@elizaos/plugin-local-inference 2.0.0-beta.1 → 2.0.3-beta.3

package/dist/services/voice/profile-store.d.ts

@@ -0,0 +1,248 @@
+/**
+ * Content-addressed voice-profile store with hot LRU + cold disk tiers.
+ *
+ * Each profile is one WeSpeaker ResNet34-LM centroid plus running
+ * variance (Welford), consent flags, and an entity binding. Profiles
+ * are content-addressed by `sha256(centroid_bytes)` so duplicate
+ * captures collapse and entity merges are safe.
+ *
+ * Layout under `$ELIZA_STATE_DIR/voice-profiles/`:
+ *
+ *   index.json                   — entityId/cluster index + LRU order
+ *   profiles/vp_<sha>.json       — one record per profile
+ *   audio/vp_<sha>/sample-*.wav  — optional, consent-gated
+ *
+ * The contract:
+ *  - **Hot LRU 30** in-memory records (default `hotCacheSize`).
+ *  - **Cold disk cap 200** (default `coldDiskMax`).
+ *  - `beginMatch()` starts at speech-start and resolves once minSpeechMs
+ *    of audio has been encoded — runs in parallel with ASR.
+ *  - `refine()` uses online running-mean + Welford variance.
+ *  - Profiles with a non-null `entityId` are never auto-evicted.
+ */
+import { type VoiceImprintMatch } from "./speaker-imprint";
+import type { VoiceInputSource } from "./types";
+/** Canonical schema version for `vp_*.json` records. */
+export declare const VOICE_PROFILE_RECORD_SCHEMA_VERSION: "eliza.voice_profile_record.v1";
+export interface VoiceProfileConsentState {
+    attributionAuthorized: boolean;
+    synthesisAuthorized: boolean;
+    grantedAt?: string;
+    grantedBy?: string;
+}
+export interface VoiceProfileAudioRef {
+    sampleId: string;
+    wavSha256: string;
+    durationMs: number;
+    recordedAt: string;
+    referenceText?: string;
+}
+export interface VoiceProfileRecord {
+    schemaVersion: typeof VOICE_PROFILE_RECORD_SCHEMA_VERSION;
+    profileId: string;
+    embeddingModel: string;
+    embeddingDim: number;
+    /** L2-normalized centroid; cosine == dot. */
+    centroid: number[];
+    /** Welford per-dim variance accumulator (M2 / max(1, n-1)). */
+    variance: number[];
+    /** Welford `M2` running sum (per-dim squared diff from running mean). */
+    welfordM2: number[];
+    sampleCount: number;
+    totalDurationMs: number;
+    firstObservedAt: string;
+    lastObservedAt: string;
+    lastRefinedAt: string;
+    entityId: string | null;
+    imprintClusterId: string;
+    confidence: number;
+    consent: VoiceProfileConsentState;
+    audioRefs?: VoiceProfileAudioRef[];
+    metadata?: Record<string, unknown>;
+}
+export interface VoiceProfileStoreOptions {
+    rootDir: string;
+    hotCacheSize?: number;
+    coldDiskMax?: number;
+    matchThreshold?: number;
+    /** Below this we open a new cluster instead of attributing. */
+    unmatchedClusterThreshold?: number;
+}
+export interface VoiceImprintMatchHandle {
+    /** Resolves once minSpeechMs of audio is encoded, or `null` if no match. */
+    result: Promise<VoiceImprintMatch | null>;
+    /** Synchronous polling for the latest match — null until first resolve. */
+    current(): VoiceImprintMatch | null;
+    cancel(): void;
+}
+/**
+ * Welford online variance update. Returns the new (mean, M2, count)
+ * triple so a caller can persist the M2 and derive variance lazily.
+ * `prevMean` is the running mean *before* the observation; the caller
+ * passes the L2-normalized embedding as the observation.
+ */
+export declare function welfordUpdate(args: {
+    count: number;
+    mean: readonly number[];
+    m2: readonly number[];
+    observation: ReadonlyArray<number>;
+}): {
+    mean: number[];
+    m2: number[];
+    count: number;
+};
+export declare function welfordVariance(m2: readonly number[], count: number): number[];
+/**
+ * Reject an observation if its per-dim distance from the centroid is
+ * more than `sigmaThreshold` standard deviations on more than half the
+ * dimensions. Used to drop cough / cross-talk samples that would
+ * corrupt the centroid.
+ */
+export declare function isOutlier(args: {
+    centroid: readonly number[];
+    variance: readonly number[];
+    observation: readonly number[];
+    sigmaThreshold?: number;
+}): boolean;
+export declare class VoiceProfileStore {
+    private readonly hotCacheSize;
+    private readonly coldDiskMax;
+    private readonly matchThreshold;
+    private readonly unmatchedThreshold;
+    private readonly rootDir;
+    private readonly profilesDir;
+    private readonly indexPath;
+    /** Hot cache: profileId → record. Insertion order = LRU order. */
+    private hot;
+    private indexCache;
+    constructor(options: VoiceProfileStoreOptions);
+    get matchThresholdValue(): number;
+    get unmatchedClusterThresholdValue(): number;
+    /** Public for tests / management UI. */
+    get hotCacheSizeValue(): number;
+    get coldDiskMaxValue(): number;
+    init(): Promise<void>;
+    private readIndex;
+    private writeIndex;
