@elizaos/plugin-local-inference 2.0.3-beta.2 → 2.0.3-beta.3

package/dist/services/voice/kokoro/kokoro-ffi-runtime.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/kokoro-runtime.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/phoneme-stream.d.ts

@@ -0,0 +1,51 @@
+/**
+ * Phoneme streaming for Kokoro-82M.
+ *
+ * Kokoro consumes a sequence of phoneme ids (espeak-ng IPA tokenised against a
+ * small fixed vocab). The scheduler emits phrases at punctuation or
+ * `phoneme-stream` boundaries (see `voice/phrase-chunker.ts` `chunkOn`
+ * option). This module is the seam between those phrase boundaries and the
+ * model's input tensor:
+ *
+ *   text → phonemizer.phonemize() → KokoroPhonemeSequence (ids) → runtime
+ *
+ * For maximum responsiveness the runtime can call `streamPhonemes()` against
+ * an async text iterator (chunked draft tokens) and forward each window of
+ * accumulated ids as soon as a phoneme boundary fires. The default `flushAt`
+ * is one phoneme — i.e. emit progress per id — but production deployments
+ * lift this to ~8 phonemes to amortise the ONNX forward pass on small
+ * windows. This file intentionally has no dependency on the rest of the
+ * voice scaffold so it can be reused by the fine-tune evaluator script.
+ */
+import type { KokoroPhonemizer } from "./types";
+export interface PhonemeStreamWindow {
+    /** Cumulative ids since stream start. The runtime can re-tokenise or
+     *  carry state by id; the simplest implementation forwards the full
+     *  window each call. */
+    ids: Int32Array;
+    /** Cumulative phoneme string for debugging / display. */
+    phonemes: string;
+    /** True for the final window in the stream. */
+    isFinal: boolean;
+}
+export interface StreamPhonemesOptions {
+    phonemizer: KokoroPhonemizer;
+    lang: string;
+    /** Emit a window every N new phoneme ids. Default 8 (≈ first audio after a
+     *  short syllable cluster — matches the phrase chunker's default cap). */
+    flushAt?: number;
+}
+/**
+ * Phonemize an async text source and emit cumulative windows. The caller
+ * consumes the iterator with `for await (const window of streamPhonemes(…))`.
+ * A pull-style API keeps this independent of the scheduler's event loop —
+ * the bench harness and the eval loop both reuse it without taking on a
+ * scheduler dependency.
+ */
+export declare function streamPhonemes(textChunks: AsyncIterable<string>, opts: StreamPhonemesOptions): AsyncIterable<PhonemeStreamWindow>;
+/** Synchronous variant for whole-phrase callers (the scheduler dispatches
+ *  one phrase at a time in the default `punctuation` mode). Returns the
+ *  full id array — equivalent to draining `streamPhonemes` on a single-item
+ *  iterator and taking the last window. */
+export declare function phonemizePhrase(text: string, opts: StreamPhonemesOptions): Promise<PhonemeStreamWindow>;
+//# sourceMappingURL=phoneme-stream.d.ts.map

package/dist/services/voice/kokoro/phoneme-stream.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/phonemizer.d.ts.map

@@ -0,0 +1 @@
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package/{src → dist}/services/voice/kokoro/pick-runtime.d.ts

@@ -6,7 +6,7 @@
  *   ffi   (default)  → KokoroFfiRuntime → in-process synthesis through the
  *                       fused `libelizainference` handle (the
  *                       `eliza_inference_kokoro_*` exports, introduced at ABI
- *                       v10; the fused library is currently ABI v11). This is
+ *                       v10; the fused library is currently ABI v12). This is
  *                       the SOLE synthesis path on every platform.
  *   mock             → KokoroMockRuntime. Tests only.
  *

package/dist/services/voice/kokoro/pick-runtime.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/runtime-selection.d.ts

