@elizaos/plugin-local-inference 2.0.0-beta.1 → 2.0.3-beta.2

package/src/services/voice/types.d.ts

@@ -0,0 +1,643 @@
+export interface TextToken {
+    index: number;
+    text: string;
+    /**
+     * Text-model vocabulary token id, when the producer knows it. ASR
+     * (fused Qwen3-ASR) and the text backbone share the Qwen2 BPE 151 936
+     * vocab + merges (AGENTS.md §1), so an ASR-emitted token id is the same
+     * id the text model would assign — a downstream in-process handoff can
+     * inject `id` directly into the text KV cache without detokenize →
+     * retokenize. Absent for producers that only have surface text (the
+     * word-chunk approximation in `splitTranscriptToTokens`).
+     */
+    id?: number;
+}
+export interface AcceptedToken extends TextToken {
+    acceptedAt: number;
+}
+export interface RejectedTokenRange {
+    fromIndex: number;
+    toIndex: number;
+}
+export interface Phrase {
+    id: number;
+    text: string;
+    fromIndex: number;
+    toIndex: number;
+    terminator: "punctuation" | "max-cap" | "phoneme-stream";
+}
+export interface AudioChunk {
+    phraseId: number;
+    fromIndex: number;
+    toIndex: number;
+    pcm: Float32Array;
+    sampleRate: number;
+}
+/**
+ * Reference-audio-token payload mirrored on `SpeakerPreset` for v2 presets.
+ * Empty (K=0, refT=0, tokens.length=0) when the preset has no reference
+ * audio attached (instruct-only voice or legacy v1 file).
+ */
+export interface SpeakerPresetRefAudioTokens {
+    K: number;
+    refT: number;
+    tokens: Int32Array;
+}
+export interface SpeakerPreset {
+    voiceId: string;
+    embedding: Float32Array;
+    bytes: Uint8Array;
+    /**
+     * Preset file format version. `1` for legacy Kokoro-style presets; `2`
+     * for the OmniVoice freeze format that also carries `refAudioTokens`,
+     * `refText`, and `instruct`. Defaulted to `1` for older readers that
+     * haven't been rebuilt yet.
+     */
+    version?: number;
+    /**
+     * OmniVoice reference-audio-token packet (`[K, refT]` int32). Empty for
+     * v1 files and v2 files that intentionally ship instruct-only voices.
+     * The FFI bridge passes the tokens through to `params.ref_audio_tokens`
+     * + `params.ref_T`.
+     */
+    refAudioTokens?: SpeakerPresetRefAudioTokens;
+    /**
+     * UTF-8 transcript of the reference clip that produced `refAudioTokens`.
+     * The FFI bridge passes this through to `params.ref_text`. Empty when
+     * the preset is instruct-only or v1.
+     */
+    refText?: string;
+    /**
+     * Resolved VoiceDesign instruct string (e.g. `"female, young adult,
+     * american accent, moderate pitch"`). The FFI bridge passes this
+     * through to `params.instruct` instead of the historical "use the
+     * voiceId as the instruct string" misreading.
+     */
+    instruct?: string;
+    /**
+     * Free-form metadata attached at freeze time (codec sha256, corpus hash,
+     * source bundle id, etc.). The runtime never relies on this for
+     * correctness.
+     */
+    metadata?: Record<string, unknown>;
+}
+export interface AudioSink {
+    write(pcm: Float32Array, sampleRate: number): void;
+    drain(): void;
+    bufferedSamples(): number;
+}
+export interface OmniVoiceBackend {
+    synthesize(args: {
+        phrase: Phrase;
+        preset: SpeakerPreset;
+        cancelSignal: {
+            cancelled: boolean;
+        };
+        onKernelTick?: () => void;
+    }): Promise<AudioChunk>;
+}
+/**
+ * One PCM segment delivered by a streaming OmniVoice runtime. This is the
+ * scheduler-facing TypeScript contract for the native streaming ABI extension:
+ * the current v1/batch ABI remains valid, and backends that implement this
+ * seam can additionally surface first-audio before a full phrase finishes.