+    private profilePath;
+    private readProfileFromDisk;
+    private writeProfileToDisk;
+    /** Touch the LRU order: re-insert at the end. */
+    private touchHot;
+    private upsertIndexEntry;
+    private enforceColdLimit;
+    private ensureLoaded;
+    /** Walk profiles + return the best match above `matchThreshold` (or null). */
+    findBestMatch(args: {
+        embedding: Float32Array;
+        embeddingModel: string;
+    }): Promise<VoiceImprintMatch | null>;
+    private recordToImprintProfile;
+    /**
+     * Speculative match handle. The caller supplies a function that
+     * resolves to a single embedding once `minSpeechMs` of audio is
+     * available. The handle starts the lookup the moment it's
+     * constructed — there is no awaitable for "the encoder finished
+     * before we wanted it to".
+     */
+    beginMatch(args: {
+        embed: () => Promise<{
+            embedding: Float32Array;
+            embeddingModel: string;
+        } | null>;
+        signal?: AbortSignal;
+    }): VoiceImprintMatchHandle;
+    /** Create a new profile from a single capture. */
+    createProfile(args: {
+        centroid: Float32Array;
+        embeddingModel: string;
+        entityId?: string | null;
+        imprintClusterId?: string;
+        confidence: number;
+        durationMs: number;
+        consent?: Partial<VoiceProfileConsentState>;
+        audioRef?: VoiceProfileAudioRef;
+        metadata?: Record<string, unknown>;
+    }): Promise<VoiceProfileRecord>;
+    /**
+     * Fold one new embedding into the existing profile via the online
+     * running mean (sampleCount-weighted) and update Welford variance.
+     * Rejects outliers if `dropOutliers` is true (default).
+     */
+    refine(args: {
+        profileId: string;
+        embedding: Float32Array;
+        durationMs: number;
+        confidence: number;
+        audioRef?: VoiceProfileAudioRef;
+        dropOutliers?: boolean;
+    }): Promise<VoiceProfileRecord | null>;
+    bindEntity(args: {
+        profileId: string;
+        entityId: string;
+        label?: string;
+    }): Promise<VoiceProfileRecord | null>;
+    /**
+     * Merge a metadata patch onto a profile. Keys mapped to `null` are
+     * deleted; other keys overwrite. Used by the management routes for
+     * rename / relationship / retention edits.
+     */
+    updateMetadata(profileId: string, patch: Record<string, unknown>): Promise<VoiceProfileRecord | null>;
+    unbindEntity(profileId: string): Promise<VoiceProfileRecord | null>;
+    get(profileId: string): Promise<VoiceProfileRecord | null>;
+    list(): Promise<VoiceProfileRecord[]>;
+    /** For tests / management — drops a profile. Refuses if entityId is set. */
+    deleteProfile(args: {
+        profileId: string;
+        allowBoundEntity?: boolean;
+    }): Promise<boolean>;
+    /**
+     * Merge `sourceId` into `targetId`: a sample-count-weighted centroid
+     * combine (with the Chan parallel-variance update for Welford M2), union
+     * of audio refs, summed counts/durations, and confidence average. The
+     * target's metadata + entity binding win; an unbound target inherits the
+     * source's `entityId`. The source profile is deleted. Returns the merged
+     * target, or `null` if either profile is missing.
+     *
+     * Refuses when both carry a *different* `entityId` unless
+     * `allowEntityOverwrite` is set — merging two bound identities is a
+     * destructive operation the caller must opt into.
+     */
+    mergeProfiles(args: {
+        sourceId: string;
+        targetId: string;
+        allowEntityOverwrite?: boolean;
+    }): Promise<VoiceProfileRecord | null>;
+    /**
+     * Split the audio samples named by `sampleIds` out of `profileId` into a
+     * new profile. Returns the updated original plus the new split profile,
+     * or `null` if the profile is missing.
+     *
+     * Limitation: per-utterance embeddings are not retained (only the running
+     * centroid + Welford accumulators), so the split cannot re-cluster — the
+     * new profile copies the parent centroid and the split is by *audio sample
+     * assignment* only. Both profiles should be re-refined from fresh captures
+     * to diverge. The new profile is unbound (`entityId: null`) and gets a
+     * fresh imprint cluster.
+     */
+    splitProfile(args: {
+        profileId: string;
+        sampleIds: string[];
+    }): Promise<{
+        original: VoiceProfileRecord;
+        split: VoiceProfileRecord;
+    } | null>;
+}
+/** Snapshot of one observation for downstream attribution code. */
+export interface VoiceProfileObservation {
+    profileId: string;
+    imprintClusterId: string;
+    entityId: string | null;
+    embedding: Float32Array;
+    embeddingModel: string;
+    confidence: number;
+    source?: VoiceInputSource;
+    startMs?: number;
+    endMs?: number;
+}
+//# sourceMappingURL=profile-store.d.ts.map