@@ -0,0 +1,31 @@
+/**
+ * Voice backend selection — Kokoro is the only on-device TTS backend.
+ *
+ * OmniVoice TTS was retired (it was an autoregressive LM-based synth: heavier,
+ * slower TTFB, and only it could voice-clone). Kokoro (StyleTTS2 / iSTFTNet,
+ * non-autoregressive) is faster, smaller, and ships in every bundle — desktop
+ * and mobile alike — so the selector collapses to a single auditable answer.
+ * The function + env reader are kept (rather than inlined) so the engine layer
+ * and tests retain one seam to ask "is a TTS backend available?".
+ */
+export type VoiceBackendChoice = "kokoro";
+/** Retained for the env override; `auto` and `kokoro` both resolve to Kokoro. */
+export type VoiceBackendMode = "kokoro" | "auto";
+export interface VoiceBackendInputs {
+    /** Caller-set mode. Defaults to `auto`; both modes resolve to Kokoro. */
+    mode?: VoiceBackendMode;
+    /** Whether Kokoro model artifacts are present on disk. The selector throws
+     *  rather than returning a backend when this is `false` — no silent downgrade. */
+    kokoroAvailable: boolean;
+    /** True on mobile (iOS / Android) builds — informational only now. */
+    mobile?: boolean;
+}
+export interface VoiceBackendDecision {
+    backend: VoiceBackendChoice;
+    /** One-line reason — surfaced to telemetry. */
+    reason: string;
+}
+/** Resolve the env override (`ELIZA_TTS_BACKEND=kokoro|auto`). */
+export declare function readVoiceBackendModeFromEnv(env?: NodeJS.ProcessEnv): VoiceBackendMode | undefined;
+export declare function selectVoiceBackend(inputs: VoiceBackendInputs): VoiceBackendDecision;
+//# sourceMappingURL=runtime-selection.d.ts.map

package/dist/services/voice/kokoro/runtime-selection.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/types.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/voice-presets.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/kokoro/voices.d.ts.map

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

@@ -0,0 +1 @@
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package/dist/services/voice/live-diarization-session.d.ts