+ */
+export interface TtsPcmChunk {
+    pcm: Float32Array;
+    sampleRate: number;
+    isFinal: boolean;
+}
+export interface StreamingTtsBackend {
+    synthesizeStream(args: {
+        phrase: Phrase;
+        preset: SpeakerPreset;
+        cancelSignal: {
+            cancelled: boolean;
+        };
+        onChunk: (chunk: TtsPcmChunk) => boolean | undefined;
+        onKernelTick?: () => void;
+    }): Promise<{
+        cancelled: boolean;
+    }>;
+}
+/** Opaque native handle for a streaming ASR session in the v2 ABI shape. */
+export type StreamingAsrHandle = bigint;
+/**
+ * TS-only v2 streaming ABI contract. Implementations can satisfy this beside
+ * the existing synchronous v1 methods; callers should test the support flags
+ * rather than probe-and-catch. Native bindings may carry context handles on
+ * top of this shape; the scheduler-facing stream semantics stay the same.
+ */
+export interface VoiceStreamingAbiV2 {
+    ttsStreamSupported(): boolean;
+    ttsSynthesizeStream(args: {
+        text: string;
+        speakerPresetId: string | null;
+        onChunk: (chunk: {
+            pcm: Float32Array;
+            isFinal: boolean;
+        }) => boolean | undefined;
+    }): {
+        cancelled: boolean;
+    };
+    cancelTts(): void;
+    asrStreamSupported(): boolean;
+    asrStreamOpen(args: {
+        sampleRateHz: number;
+    }): StreamingAsrHandle;
+    asrStreamFeed(args: {
+        stream: StreamingAsrHandle;
+        pcm: Float32Array;
+    }): void;
+    asrStreamPartial(args: {
+        stream: StreamingAsrHandle;
+        maxTextBytes?: number;
+        maxTokens?: number;
+    }): {
+        partial: string;
+        tokens?: number[];
+    };
+    asrStreamFinish(args: {
+        stream: StreamingAsrHandle;
+        maxTextBytes?: number;
+        maxTokens?: number;
+    }): {
+        partial: string;
+        tokens?: number[];
+    };
+    asrStreamClose(stream: StreamingAsrHandle): void;
+}
+export interface TranscriptionAudio {
+    pcm: Float32Array;
+    sampleRate: number;
+}
+export type VoiceInputKind = "local_mic" | "discord" | "telegram" | "signal" | "whatsapp" | "phone" | "browser" | "file" | "unknown";
+/**
+ * Where speech audio entered the voice loop. Keep this structural so local
+ * mic, Discord, phone, and connector captures can share the same
+ * turn-taking and attribution path without branching on prompt text.
+ */
+export interface VoiceInputSource {
+    kind: VoiceInputKind;
+    /** Connector account, device, guild/channel, call, or upload id. */
+    sourceId?: string;
+    roomId?: string;
+    conversationId?: string;
+    messageId?: string;
+    deviceId?: string;
+    connectorAccountId?: string;
+    channelId?: string;
+    guildId?: string;
+    callId?: string;
+    participantId?: string;
+    metadata?: Record<string, unknown>;
+}
+/**
+ * Speaker attribution for diarized speech. `imprintClusterId` is evidence,
+ * not identity: callers that want to attach this to a LifeOps person must
+ * submit a normal `EntityStore.observeIdentity` observation with this
+ * cluster/observation id in its evidence list. Do not use voice imprints as
+ * a parallel identity graph or as authorization for voice synthesis.