package/dist/services/voice/profile-store.d.ts.map

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package/dist/services/voice/ring-buffer.d.ts

@@ -0,0 +1,40 @@
+import type { AudioSink } from "./types";
+export interface PcmRingBufferOptions {
+    /**
+     * Fired when the buffer is full and a write overwrites unread samples.
+     * Reports the count of dropped samples in this write call. Schedulers
+     * should use this signal to apply backpressure upstream — silent
+     * overwrites produce audible glitches.
+     */
+    onOverflow?: (droppedSamples: number) => void;
+}
+export declare class PcmRingBuffer {
+    private readonly capacity;
+    private readonly sampleRate;
+    private readonly sink;
+    private readonly buf;
+    private readPos;
+    private writePos;
+    private filled;
+    private readonly onOverflow?;
+    constructor(capacity: number, sampleRate: number, sink: AudioSink, options?: PcmRingBufferOptions);
+    write(pcm: Float32Array): void;
+    /** Fill ratio in [0, 1]. Schedulers can throttle TTS dispatches as this approaches 1. */
+    pressure(): number;
+    flushToSink(): number;
+    drain(): void;
+    size(): number;
+    capacityHint(): number;
+}
+export declare class InMemoryAudioSink implements AudioSink {
+    readonly chunks: Array<{
+        pcm: Float32Array;
+        sampleRate: number;
+    }>;
+    private buffered;
+    write(pcm: Float32Array, sampleRate: number): void;
+    drain(): void;
+    bufferedSamples(): number;
+    totalWritten(): number;
+}
+//# sourceMappingURL=ring-buffer.d.ts.map

package/dist/services/voice/ring-buffer.d.ts.map

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package/dist/services/voice/rollback-queue.d.ts