@@ -0,0 +1,196 @@
+/**
+ * Live on-device diarization session — the agent-process owner of an
+ * {@link AudioFrameConsumer} wired to the REAL fused VAD / encoder / diarizer /
+ * attribution stack.
+ *
+ * The Android `audioFrame` PCM stream is produced in the Capacitor WebView
+ * (JS renderer) but the voice FFI runs in the embedded bun agent process. The
+ * agent's `/api/voice/audio-frames` route pumps batched frames into the single
+ * session this module owns, where the consumer segments turns, runs
+ * diarization + speaker attribution, and emits VOICE_TURN_OBSERVED.
+ *
+ * This module is the agent-side mirror of the host smoke harness
+ * (`packages/app-core/scripts/voice-attribution-smoke.ts`): same real models,
+ * same consumer, fed live frames over HTTP instead of a WAV.
+ *
+ * Single fused engine: VAD, the WeSpeaker speaker encoder, and the pyannote
+ * diarizer all run through the ONE fused `libelizainference` handle via its
+ * `eliza_inference_vad_*` / `_speaker_*` / `_diariz_*` ABI (the user directive:
+ * no separate bun:ffi-musl libs). Resolution:
+ *   - fused lib: `$ELIZA_INFERENCE_LIBRARY` (exact) or `$ELIZA_INFERENCE_LIB_DIR`
+ *     (dir) — both exported by ElizaAgentService on Android to the app
+ *     nativeLibraryDir.
+ *   - context bundle root: `$ELIZA_VOICE_MODEL_DIR` (the same dir the GGUFs
+ *     live under); the fused runtime resolves the per-model GGUFs from there.
+ */
+import { type AttributionPipelineLike, type AudioFrameConsumerDeps, type AudioFrameEvent, type EchoReferenceProvider, type RuntimeEventSink, type TurnTranscriber, type VadSegmenter } from "./audio-frame-consumer.js";
+export type { RuntimeEventSink } from "./audio-frame-consumer.js";
+export interface LiveDiarizationStatus {
+    /** True once the consumer + real fused deps are loaded and accepting frames. */
+    ready: boolean;
+    /** Resolved fused-library path (null when it could not be resolved). */
+    libs: {
+        fusedInference: string | null;
+    };
+    /** Resolved context-bundle dir for the fused runtime. */
+    models: {
+        dir: string;
+    };
+    /** Frames received from the WebView across this session. */
+    framesReceived: number;
+    /** Frames dropped at the decode boundary. */
+    framesDropped: number;
+    /** Turns segmented + attributed so far. */
+    turnsObserved: number;
+    /** Live AEC wiring status. Echo cancellation runs only when this is true. */
+    aec: {
+        echoReferenceWired: boolean;
+        /** Playback→mic delay (samples @16 kHz) currently applied to align the
+         * far-end reference — self-calibrated from real echo when confident,
+         * otherwise the `ELIZA_VOICE_ECHO_DELAY_MS` seed (default 0). */
+        echoDelaySamples: number;
+        /** Peak cross-correlation [0,1] of the last accepted delay calibration;
+         * 0 until a confident estimate replaces the seed. */
+        echoDelayConfidence: number;
+    };
+    /** The most recent attributed turns (capped), for device-evidence reads. */
+    recentTurns: LiveDiarizationTurnSummary[];
+    /** Populated only when readiness failed — the precise blocker. */
+    error?: string;
+}
+/** A compact, JSON-safe summary of one attributed turn (no PCM/embeddings). */
+export interface LiveDiarizationTurnSummary {
+    turnId: string;
+    startedAtMs: number;
+    endedAtMs: number;
+    samples: number;
+    durationMs: number;
+    hasSpeaker: boolean;
+    speakerEntityId: string | null;
+    speakerConfidence: number | null;
+    segments: number;
+    agentShouldSpeak: boolean | null;
+    nextSpeaker: string | null;
+}
+export interface LiveDiarizationSessionOptions {
+    /**
+     * Agent-playback PCM provider for AEC. The caller owns playback capture and
+     * delay calibration when supplied. Without an external provider, the session
+     * uses its built-in playback buffer fed by /api/voice/playback-frames.
+     */
+    echoReference?: EchoReferenceProvider | null;
+}
+export interface LiveDiarizationConsumerDepsInput {
+    vad: VadSegmenter;
+    pipeline: AttributionPipelineLike;
+    runtime: RuntimeEventSink;
+    transcribe?: TurnTranscriber | null;
+    echoReference?: EchoReferenceProvider | null;
+}
+export declare function buildLiveDiarizationConsumerDeps({ vad, pipeline, runtime, transcribe, echoReference, }: LiveDiarizationConsumerDepsInput): AudioFrameConsumerDeps;
+/**
+ * Owns the single live diarization consumer for the agent process. Built
+ * lazily on first frame batch so it does not load voice models at boot.
+ */
+export declare class LiveDiarizationSession {
+    private readonly runtime;
+    private readonly options;
+    private consumer;
+    private ffi;
+    private ctx;
+    private encoder;
+    private diarizer;
+    private vad;
+    private building;
+    private framesReceived;
