a streaming
+ * transcriber, each final (or each detector turn) into a sink as a numbered
+ * turn. No synthesis, no barge-in, no agent turn. Built for a notetaker, a
+ * meeting bot or a transcription job, where {@link VoicePipelineOrchestrator}
+ * is built for a conversation.
+ */
+import { EventEmitter } from 'node:events';
+import { type ReconnectingSTTOptions } from './ReconnectingSTT.js';
+import type { IEndpointDetector, IStreamingSTT, IStreamTransport, ITranscriptInputSink, StreamingSTTConfig, TranscriptEvent } from './types.js';
+export type { ITranscriptSink, ITranscriptInputSink, TranscriptInput, TranscriptTurn } from './types.js';
+/**
+ * Why a listen session ended. `stopped` drains the last utterance first; `purged` drains nothing and drops what is
+ * queued; faults are `stt_failed` (the transcriber failed for good, past {@link ReconnectingSTT}'s cap or at once
+ * without it), `transport_closed`, `transport_failed` (the transport emitted an error, included in `ended`)
+ * and `sink_failed` (the sink threw on a write).
+ */
+export type ListenEndReason = 'stopped' | 'purged' | 'stt_failed' | 'transport_closed' | 'transport_failed' | 'sink_failed';
+/** What {@link startListenSession} takes besides the transport and the sink: only the transcriber and the run are required. */
+export interface ListenSessionOverrides {
+    /** The streaming transcriber; wrapped in {@link ReconnectingSTT} unless `reconnect` is `false`. */
+    streamingSTT: IStreamingSTT;
+    /** The run the turns belong to; turn ids are `<runId>:<n>`. */
+    runId: string;
+    /**
+     * When given, its `turn_complete` events are the turns; without one, each final is a turn.
+     * An empty detector transcript uses the STT finals since the previous turn, joined with spaces,
+     * and their first start and last end times (including with {@link AcousticEndpointDetector}).
+     */
+    endpointDetector?: IEndpointDetector;
+    /** Passed to the transcriber's `startSession()`. */
+    sttConfig?: StreamingSTTConfig;
+    /** Wrap the transcriber in {@link ReconnectingSTT} (default true). A chain of several providers passes `false`. */
+    reconnect?: boolean | ReconnectingSTTOptions;
+    /** Finals waiting for the sink before the transport is paused. Default 8. */
+    sinkQueueLimit?: number;
+    /**
+     * Names who spoke a turn, from the transcriber's final. A detector turn carries its text and no words,
+     * unless empty detector text uses the accumulated STT finals and their words.
+     */
+    speaker?: (event: TranscriptEvent) => string | undefined;
+    /** How long a stop waits for the flushed final to arrive, before draining a pending detector turn. Default 250. */
+    drainMs?: number;
+}
+/** A listen session's counts so far. */
+export interface ListenSessionStats {
+    /** Turns the sink stored; one it dropped (behind a fence, or for a run that is over) is not counted. */
+    turns: number;
+    /**
+     * Audio frames that came while the transport was asked to pause. They still reach the transcriber, so a count
+     * above zero shows a transport that does not hold its buffer.
+     */
+    dropped: number;
+    /** Reconnects of the transcriber. */
+    reconnects: number;
+}
+/**
+ * `listening`: audio flows and finals are written. `draining`: a stop was asked; the transcriber is flushed and the
+ * finals it still emits are written. `ending`: nothing new is accepted; the queue is written out. `ended`: done.
+ */
+export type ListenSessionState = 'listening' | 'draining' | 'ending' | 'ended';
+/** A running listen session; {@link startListenSession} lists its events. */
+export interface ListenSession extends EventEmitter {
+    /** The run its turns belong to. */
+    readonly runId: string;
+    /** Where the session is in its life, read when asked. */
+    readonly state: ListenSessionState;
+    /** The transport it reads. */
+    readonly transport: IStreamTransport;
+    /** The counts so far. */
+    stats(): ListenSessionStats;
+    /**
+     * `stopped` flushes the transcriber, waits `drainMs`, then waits for a pending detector turn up to
+     * its `turnCompletionTimeoutMs` (3250 ms if omitted), plus one timer tick, before resetting. `purged` drops everything
+     * queued without draining. Every end releases an outstanding transport pause; resolves at `ended`.