@@ -0,0 +1,24 @@
+import type { Phrase, RejectedTokenRange } from "./types";
+type PhraseState = "queued" | "synthesizing" | "ringbuffered" | "played";
+export interface RollbackEvent {
+    phraseId: number;
+    reason: "rejected-tokens";
+    rejectedRange: RejectedTokenRange;
+}
+export declare class RollbackQueue {
+    private readonly tracked;
+    track(phrase: Phrase): void;
+    markSynthesizing(phraseId: number): void;
+    markRingBuffered(phraseId: number): void;
+    markPlayed(phraseId: number): void;
+    drop(phraseId: number): void;
+    onRejected(range: RejectedTokenRange): RollbackEvent[];
+    snapshot(): ReadonlyArray<{
+        phrase: Phrase;
+        state: PhraseState;
+    }>;
+    private requireEntry;
+    private overlaps;
+}
+export {};
+//# sourceMappingURL=rollback-queue.d.ts.map

package/dist/services/voice/rollback-queue.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/samantha-preset-placeholder.d.ts

@@ -0,0 +1,67 @@
+/**
+ * Samantha-preset placeholder detection + on-the-fly regeneration.
+ *
+ * Background: the canonical default voice (Samantha, `af_same`) ships with
+ * the Eliza-1 bundle as `cache/voice-preset-default.bin` (ELZ1 format).
+ * The first I-wave shipped a 1052-byte zero-filled placeholder before the
+ * real preset bytes were produced — the runtime treats this placeholder as
+ * "not yet generated" and synthesises a fresh preset on first boot via the
+ * fused OmniVoice TTS (Path A).
+ *
+ * The detection rule is intentionally narrow: only an ELZ1 v1/v2 file whose
+ * (a) speaker embedding is exactly zero AND (b) ref_audio_tokens / ref_text
+ * are empty AND (c) phrase-cache seed is empty counts as a placeholder. Any
+ * file that has even one non-zero region is considered real and is left
+ * alone — the runtime never silently overwrites operator-supplied presets.
+ *
+ * The fallback chain executed by `EngineVoiceBridge.start()`:
+ *
+ *   1. If `cache/voice-preset-default.bin` is missing → throw (existing
+ *      behaviour, the bundle is malformed).
+ *   2. If the file exists and is NOT a placeholder → load it as-is.
+ *   3. If it IS a placeholder AND the OmniVoice reference-encode FFI is
+ *      available → regenerate the preset using the bundled Samantha
+ *      reference clip + transcript, write the new bytes back to disk,
+ *      then load.
+ *   4. If it IS a placeholder AND OmniVoice reference-encode is unavailable
+ *      → log a loud warning and fall through to the bundled Kokoro default
+ *      voice (`kokoro.defaultVoiceId`) by re-pointing the discovery layer.
+ */
+/** The exact byte-length the I-wave placeholder shipped at. Used as a fast
+ *  pre-check before the more expensive structural parse — files of any
+ *  other size cannot be the placeholder. */
+export declare const SAMANTHA_PLACEHOLDER_BYTE_LENGTH = 1052;
+/** Reasons `detectSamanthaPlaceholder` returns. Each is structurally
+ *  distinct so the caller can branch without re-parsing. */
+export type SamanthaPlaceholderState = {
+    kind: "missing";
+} | {
+    kind: "real-preset";
+    reason: string;
+} | {
+    kind: "placeholder";
+} | {
+    kind: "unreadable";
+    reason: string;
+};
+/**
+ * Inspect `presetPath` and report whether it is the I-wave zero-fill
+ * placeholder, a real preset, missing, or unreadable. Pure I/O — no
+ * mutation of the file.
+ */
+export declare function detectSamanthaPlaceholder(presetPath: string): SamanthaPlaceholderState;
+/**
+ * The text the on-the-fly regeneration uses as the Samantha reference
+ * transcript. Stable across versions so the produced preset is
+ * deterministic — pinning the prompt is part of the cacheability contract
+ * (same inputs -> same output bytes for byte-for-byte comparison).
+ */
+export declare const SAMANTHA_REFERENCE_TRANSCRIPT = "Hi, I'm Samantha. It's nice to finally talk to you about something real.";
+/**
+ * The closed-vocabulary VoiceDesign instruct string baked into the
+ * regenerated Samantha preset. Mirrors the publish-time freeze CLI's
+ * settings so the runtime regen and a future operator-built preset land on
+ * the same surface. Pinned for deterministic regeneration.
+ */
+export declare const SAMANTHA_INSTRUCT = "female, american accent, young adult, moderate pitch";
+//# sourceMappingURL=samantha-preset-placeholder.d.ts.map

package/dist/services/voice/samantha-preset-placeholder.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/samantha-preset-regenerator.d.ts