+    private turnsObserved;
+    private readonly recentTurns;
+    private resolvedLibPath;
+    private buildError;
+    /** True once the fused ASR region is mmap-acquired for per-turn transcribe. */
+    private asrRegionAcquired;
+    /**
+     * Far-end (agent TTS playback) alignment buffer for echo cancellation
+     * (#9583/#9455). Fed by {@link pushPlayback}; read per mic frame via the
+     * consumer's `echoReference` seam. Inert (zero far-end ⇒ NLMS passthrough)
+     * until the device streams playback, so wiring it never regresses the
+     * no-playback case.
+     */
+    private readonly echoBuffer;
+    /**
+     * Playback→mic delay applied when reading the far-end reference. Seeded from
+     * `ELIZA_VOICE_ECHO_DELAY_MS` (default 0) and then SELF-CALIBRATED on the live
+     * path: once enough playback-active echo is observed, `estimateEchoDelaySamples`
+     * (#9586) recovers the bulk transport lag by cross-correlation and replaces the
+     * seed (#9583). Mutable for that reason.
+     */
+    private echoDelaySamples;
+    private echoDelayConfidence;
+    private echoDelayCalibrated;
+    /** Rolling near/far windows accumulated only while the far-end is active, used
+     * once to estimate the playback→mic delay. Cleared after a confident estimate
+     * and on {@link resetPlayback}. */
+    private calNear;
+    private calFar;
+    private calSampleCount;
+    constructor(runtime: RuntimeEventSink, options?: LiveDiarizationSessionOptions);
+    /** Ensure the real-deps consumer exists; idempotent + concurrency-safe. */
+    private ensureBuilt;
+    private build;
+    /**
+     * Build a per-turn ASR transcriber over the fused batch decoder
+     * (`eliza_inference_asr_transcribe`). Returns null when the fused build
+     * exposes no ASR decoder; acquiring the ASR mmap region is best-effort (a
+     * missing bundled ASR model leaves the path diarization-only rather than
+     * failing the whole session). One batch decode per finalized turn — the turn
+     * is already fully buffered for attribution, so no streaming state is needed.
+     */
+    private buildTurnTranscriber;
+    private recordTurn;
+    /**
+     * The far-end (agent TTS playback) reference aligned to a mic frame of
+     * `samples` samples — the consumer's `echoReference` seam (#9455/#9583).
+     * Reads the alignment buffer at the configured playback→mic delay; the slice
+     * is zero-filled (⇒ NLMS passthrough) until the device streams playback.
+     * Public so the wiring is unit-testable without the fused FFI.
+     */
+    echoReferenceFrame(timestampMs: number, samples: number): Float32Array;
+    /** Current self-calibrated AEC delay state (for status + tests). */
+    aecDelayState(): {
+        delaySamples: number;
+        confidence: number;
+        calibrated: boolean;
+    };
+    /**
+     * Self-calibrate the playback→mic delay (#9583/#9586) from real echo. Called
+     * per mic frame while uncalibrated: when the far-end is active (the agent is
+     * playing TTS), accumulate the time-aligned near/far windows; once ~0.75 s of
+     * playback-active audio is buffered, recover the bulk transport lag by
+     * cross-correlation and, if confident, replace the static seed. One-shot — the
+     * device's speaker→mic path is stable, so we lock the first confident estimate
+     * and stop re-measuring. Public so it can be unit-tested without the fused FFI.
+     */
+    observeForDelayCalibration(nearPcm: Float32Array, timestampMs: number): void;
+    /**
+     * Feed a batch of agent-playback (far-end) frames for echo cancellation. The
+     * device captures the agent's TTS output in the SAME base64 LE-s16 16 kHz
+     * mono wire format as the mic and POSTs it in real time as it renders; we
+     * decode + append to the alignment buffer. The device MUST also call
+     * {@link resetPlayback} when playback stops (or on barge-in) so the canceller
+     * never aligns a later mic frame to stale, no-longer-playing audio.
+     */
+    pushPlayback(frames: AudioFrameEvent[]): void;
+    /** Drop buffered far-end playback (playback stopped / barge-in). Also clears
+     * the in-progress delay-calibration window (it would otherwise straddle a
+     * playback gap); the already-learned delay is kept. */
+    resetPlayback(): void;
+    /** Feed a batch of WebView-captured frames; resolves once VAD has processed them. */
+    ingest(frames: AudioFrameEvent[]): Promise<void>;
+    /** Flush any open segment (call on stopAudioFrames) and await attribution. */
+    flush(): Promise<void>;
+    /** Build (if needed) and report status — the device-evidence read. */
+    status(): Promise<LiveDiarizationStatus>;
+    /** Release native handles + listeners. */
+    close(): Promise<void>;
+}
+//# sourceMappingURL=live-diarization-session.d.ts.map