+     */
+    close(reason?: ListenEndReason): Promise<void>;
+}
+/**
+ * Starts listening. Emits `turn` (a {@link TranscriptTurn}), `interim` (a {@link TranscriptEvent}), `stt:reconnected`,
+ * `stt:gap`, `paused`, `resumed` and `ended` (`{ reason, error? }`).
+ *
+ * The transport's frames go to the transcriber. Each final, or with an endpoint detector each `turn_complete`, goes
+ * to the sink, one write at a time in arrival order; the sink numbers it. When `sinkQueueLimit` finals wait for the
+ * sink, the session emits `pause` on the transport and `paused` on itself, and `resume` and `resumed` once the queue
+ * is down to half or the session ends; a final is never dropped for a slow sink. The transcriber's `speech_start`
+ * and `speech_end` reach the detector as voice activity, as the orchestrator gives them. The session ends once, for the first
+ * {@link ListenEndReason}; it closes the transcriber and leaves the transport open, since the transport is the
+ * caller's. A transport error ends it with `transport_failed` and the error in `ended`. Its transport listeners
+ * are removed on every end. An empty detector turn uses the accumulated STT finals and their timing.
+ *
+ * @param transport - Where the audio comes from: the session listens for its `audio`, `close` and `error`.
+ * @param sink - Where the turns go; it assigns each turn's number and id.
+ * @param overrides - The transcriber, the run and the session's options.
+ * @returns The session, listening.
+ * @throws {Error} When the transcriber's first session fails to open.
+ *
+ * @example
+ * ```typescript
+ * const session = await startListenSession(transport, sink, {
+ *   streamingSTT: new DeepgramStreamingSTT({ apiKey: process.env.DEEPGRAM_API_KEY! }),
+ *   runId: 'run-1',
+ * });
+ * session.on('ended', ({ reason }: { reason: ListenEndReason }) => console.log(`listening ended: ${reason}`));
+ * await session.close('stopped');
+ * ```
+ */
+export declare function startListenSession(transport: IStreamTransport, sink: ITranscriptInputSink, overrides: ListenSessionOverrides): Promise<ListenSession>;
+//# sourceMappingURL=ListenSession.d.ts.map

package/dist/io/voice-pipeline/ListenSession.d.ts.map

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package/dist/io/voice-pipeline/ListenSession.js

@@ -0,0 +1,242 @@
+/**
+ * @module voice-pipeline/ListenSession
+ *
+ * A session that listens and writes: a transport's audio into a streaming
+ * transcriber, each final (or each detector turn) into a sink as a numbered
+ * turn. No synthesis, no barge-in, no agent turn. Built for a notetaker, a
+ * meeting bot or a transcription job, where {@link VoicePipelineOrchestrator}
+ * is built for a conversation.
+ */
+import { EventEmitter } from 'node:events';
+import { ReconnectingSTT } from './ReconnectingSTT.js';
+/**
+ * Starts listening. Emits `turn` (a {@link TranscriptTurn}), `interim` (a {@link TranscriptEvent}), `stt:reconnected`,
+ * `stt:gap`, `paused`, `resumed` and `ended` (`{ reason, error? }`).
+ *
+ * The transport's frames go to the transcriber. Each final, or with an endpoint detector each `turn_complete`, goes
+ * to the sink, one write at a time in arrival order; the sink numbers it. When `sinkQueueLimit` finals wait for the
+ * sink, the session emits `pause` on the transport and `paused` on itself, and `resume` and `resumed` once the queue
+ * is down to half or the session ends; a final is never dropped for a slow sink. The transcriber's `speech_start`
+ * and `speech_end` reach the detector as voice activity, as the orchestrator gives them. The session ends once, for the first
+ * {@link ListenEndReason}; it closes the transcriber and leaves the transport open, since the transport is the
+ * caller's. A transport error ends it with `transport_failed` and the error in `ended`. Its transport listeners
+ * are removed on every end. An empty detector turn uses the accumulated STT finals and their timing.
+ *
+ * @param transport - Where the audio comes from: the session listens for its `audio`, `close` and `error`.
+ * @param sink - Where the turns go; it assigns each turn's number and id.
+ * @param overrides - The transcriber, the run and the session's options.
+ * @returns The session, listening.
+ * @throws {Error} When the transcriber's first session fails to open.