@@ -0,0 +1,87 @@
+/**
+ * Samantha-preset on-the-fly regeneration via the fused OmniVoice FFI.
+ *
+ * Path A from W3-11: when the bundle ships the I-wave zero-fill placeholder
+ * for `cache/voice-preset-default.bin`, the runtime synthesises a real
+ * preset by encoding the bundled Samantha reference clip through the FFI's
+ * `eliza_inference_encode_reference` entrypoint and writing the resulting
+ * `ref_audio_tokens` + canonical instruct/refText into a v2 preset blob.
+ *
+ * Determinism contract:
+ *   - The reference clip bytes (24 kHz mono fp32 WAV) are pinned in the
+ *     bundle.
+ *   - The reference transcript (`SAMANTHA_REFERENCE_TRANSCRIPT`) is pinned.
+ *   - The instruct string (`SAMANTHA_INSTRUCT`) is pinned.
+ *   - The OmniVoice encode entrypoint does not consume randomness (the
+ *     HuBERT semantic + RVQ codec passes are pure functions of the input
+ *     PCM + the model weights).
+ * Therefore the produced preset bytes are reproducible byte-for-byte across
+ * boots given the same FFI library + bundle.
+ */
+/** Outcome of `ensureSamanthaPresetReady`. Distinct kinds let the caller
+ *  log the right thing at the right level without re-doing detection. */
+export type EnsureSamanthaPresetOutcome = {
+    kind: "real-preset";
+} | {
+    kind: "missing-bundle-preset";
+} | {
+    kind: "regenerated";
+    bytes: number;
+    K: number;
+    refT: number;
+} | {
+    kind: "placeholder-no-regen";
+    reason: "missing-reference-wav" | "missing-ffi-library" | "ffi-no-encode-reference" | "encode-reference-failed";
+    detail: string;
+};
+export interface RegenerateOptions {
+    bundleRoot: string;
+    /** Absolute path the regenerated preset bytes should target. The caller
+     *  performs the write — this function only produces the bytes + metadata. */
+    presetPath: string;
+    /** Override path to the Samantha reference WAV. Defaults to the bundle's
+     *  `tts/omnivoice/samantha-ref.wav`. */
+    referenceWav?: string;
+    /** Override the canonical reference transcript. Defaults to the pinned
+     *  `SAMANTHA_REFERENCE_TRANSCRIPT`. */
+    referenceText?: string;
+}
+export interface RegenerateResult {
+    bytes: Uint8Array;
+    K: number;
+    refT: number;
+    embeddingDim: number;
+}
+/**
+ * Decode a 24 kHz mono Float32 LE WAV file into a Float32Array of PCM
+ * samples. Refuses anything that is not the canonical OmniVoice reference
+ * format — encoders happily accept stereo / 16-bit / 48 kHz inputs and
+ * silently degrade, which is exactly the kind of fallback sludge AGENTS.md
+ * §3 forbids. We require the file be in the right format up front.
+ */
+export declare function decodeMonoFloat32Wav24kHz(bytes: Uint8Array): Float32Array;
+/**
+ * Run the on-the-fly regeneration. Loads the bundle's OmniVoice FFI, calls
+ * `encodeReference` against the Samantha reference clip, and serialises the
+ * result into an ELZ1 v2 preset blob. The caller writes the bytes to disk.
+ */
+export declare function regenerateSamanthaPresetFromBundle(opts: RegenerateOptions): Promise<RegenerateResult>;
+/**
+ * Pre-flight: detect a placeholder preset at the bundle's canonical path
+ * and regenerate it via OmniVoice when possible. Called by the engine's
+ * `ensureActiveBundleVoiceReady()` before the synchronous preset load.
+ *
+ * Outcomes:
+ *
+ *   - `real-preset`            — nothing to do; the file is a real preset.
+ *   - `missing-bundle-preset`  — file does not exist; the engine's existing
+ *                                error path runs (loud failure).
+ *   - `regenerated`            — preset bytes were generated and written.
+ *   - `placeholder-no-regen`   — placeholder detected but regen could not
+ *                                run (FFI missing, reference clip missing,
+ *                                etc.). Returned for the caller to log; the
+ *                                engine then falls through to the bundled
+ *                                Kokoro default voice.
+ */
+export declare function ensureSamanthaPresetReady(bundleRoot: string): Promise<EnsureSamanthaPresetOutcome>;
+//# sourceMappingURL=samantha-preset-regenerator.d.ts.map