+ */
+export interface VoiceSpeaker {
+    id: string;
+    label?: string;
+    displayName?: string;
+    source?: VoiceInputSource;
+    imprintClusterId?: string;
+    imprintObservationId?: string;
+    entityId?: string;
+    confidence?: number;
+    isLocalUser?: boolean;
+    metadata?: Record<string, unknown>;
+}
+/** One diarized span within a transcript snapshot or finalized voice turn. */
+export interface VoiceSegment {
+    id?: string;
+    text: string;
+    startMs: number;
+    endMs: number;
+    speaker?: VoiceSpeaker;
+    speakerId?: string;
+    source?: VoiceInputSource;
+    confidence?: number;
+    tokens?: number[];
+    metadata?: Record<string, unknown>;
+}
+export interface VoiceDiarizationMetadata {
+    provider: "local" | "connector" | "cloud" | "unknown";
+    model?: string;
+    version?: string;
+    confidence?: number;
+    metadata?: Record<string, unknown>;
+}
+export interface VoiceTurnMetadata {
+    turnId?: string;
+    source?: VoiceInputSource;
+    primarySpeaker?: VoiceSpeaker;
+    segments?: VoiceSegment[];
+    startedAtMs?: number;
+    endedAtMs?: number;
+    diarization?: VoiceDiarizationMetadata;
+    metadata?: Record<string, unknown>;
+}
+/** A running or final transcript snapshot from a `StreamingTranscriber`. */
+export interface TranscriptUpdate {
+    /** The full running transcript (not a delta) at this point. */
+    partial: string;
+    /** True for the snapshot emitted by `flush()` / on `speech-end`. */
+    isFinal: boolean;
+    /** Channel/device/call metadata for attribution and storage. */
+    source?: VoiceInputSource;
+    /** Best speaker attribution for single-speaker snapshots. */
+    speaker?: VoiceSpeaker;
+    /** Diarized spans for multi-speaker snapshots, when available. */
+    segments?: VoiceSegment[];
+    /** Turn-level metadata carried through to generation and storage. */
+    turn?: VoiceTurnMetadata;
+    /**
+     * Text-model token ids for `partial`, when the backend can supply them
+     * cheaply (fused Qwen3-ASR shares the text vocabulary). Absent when the
+     * decoder reports surface text only (re-tokenization is the LLM stage's
+     * job there).
+     */
+    tokens?: number[];
+    /**
+     * Voice-side emotion attribution attached to `isFinal` snapshots only.
+     * Running partials never carry this — the acoustic classifier wants a
+     * stable utterance window and the lexicon read on partial text is noise.
+     * Produced by `attributeVoiceEmotion()` after fusing the acoustic
+     * classifier output (`VoiceEmotionClassifier`) with text-side evidence;
+     * the fusion rule lives in `emotion-attribution.ts` so no consumer
+     * re-implements it. See R3-emotion §3 + §5.
+     */
+    voiceEmotion?: import("./emotion-attribution").VoiceEmotionAttribution;
+}
+/** Events a `StreamingTranscriber` emits while consuming PCM frames. */
+export type TranscriberEvent = {
+    kind: "partial";
+    update: TranscriptUpdate;
+} | {
+    kind: "final";
+    update: TranscriptUpdate;
+}
+/**
+ * Fired the first instant ≥1 real word is recognized in the current
+ * speech segment. Wired to W1's barge-in word-confirm gate
+ * (`onWordsDetected`) so the agent hard-stops TTS + aborts in-flight
+ * LLM/drafter generation only on real speech, not a blip.
+ */
+ | {
+    kind: "words";
+    words: string[];
+};
+export type TranscriberEventListener = (event: TranscriberEvent) => void;
+/**
+ * Live transcription. `feed()` is called per PCM frame off a `MicSource`.
+ * The adapter runs windowed decode passes internally and emits `partial`
+ * events as the running transcript grows; `flush()` force-finalizes (call
+ * it when the VAD reports `speech-end`). Implementations gate on the VAD
+ * event stream — they only decode while the VAD is in `speech-active`.
+ *
+ * No silent degrade: a transcriber whose backend is unavailable throws on
+ * construction (or on first `feed`), it does not quietly produce empty
+ * transcripts.
+ */
+export interface StreamingTranscriber {
+    /** Feed one PCM frame. Frames received while VAD is not active are buffered/ignored per the VAD-gating policy. */
+    feed(frame: PcmFrame): void;
+    /**
+     * Force-finalize: drain any buffered audio, run a final decode pass,
+     * emit the `final` event, and resolve with the final transcript. Safe
+     * to call when no audio is buffered (resolves with an empty final).
+     * After `flush()` the transcriber is reset and ready for the next
+     * speech segment.
+     */
+    flush(): Promise<TranscriptUpdate>;
+    /** Subscribe to transcriber events. Returns an unsubscribe fn. */
+    on(listener: TranscriberEventListener): () => void;
+    /** Release any held native resources (FFI stream handle, temp files). Idempotent. */
+    dispose(): void;
+}
+export interface PhraseChunkerConfig {
+    /**
+     * Hard word cap before a phrase is force-flushed even without a
+     * `, . ! ? ; :` boundary. Defaults to 30 (the brief's A6 "first 30 words").