+ *
+ * @example
+ * ```typescript
+ * const session = await startListenSession(transport, sink, {
+ *   streamingSTT: new DeepgramStreamingSTT({ apiKey: process.env.DEEPGRAM_API_KEY! }),
+ *   runId: 'run-1',
+ * });
+ * session.on('ended', ({ reason }: { reason: ListenEndReason }) => console.log(`listening ended: ${reason}`));
+ * await session.close('stopped');
+ * ```
+ */
+export async function startListenSession(transport, sink, overrides) {
+    const stt = overrides.reconnect === false ? overrides.streamingSTT : new ReconnectingSTT(overrides.streamingSTT, typeof overrides.reconnect === 'object' ? overrides.reconnect : {});
+    const sttSession = await stt.startSession(overrides.sttConfig);
+    const detector = overrides.endpointDetector;
+    const queueLimit = overrides.sinkQueueLimit ?? 8;
+    const drainMs = overrides.drainMs ?? 250;
+    const emitter = new EventEmitter();
+    let state = 'listening';
+    let turns = 0;
+    let dropped = 0;
+    let reconnects = 0;
+    let paused = false;
+    let sinkFailed = false;
+    const pending = [];
+    const finals = [];
+    let speechEnded = false;
+    let writing = Promise.resolve();
+    let endPromise;
+    const accepting = () => state === 'listening' || state === 'draining';
+    const resume = () => {
+        if (!paused)
+            return;
+        paused = false;
+        transport.emit('resume');
+        emitter.emit('resumed');
+    };
+    const drainDetector = async () => {
+        if (!detector || finals.length === 0)
+            return;
+        await new Promise((resolve) => {
+            const done = () => {
+                clearTimeout(timer);
+                detector.off('turn_complete', done);
+                resolve();
+            };
+            // One timer tick lets a detector scheduled by the synthetic speech_end
+            // below emit at its bound before the fallback removes its listener.
+            const timer = setTimeout(done, (detector.turnCompletionTimeoutMs ?? 3250) + 1);
+            detector.on('turn_complete', done);
+            // A provider need not report speech_end when flushed. End the input so
+            // the detector can start its silence timer, without restarting a pending one.
+            if (!speechEnded)
+                detector.pushVadEvent({ type: 'speech_end', timestamp: Date.now(), source: 'stt' });
+        });
+    };
+    /** One end for every path. The writer calls it with `fromWriter` and never waits for its own chain. */
+    const end = (reason, error, fromWriter = false) => {
+        if (endPromise)
+            return endPromise;
+        // Reserve the end before cleanup emits resume: its listeners may call close again.
+        let resolveEnd = () => { };
+        let rejectEnd = () => { };
+        endPromise = new Promise((resolve, reject) => { resolveEnd = resolve; rejectEnd = reject; });
+        void (async () => {
+            if (reason === 'stopped') {
+                // the transcriber is flushed and its last finals are still written
+                state = 'draining';
+                try {
+                    await sttSession.flush();
+                }
+                catch {
+                    // the transcriber is gone; nothing to drain
+                }
+                await new Promise((r) => setTimeout(r, drainMs));
+                await drainDetector();
+            }
+            state = 'ending';
+            transport.off('audio', onAudio);
+            transport.off('close', onTransportClose);
+            detector?.off('turn_complete', onTurnComplete);
+            finals.length = 0;
+            if (reason === 'purged')
+                pending.length = 0;
+            resume();
+            sttSession.close();
+            detector?.reset();
+            if (!fromWriter)
+                await writing;
+            transport.off('error', onTransportError);
+            state = 'ended';
+            emitter.emit('ended', error ? { reason, error } : { reason });
+        })().then(resolveEnd, rejectEnd);
+        return endPromise;
+    };
+    const write = (text, event) => {
+        if (!accepting() || sinkFailed)
+            return;
+        const turn = { runId: overrides.runId, at: Date.now(), text, startMs: event.startMs, endMs: event.endMs, speaker: overrides.speaker?.(event) };
+        pending.push(turn);
+        if (pending.length >= queueLimit && !paused) {
+            paused = true;
+            transport.emit('pause');
+            emitter.emit('paused');
+        }
+        writing = writing.then(async () => {
+            const next = pending.shift();
+            if (!next || sinkFailed)
+                return;
+            try {
+                const stored = await sink.appendNext(next);
+                if (stored) {
+                    turns += 1;
+                    emitter.emit('turn', stored);
+                }
+            }
+            catch (err) {
+                sinkFailed = true;
+                pending.length = 0;
+                void end('sink_failed', err instanceof Error ? err : new Error(String(err)), true);
+                return;
+            }
+            if (paused && pending.length < Math.ceil(queueLimit / 2)) {
+                resume();
+            }
+        });