package/dist/services/voice/samantha-preset-regenerator.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/scheduler.d.ts

@@ -0,0 +1,146 @@
+import { BargeInController } from "./barge-in";
+import type { PhonemeTokenizer } from "./phoneme-tokenizer";
+import { PhraseCache } from "./phrase-cache";
+import { PhraseChunker } from "./phrase-chunker";
+import { PrefixPreservingQueue } from "./prefix-preserving-queue";
+import { PcmRingBuffer } from "./ring-buffer";
+import { RollbackQueue } from "./rollback-queue";
+import type { AudioChunk, AudioSink, OmniVoiceBackend, Phrase, RejectedTokenRange, SchedulerConfig, SpeakerPreset, TextToken, VoiceSchedulerTelemetryListener } from "./types";
+/**
+ * T2 — per-phrase TTS chunk-size telemetry, emitted once per
+ * `synthesizePhraseStream` call when `SchedulerEvents.onChunkMetrics` is
+ * wired. `chunks` is the in-arrival-order distribution of streamed PCM
+ * chunks (size in PCM bytes assuming Float32 samples, duration in ms
+ * derived from samples / sampleRate). Used to debug T1-class chunk-size
+ * pathologies and to verify T3 time-budget effects.
+ */
+export interface TtsPhraseChunkMetrics {
+    phraseId: number;
+    /** Order-preserving list of per-chunk sizes. Empty when no chunks landed. */
+    chunks: ReadonlyArray<{
+        chunkBytes: number;
+        chunkDurationMs: number;
+    }>;
+    /** Sum of chunk durations in ms. */
+    totalDurationMs: number;
+    /** Sum of chunk bytes. */
+    totalBytes: number;
+    /** Whether the phrase synthesis was cancelled mid-stream. */
+    cancelled: boolean;
+}
+export type TtsChunkMetricsListener = (metrics: TtsPhraseChunkMetrics) => void;
+export interface SchedulerEvents {
+    onPhrase?(phrase: Phrase): void;
+    onRollback?(phraseId: number, range: RejectedTokenRange): void;
+    onAudio?(chunk: AudioChunk): void;
+    /**
+     * Barge-in hard-stop: ring buffer drained, chunker reset, in-flight TTS
+     * cancelled. The engine layer's `voiceStreamingArgs` separately threads
+     * the `BargeInCancelToken.signal` (`bargeIn.onSignal` → `hard-stop`)
+     * into `dispatcher.generate` so the LLM/drafter abort too.
+     */
+    onCancel?(): void;
+    /** Provisional barge-in: a VAD voice hit while the agent is speaking paused TTS playback. */
+    onTtsPause?(): void;
+    /** Blip resolved the provisional barge-in — TTS playback resumed. */
+    onTtsResume?(): void;
+    /** Structured scheduler telemetry for latency, cache, rollback, and barge-in metrics. */
+    onTelemetry?: VoiceSchedulerTelemetryListener;
+    /**
+     * T2 — per-phrase TTS chunk-size distribution. Optional; when set, the
+     * scheduler emits one summary per streamed phrase synthesis (success or
+     * cancelled). Lets test harnesses and metrics consumers verify T1/T3
+     * effects without scraping the audio bus.
+     */
+    onChunkMetrics?: TtsChunkMetricsListener;
+}
+export interface SchedulerDeps {
+    backend: OmniVoiceBackend;
+    sink?: AudioSink;
+    phraseCache?: PhraseCache;
+    /** Optional. Required only when `config.chunkerConfig.chunkOn ===
+     *  'phoneme-stream'`. Defaults are available from
+     *  `createDefaultPhonemeTokenizer()`. */
+    phonemeTokenizer?: PhonemeTokenizer;
+}
+export declare class VoiceScheduler {
+    readonly chunker: PhraseChunker;
+    readonly rollback: RollbackQueue;
+    readonly bargeIn: BargeInController;