package/dist/services/voice/live-diarization-session.d.ts.map

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package/dist/services/voice/metric-math.d.ts

@@ -0,0 +1,10 @@
+/**
+ * Pure rounding + percentile helpers shared by the voice E2E harness and the
+ * voice workbench report (#8785). No models, filesystem, or network — just
+ * numeric formatting so both consumers report identical metric values.
+ */
+export declare function round1(value: number): number;
+export declare function round4(value: number): number;
+/** Nearest-rank percentile over a sample (null when empty), non-finite filtered. */
+export declare function percentile(values: ReadonlyArray<number>, p: number): number | null;
+//# sourceMappingURL=metric-math.d.ts.map

package/dist/services/voice/metric-math.d.ts.map

@@ -0,0 +1 @@
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package/{src → dist}/services/voice/mic-source.d.ts

@@ -54,7 +54,7 @@ declare abstract class BaseMicSource implements MicSource {
     protected emitError(error: Error): void;
 }
 export interface DesktopMicSourceOptions {
-    /** Capture sample rate. Default 16 kHz (what the VAD + Qwen3-ASR expect). */
+    /** Capture sample rate. Default 16 kHz (what VAD + Gemma ASR expect). */
     sampleRate?: number;
     /** Frame duration in ms. Default 32 ms (one Silero window @ 16 kHz). */
     frameMs?: number;

package/dist/services/voice/mic-source.d.ts.map

@@ -0,0 +1 @@
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package/dist/services/voice/nlms-echo-canceller.d.ts