+    };
+    const onAudio = (frame) => {
+        if (state !== 'listening')
+            return;
+        if (paused)
+            dropped += 1; // the transport was asked to pause; a frame that still comes is counted, and the transcriber keeps its own buffer
+        sttSession.pushAudio(frame);
+    };
+    const onTransportClose = () => {
+        void end('transport_closed');
+    };
+    const onTransportError = (error) => {
+        void end('transport_failed', error);
+    };
+    const onTurnComplete = (turn) => {
+        if (!accepting())
+            return;
+        const accumulated = finals.splice(0);
+        if (!turn.transcript.trim()) {
+            const text = accumulated.map((event) => event.text).filter(Boolean).join(' ');
+            if (text)
+                write(text, {
+                    text, confidence: accumulated[accumulated.length - 1].confidence,
+                    words: accumulated.flatMap((event) => event.words), isFinal: true,
+                    startMs: accumulated[0].startMs, endMs: accumulated[accumulated.length - 1].endMs,
+                });
+            return;
+        }
+        // a detector turn has its text and confidence, and no audio boundaries of its own
+        write(turn.transcript, { text: turn.transcript, confidence: turn.confidence, words: [], isFinal: true, durationMs: turn.durationMs });
+    };
+    sttSession.on('transcript', (event) => {
+        if (!accepting())
+            return;
+        if (detector) {
+            if (event.isFinal)
+                finals.push(event);
+            detector.pushTranscript(event);
+            if (!event.isFinal)
+                emitter.emit('interim', event);
+            return;
+        }
+        if (event.isFinal)
+            write(event.text, event);
+        else
+            emitter.emit('interim', event);
+    });
+    const onVad = (event) => {
+        if (!accepting())
+            return;
+        if (event.type === 'speech_start')
+            speechEnded = false;
+        else if (event.type === 'speech_end')
+            speechEnded = true;
+        detector?.pushVadEvent(event);
+    };
+    sttSession.on('vad', onVad);
+    // the transcriber's own speech events reach the detector as voice activity, as the orchestrator gives them
+    sttSession.on('speech_start', () => onVad({ type: 'speech_start', timestamp: Date.now(), source: 'stt' }));
+    sttSession.on('speech_end', () => onVad({ type: 'speech_end', timestamp: Date.now(), source: 'stt' }));
+    sttSession.on('stt:reconnected', (e) => {
+        reconnects += 1;
+        emitter.emit('stt:reconnected', e);
+    });
+    sttSession.on('stt:gap', (e) => emitter.emit('stt:gap', e));
+    sttSession.on('error', (err) => {
+        if (state === 'listening')
+            void end('stt_failed', err);
+    });
+    sttSession.on('close', () => {
+        if (state === 'listening')
+            void end('stt_failed', new Error('the transcriber closed'));
+    });
+    detector?.on('turn_complete', onTurnComplete);
+    transport.on('audio', onAudio);
+    transport.on('close', onTransportClose);
+    transport.on('error', onTransportError);
+    const session = Object.assign(emitter, {
+        runId: overrides.runId,
+        transport,
+        stats: () => ({ turns, dropped, reconnects }),
+        close: (reason = 'stopped') => end(reason),
+    });
+    Object.defineProperty(session, 'state', { get: () => state, enumerable: true });
+    return session;
+}
+//# sourceMappingURL=ListenSession.js.map

package/dist/io/voice-pipeline/ListenSession.js.map

@@ -0,0 +1 @@
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package/dist/io/voice-pipeline/ReconnectingSTT.d.ts

@@ -0,0 +1,90 @@
+import type { IStreamingSTT, StreamingSTTConfig, StreamingSTTSession } from './types.js';
+/** Options of {@link ReconnectingSTT}. */
+export interface ReconnectingSTTOptions {
+    /** How much audio is kept for replay. Default 3000. */
+    ringBufferCapacityMs?: number;
+    /** Replays below this are skipped. Default 400. */
+    minReplayMs?: number;
+    /** Attempts allowed in any sixty seconds before the session gives up. Default 3. */
+    maxAttemptsPerMinute?: number;
+    /** Frames queued while a reconnect runs; beyond it the oldest are dropped and counted. Default 2000 ms. */
+    queueCapacityMs?: number;
+    /** Milliseconds now, for the attempt window and the gap's length. Default `Date.now`. */
+    clock?: () => number;
+}
+/** Payload of a session's `stt:reconnected`: a new session of the provider took over. */
+export interface SttReconnectedEvent {
+    /** The attempt's number among the attempts of the last sixty seconds, from 1. */
+    attempt: number;
+    /** Milliseconds of buffered audio replayed into the new session; 0 when there was less than `minReplayMs`. */
+    replayedMs: number;
+}
+/** Payload of a session's `stt:gap`, emitted right after `stt:reconnected`. */
+export interface SttGapEvent {
+    /** The attempt's number, as in {@link SttReconnectedEvent}. */
+    attempt: number;
+    /**
+     * Milliseconds the reconnect took, by the options' clock. The audio pushed in that time was queued for the new
+     * session, less the oldest frames a full queue dropped (the session's `droppedFrames`).