+     */
+    maxTokensPerPhrase?: number;
+    /**
+     * Characters that close a phrase. Default `, . ! ? ; :` — punctuation
+     * boundaries let the first clause reach TTS without waiting for a
+     * sentence-final mark.
+     */
+    sentenceTerminators?: ReadonlySet<string>;
+    /**
+     * Where the chunker emits a phrase boundary.
+     *   'punctuation'    — default. Wait for `, . ! ? ; :` or the max-token cap.
+     *   'phoneme-stream' — additionally emit a sub-phrase chunk every
+     *                      `phonemesPerChunk` phonemes. Cuts first-audio
+     *                      latency by handing partial phrases to TTS at
+     *                      phoneme boundaries.
+     */
+    chunkOn?: "punctuation" | "phoneme-stream";
+    /** Phonemes per chunk in `phoneme-stream` mode. Default 8. */
+    phonemesPerChunk?: number;
+    /**
+     * Maximum milliseconds a phrase may sit in the chunker before the
+     * scheduler force-flushes it even without punctuation / phoneme / cap
+     * boundaries. Default 700 ms. Set to 0 to disable.
+     */
+    maxAccumulationMs?: number;
+    /**
+     * Shorter budget applied ONLY to the first phrase of each reply, so first
+     * audio (TTFA) plays sooner on punctuation-sparse openings while later
+     * phrases keep `maxAccumulationMs` (no fragmentation). When omitted,
+     * derives from `maxAccumulationMs` (half, capped at 350 ms) and honors the
+     * `ELIZA_PHRASE_FLUSH_FIRST_MS` env override. Clamped to `maxAccumulationMs`.
+     */
+    firstPhraseMaxAccumulationMs?: number;
+}
+export interface VerifierStreamEvent {
+    kind: "accept" | "reject";
+    tokens: TextToken[];
+    /**
+     * Optional per-event metadata. Today only the very first `accept` of a
+     * streaming completion carries `firstTokenMs` (L5 — time from the fetch
+     * being issued to the first SSE chunk arriving). Other consumers MAY
+     * ignore this field; producers MUST omit it on non-first events.
+     */
+    meta?: {
+        /** Milliseconds from request issue (`performance.now()`) to first chunk. */
+        firstTokenMs?: number;
+    };
+}
+/** A fixed-size block of mono PCM samples in [-1, 1] at a known sample rate. */
+export interface PcmFrame {
+    pcm: Float32Array;
+    sampleRate: number;
+    /**
+     * Monotonic timestamp (ms, `performance.now()` domain) of the *first*
+     * sample in this frame. Used to age VAD events and barge-in latency.
+     */
+    timestampMs: number;
+}
+/**
+ * Event emitted by `VadDetector` on the authoritative (Silero) timeline.
+ *
+ *   - `speech-start`  — speech onset (a run of speech frames crossed the
+ *                       onset threshold). Carries the probability of the
+ *                       triggering frame.
+ *   - `speech-active` — a periodic heartbeat while speech is ongoing. The
+ *                       barge-in controller uses this to pause TTS.
+ *   - `speech-pause`  — speech has been quiet for `pauseStartedMs..now`
+ *                       but not long enough to count as end-of-utterance.
+ *                       The turn controller uses this to kick a speculative
+ *                       response off the partial transcript.
+ *   - `speech-end`    — end of utterance (silence held past the hangover
+ *                       window). Carries the total speech duration.
+ *   - `blip`          — a short burst of energy that the Silero VAD rejected
+ *                       (or that was too short to be speech). The barge-in
+ *                       controller treats this as "resume TTS".
+ */
+export type VadEvent = {
+    type: "speech-start";
+    timestampMs: number;
+    probability: number;
+} | {
+    type: "speech-active";
+    timestampMs: number;
+    probability: number;
+    speechDurationMs: number;
+} | {
+    type: "speech-pause";
+    timestampMs: number;
+    pauseDurationMs: number;
+} | {
+    type: "speech-end";
+    timestampMs: number;
+    speechDurationMs: number;
+} | {
+    type: "blip";
+    timestampMs: number;
+    durationMs: number;
+    peakRms: number;
+};
+/** Cheap RMS energy gate event — the fast pre-warm path. Distinct timeline
+ *  from `VadEvent`; this fires with sub-frame latency and never blocks on a
+ *  model forward pass. */
+export type EnergyGateEvent = {
+    type: "energy-rise";
+    timestampMs: number;
+    rms: number;
+} | {
+    type: "energy-fall";
+    timestampMs: number;
+    quietMs: number;
+};
+export type VadEventListener = (event: VadEvent) => void;
+export type EnergyGateListener = (event: EnergyGateEvent) => void;
+/**
+ * Subscribable VAD event stream. `VadDetector` (`voice/vad.ts`) is the
+ * concrete implementation; the streaming transcriber and the barge-in
+ * controller take this structural view so they don't pull in the optional
+ * `onnxruntime-node` surface.