@@ -0,0 +1,137 @@
+/**
+ * nlms-echo-canceller.ts — PCM acoustic echo cancellation for the live
+ * half-duplex voice pipeline (#9455).
+ *
+ * When the agent speaks, its TTS playback leaks back into the microphone and
+ * corrupts ASR / VAD / diarization (the agent hears itself). This is a
+ * single-channel adaptive echo canceller: a normalized least-mean-squares
+ * (NLMS) FIR filter models the playback→mic acoustic path and subtracts the
+ * estimated echo from the near-end (mic) signal sample-by-sample.
+ *
+ *   near-end d[n] = local_speech[n] + echo[n]        (the raw mic)
+ *   far-end  x[n] = agent TTS playback (the reference)
+ *   estimate ŷ[n] = Σ_k w[k]·x[n−k]                  (modeled echo)
+ *   output   e[n] = d[n] − ŷ[n]                       (echo-cancelled mic → ASR)
+ *   update   w[k] += μ·e[n]·x[n−k] / (‖x‖² + ε)       (NLMS adaptation)
+ *
+ * All audio is 16 kHz mono Float32 [-1, 1] — the pipeline's internal format
+ * (see audio-frame-consumer.ts). The filter length must cover the
+ * playback→mic delay plus the room's reverberation tail; for tails longer than
+ * the filter, calibrate `delaySamples` (the bulk transport delay) so the
+ * adaptive taps only have to model the short residual impulse.
+ *
+ * Scope: this targets the dominant failure mode — the agent transcribing its
+ * own TTS while the *user is silent* (echo-only), where it achieves ~29 dB of
+ * echo-return-loss-enhancement. A far-end-vs-near-end double-talk detector
+ * freezes adaptation when a local talker is active so the filter cannot learn
+ * (and cancel) the user's voice; barge-in itself is handled upstream by the
+ * barge-in detector (which stops playback). Full double-talk residual-echo
+ * suppression is AEC3-class work and intentionally out of scope here.
+ *
+ * Pure DSP, zero dependencies — verified by nlms-echo-canceller.test.ts
+ * (ERLE on synthetic echo, passthrough, stability, reset).
+ */
+export interface NlmsEchoCancellerOptions {
+    /** Adaptive FIR length in samples. 256 ≈ 16 ms of impulse response @16 kHz. */
+    filterTaps?: number;
+    /** NLMS step size in (0, 2). Larger = faster adaptation, less stable. */
+    mu?: number;
+    /** Regularization added to the reference energy to avoid divide-by-zero. */
+    epsilon?: number;
+    /**
+     * Bulk playback→mic transport delay in samples. The reference is consumed
+     * `delaySamples` ahead of the near-end so the adaptive taps only model the
+     * residual room impulse, not the (potentially large) transport latency.
+     */
+    delaySamples?: number;
+    /**
+     * Double-talk detector ratio. When the smoothed near-end power exceeds
+     * `dtdRatio`× the smoothed far-end reference power (a passive echo path
+     * attenuates, so echo power stays below the reference), a local talker is
+     * assumed active and adaptation is frozen so the filter cannot learn (and
+     * cancel) the user's voice. Set 0 to disable.
+     */
+    dtdRatio?: number;
+    /**
+     * Opt-in nonlinear residual-echo suppressor (#9583/#9649). After the linear
+     * NLMS subtracts the modeled echo, an echo-only frame (the agent is speaking,
+     * the user is NOT — far-end power exceeds near-end power and no double-talk)
+     * still carries the residual echo the finite-length filter could not remove.
+     * When enabled, the residual on those frames is scaled toward zero. It is
+     * **default-off** and never engages during double-talk or near-end-dominant
+     * frames, so it can never attenuate the user's voice. Pass `true` for the
+     * default gain or `{ gain }` to tune.
+     */
+    residualSuppression?: boolean | ResidualSuppressionOptions;
+}
+export interface ResidualSuppressionOptions {
+    /**
+     * Gain (0,1] applied to the residual on echo-only frames. Lower = stronger
+     * suppression. Default 0.15 (~−16 dB) — aggressive enough to flatten residual
+     * echo while leaving headroom for the gate's hysteresis.
+     */
+    gain?: number;
+}
+export declare class NlmsEchoCanceller {
+    private readonly w;
+    private readonly x;
+    private readonly taps;
+    private readonly mu;
+    private readonly eps;
+    private readonly delay;
+    private readonly dtdRatio;
+    /** Residual-suppressor gain, or null when the suppressor is disabled (default). */
+    private readonly resGain;
+    /** Pending far-end samples not yet aligned to a near-end sample (delay line). */
+    private readonly delayLine;
+    private xEnergy;
+    private peakXEnergy;
+    private pNear;
+    private pFar;
+    private hangover;
+    private lastEchoPow;
+    /** Stay frozen ~30 ms after the last double-talk trigger so the filter is not
+     * corrupted by the bursty onset/offset of the near-end talker. */
+    private static readonly HANGOVER_SAMPLES;
+    /** Per-sample decay of the far-end energy envelope (~1 s time constant) so a
+     * short TTS pause keeps the far-end-active gate closed through the gap. */
+    private static readonly PEAK_DECAY;
+    /** Far-end is "active" only when the instantaneous ‖x‖² is within this
+     * fraction (−20 dB) of the recent envelope. Below it there is no echo to
+     * learn, so adaptation freezes (see process()). */
+    private static readonly FAR_ACTIVITY_FRAC;
+    /** NLMS regularization as a fraction of the far-end envelope. Keeps the step
+     * bounded when ‖x‖² momentarily underflows, so a quiet far-end passage can't
+     * make the normaliser collapse to the absolute `eps` and blow the filter up. */
+    private static readonly REG_FRAC;
+    private lastResidualPow;
+    constructor(opts?: NlmsEchoCancellerOptions);
+    /**
+     * Cancel echo from one block of mic audio.
+     *
+     * @param nearEnd raw mic block (local speech + echo), Float32 [-1, 1]
+     * @param farEnd  agent playback reference for the same time window. Pass an
+     *                empty/zero array when the agent is NOT speaking — the filter
+     *                then passes the mic through unchanged (output ≈ input).
+     * @returns echo-cancelled near-end block (same length as `nearEnd`).
+     */
+    process(nearEnd: Float32Array, farEnd: Float32Array): Float32Array;
+    /**
+     * Advance cheap detector/reference state while the far-end is silent without
+     * running the FIR echo-estimation loop. Learned filter weights are preserved,
+     * but stale playback samples are removed from the delay line/ring so the next
+     * non-empty reference frame cannot subtract an echo estimate from a previous
+     * utterance.
+     */
+    observeFarEndSilence(nearEnd: Float32Array): void;
+    /** Echo-return-loss-enhancement (dB) over the last processed block. Higher is
+     * better; >10 dB is a meaningful cancellation. Returns 0 when there is no
+     * modeled echo (agent silent) so a passthrough block reads as "no gain". */
+    get lastErleDb(): number;
+    /** Reset adaptation (e.g. when the playback path changes). */
+    reset(): void;
+    /** Shift a new far-end sample into the ring buffer, maintaining ‖x‖²
+     * incrementally (drop the oldest sample's energy, add the newest). */
+    private pushRef;
+}
+//# sourceMappingURL=nlms-echo-canceller.d.ts.map

package/dist/services/voice/nlms-echo-canceller.d.ts.map

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package/dist/services/voice/optimistic-policy.d.ts.map

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