+     */
+    lostMs: number;
+}
+/**
+ * An {@link IStreamingSTT} over one provider whose session is reconnected.
+ *
+ * An `error` or `close` of the provider's session that the caller did not ask
+ * for opens a new session of the same provider, at most
+ * `maxAttemptsPerMinute` times in any sixty seconds. The new session hears the
+ * buffered audio first (the last `ringBufferCapacityMs`, when there is at least
+ * `minReplayMs` of it), then the frames pushed while it opened, then live
+ * audio. Its `startMs` and `endMs` are moved onto one timeline, the audio
+ * pushed since the session began, and a final that repeats one already
+ * emitted over the same audio is dropped. Only finals with finite start and end
+ * bounds and a positive duration enter the dedupe history; others pass through.
+ * Queue overflow leaves a gap after the replay, preserving the original times
+ * of both replayed and retained audio. Each reconnect emits
+ * `stt:reconnected` ({@link SttReconnectedEvent}) and `stt:gap`
+ * ({@link SttGapEvent}). Past the cap the session emits `error`, then `close`.
+ *
+ * The session forwards `transcript`, `vad`, `speech_start`, `speech_end`,
+ * `usage` and `warning` from the provider's sessions; `usage` comes from every
+ * one of them, a replaced one too, since a session reports its audio as it
+ * closes. `flush()` during a reconnect waits for the replacement, replay and
+ * queued audio before flushing it. A close during that flush, or after it with
+ * no further provider activity or pushed audio, ends the stream without a
+ * reconnect. Activity after a completed flush makes a later close a failure.
+ *
+ * @example
+ * ```typescript
+ * const stt = new ReconnectingSTT(new DeepgramStreamingSTT({ apiKey: process.env.DEEPGRAM_API_KEY! }));
+ * const session = await stt.startSession({ language: 'en-US' });
+ * session.on('stt:gap', ({ lostMs }: SttGapEvent) => console.log(`the transcriber was away for ${lostMs} ms`));
+ * ```
+ */
+export declare class ReconnectingSTT implements IStreamingSTT {
+    private readonly inner;
+    /** `reconnecting:` followed by the inner provider's id. */
+    readonly providerId: string;
+    private readonly capacityMs;
+    private readonly minReplayMs;
+    private readonly maxAttempts;
+    private readonly queueCapacityMs;
+    private readonly clock;
+    /**
+     * @param inner - The provider each session opens, again on every reconnect.
+     * @param options - The replay buffer, the attempt cap, the queue and the clock.
+     */
+    constructor(inner: IStreamingSTT, options?: ReconnectingSTTOptions);
+    /** Whether the inner provider has a session open. */
+    get isStreaming(): boolean;
+    /**
+     * Opens a session of the inner provider behind a session that reconnects it.
+     *
+     * @param config - Passed to the inner provider for the first session and for every reconnect.
+     * @returns The session, with a read-only `droppedFrames`: the queued frames dropped during reconnects so far.
+     * @throws {Error} When the first session fails to open; a session that fails to open on a reconnect is retried
+     *   within the cap.
+     */
+    startSession(config?: StreamingSTTConfig): Promise<StreamingSTTSession>;
+}
+//# sourceMappingURL=ReconnectingSTT.d.ts.map

package/dist/io/voice-pipeline/ReconnectingSTT.d.ts.map

@@ -0,0 +1 @@
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