+ */
+export interface VadEventSource {
+    onVadEvent(listener: VadEventListener): () => void;
+}
+/**
+ * Source of mic PCM. The desktop/Electrobun impl in `mic-source.ts` is the
+ * first concrete implementation; Discord / Telegram / mobile connectors
+ * implement the same interface so the rest of the voice loop is source-
+ * agnostic. A `MicSource` produces fixed-size mono frames at a fixed sample
+ * rate and tees them to any number of consumers (the VAD, the ring buffer
+ * the ASR reads from, instrumentation taps).
+ */
+export interface MicSource {
+    /** Nominal sample rate of every emitted frame (Hz). */
+    readonly sampleRate: number;
+    /** Samples per emitted frame. */
+    readonly frameSamples: number;
+    /** True once `start()` has resolved and frames are flowing. */
+    readonly running: boolean;
+    /** Begin capture. Resolves when the underlying device is producing audio.
+     *  Throws (never silently no-ops) when no mic backend is available. */
+    start(): Promise<void>;
+    /** Stop capture and release the device. Idempotent. */
+    stop(): Promise<void>;
+    /** Subscribe to PCM frames. Returns an unsubscribe function. */
+    onFrame(listener: (frame: PcmFrame) => void): () => void;
+    /** Subscribe to fatal capture errors (device lost, process died). The
+     *  source is no longer `running` after one of these. */
+    onError(listener: (error: Error) => void): () => void;
+}
+/**
+ * Cancellation token threaded from the barge-in controller down through the
+ * voice scheduler (TTS) *and* the engine layer (in-flight LLM / MTP
+ * drafter generation). `cancelled` is a plain boolean so the synthesis loop
+ * and the SSE-consuming generate loop can both poll it cheaply at a kernel
+ * boundary; `reason` records *why* for diagnostics; `signal` is the standard
+ * `AbortSignal` the engine's HTTP/stream layer aborts on.
+ *
+ * (W1 owns the controller; W9 threads `signal` into `dispatcher.generate`.)
+ */
+export interface BargeInCancelToken {
+    cancelled: boolean;
+    reason: "barge-in-words" | "manual" | null;
+    readonly signal: AbortSignal;
+}
+/** Signal emitted by `BargeInController` to the scheduler / engine. */
+export type BargeInSignal = {
+    type: "pause-tts";
+    timestampMs: number;
+} | {
+    type: "resume-tts";
+    timestampMs: number;
+} | {
+    type: "hard-stop";
+    timestampMs: number;
+    token: BargeInCancelToken;
+};
+export type BargeInSignalListener = (signal: BargeInSignal) => void;
+/**
+ * Contract the ASR layer (W2's `StreamingTranscriber`) calls into the
+ * barge-in controller with. When the transcriber has parsed at least one
+ * real word from the user's barge-in audio, it calls `onWordsDetected` with
+ * the running word count; the controller promotes a `pause-tts` into a
+ * `hard-stop`. This is the *authoritative* blip-vs-words gate — the energy-
+ * duration heuristic is only a fast provisional guess until ASR confirms.
+ */
+export interface WordsDetectedSink {
+    onWordsDetected(args: {
+        /** Number of parsed words observed so far in this barge-in segment. */
+        wordCount: number;
+        /** Best partial transcript so far (may be empty). */
+        partialText: string;
+        timestampMs: number;
+    }): void;
+}
+export interface SchedulerConfig {
+    chunkerConfig: PhraseChunkerConfig;
+    preset: SpeakerPreset;
+    ringBufferCapacity: number;
+    sampleRate: number;
+    /**
+     * Max concurrent TTS dispatches. When this many phrases are in flight,
+     * `accept()` awaits the oldest before dispatching the next, propagating
+     * backpressure upstream to the verifier loop. Default 4 — small enough
+     * to bound memory under runaway producers without serialising the
+     * common case (text gen leads TTS by a phrase or two).