+    readonly ringBuffer: PcmRingBuffer;
+    readonly sink: AudioSink;
+    readonly preset: SpeakerPreset;
+    /**
+     * Prefix-preserving barge-in queue. When the streaming TTS path is active,
+     * each audio chunk is enqueued here tagged with its token range. On
+     * hard-stop (barge-in), `rollbackAt(divergencePoint)` partitions the
+     * queue: chunks at or before the divergence point are replayed into the
+     * sink; chunks after are dropped. This lets audio that was already
+     * correct play through without re-synthesizing.
+     */
+    readonly prefixQueue: PrefixPreservingQueue;
+    private readonly backend;
+    private readonly phraseCache;
+    private readonly events;
+    private readonly sampleRate;
+    private readonly inFlight;
+    private readonly maxInFlight;
+    private readonly streamingTtsActive;
+    private kernelTicks;
+    private nextStandalonePhraseId;
+    /** True while a provisional barge-in (`pause-tts`) has paused playback. */
+    private paused;
+    /**
+     * The last committed token index — updated whenever a phrase is dispatched
+     * to TTS. Used as the divergence point when a barge-in fires mid-response.
+     */
+    private lastCommittedTokenIndex;
+    private agentSpeakingUntilMs;
+    private agentSpeakingTimer;
+    private phraseFlushTimer;
+    constructor(config: SchedulerConfig, deps: SchedulerDeps, events?: SchedulerEvents);
+    accept(token: TextToken, acceptedAt?: number): Promise<void>;
+    reject(range: RejectedTokenRange): Promise<void>;
+    flushPending(): Promise<void>;
+    waitIdle(): Promise<void>;
+    synthesizeText(text: string, signal?: AbortSignal): Promise<AudioChunk>;
+    prewarmPhrases(texts: ReadonlyArray<string>, opts?: {
+        concurrency?: number;
+    }): Promise<{
+        warmed: number;
+        cached: number;
+    }>;
+    tickKernel(): void;
+    kernelTickCount(): number;
+    /**
+     * Mark the agent as audibly speaking for the duration of audio handed to the
+     * sink. This is the barge-in gate: VAD blips only pause/resume TTS while this
+     * flag is true, and ASR-confirmed words hard-stop playback plus generation.
+     */
+    markAgentSpeakingForAudio(samples: number, sampleRate: number): void;
+    /** True while a provisional barge-in has paused TTS playback. */
+    get ttsPaused(): boolean;
+    /**
+     * Drop not-yet-spoken TTS without signalling a barge-in: drain the ring
+     * buffer, reset the chunker, cancel in-flight synthesis. Used by the turn
+     * controller when a speculative response is invalidated (speech resumed) —
+     * the speculative TTS was streamed off a stale partial transcript, so it
+     * must go, but this is not a user barge-in (`onCancel` is NOT fired).
+     */
+    cancelPendingTts(): void;
+    private dispatchPhrase;
+    private runPhraseSynthesis;
+    private synthesizePhraseStream;
+    private isPhraseTracked;
+    private cancelNativeTts;
+    private commitAudio;
+    private onBargeInSignal;
+    private handleBargeIn;
+    private emitTtsCancel;
+    private emitTelemetry;
+    private armPhraseFlushTimer;
+    private clearPhraseFlushTimer;
+    private armAgentSpeakingTimer;
+    private clearAgentSpeaking;
+}
+//# sourceMappingURL=scheduler.d.ts.map

package/dist/services/voice/scheduler.d.ts.map

@@ -0,0 +1 @@
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