+     */
+    maxInFlightPhrases?: number;
+    /**
+     * Enable the streaming-TTS path (`synthesizeStream`) for phrase
+     * synthesis. When `true` (default), the scheduler uses the chunk-by-chunk
+     * streaming ABI when the backend supports it, delivering first audio
+     * before the full phrase finishes synthesizing and enabling per-chunk
+     * prefix-preserving barge-in rollback.
+     *
+     * Previously this was implicitly gated by `ttsStreamSupported()` from the
+     * native FFI layer. On macOS, a `ggml_conv_transpose_1d` stall in the
+     * DAC codec region caused the Metal path to hang — that stall is now
+     * fixed in the llama.cpp merge (native Metal kernels for
+     * `ggml_conv_transpose_1d`; the CPU fallback causing the hang is gone).
+     * The flag is therefore `true` by default. Set to `false` only when
+     * testing against a non-streaming build or reproducing the pre-fix
+     * behaviour.
+     */
+    streamingTtsActive?: boolean;
+}
+export interface VoiceSchedulerPhraseTelemetry {
+    id: number;
+    text: string;
+    fromIndex: number;
+    toIndex: number;
+    terminator: Phrase["terminator"];
+    tokenCount: number;
+    textBytes: number;
+}
+export type VoiceAudioSource = "cache" | "synthesis";
+export type VoiceTtsCancelReason = "barge-in" | "rollback" | "pending-tts" | "synthesis-cancelled";
+export type VoiceSchedulerTelemetryEvent = {
+    type: "phrase-dispatch";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+    inFlightPhrases: number;
+} | {
+    type: "phrase-cache-hit" | "phrase-cache-miss";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+} | {
+    type: "tts-start";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+    inFlightPhrases: number;
+} | {
+    type: "tts-first-audio";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+    source: VoiceAudioSource;
+    samples: number;
+    sampleRate: number;
+} | {
+    type: "audio-committed";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+    source: VoiceAudioSource;
+    samples: number;
+    sampleRate: number;
+    flushedSamples: number;
+    paused: boolean;
+    ringBufferSamples: number;
+    sinkBufferedSamples: number;
+} | {
+    type: "tts-cancel";
+    atMs: number;
+    phrase: VoiceSchedulerPhraseTelemetry;
+    reason: VoiceTtsCancelReason;
+} | {
+    type: "rollback";
+    atMs: number;
+    phraseId: number;
+    range: RejectedTokenRange;
+    reason: "rejected-tokens";
+} | {
+    type: "barge-in";
+    atMs: number;
+    ringBufferSamplesDrained: number;
+    sinkBufferedSamplesDrained: number;
+    inFlightPhrasesCancelled: number;
+    wasPaused: boolean;
+} | {
+    /**
+     * Fired when the prefix-preserving rollback queue partitions
+     * in-flight audio chunks on barge-in. `retainedChunks` are replayed
+     * into the sink; `droppedChunks` are discarded. Present only when
+     * `PrefixPreservingQueue` is active (at least one chunk was tagged).
+     */
+    type: "barge-in-prefix-rollback";
+    atMs: number;
+    divergencePoint: number;
+    retainedChunks: number;
+    droppedChunks: number;
+    straddledChunks: number;
+    retainedDurationMs: number;
+    droppedDurationMs: number;
+};
+export type VoiceSchedulerTelemetryListener = (event: VoiceSchedulerTelemetryEvent) => void;
+/** Minimal VAD model contract consumed by the fused `GgmlSileroVad` and the
+ *  optional injected qwen-toolkit adapter. */
+export interface VadLike {
+    readonly windowSamples: number;
+    readonly sampleRate: number;
+    process(window: Float32Array): Promise<number>;
+    reset(): void;
+}
+/** Minimal wake-word model contract consumed by OpenWakeWordGgmlModel. */
+export interface WakeWordModel {
+    readonly frameSamples: number;
+    readonly sampleRate: number;
+    scoreFrame(frame: Float32Array): Promise<number>;
+    reset(): void;
+}
+//# sourceMappingURL=types.d.ts.map

package/src/services/voice/types.d.ts.map

@@ -0,0 +1 @@
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