daemora 1.0.0-alpha.0

package/dist/voice/ElevenLabsSTT.js

@@ -0,0 +1,98 @@
+/**
+ * ElevenLabsSTT — ElevenLabs "Scribe" speech-to-text via REST.
+ *
+ * The @livekit/agents-plugin-elevenlabs package only ships TTS, so we
+ * wrap Scribe ourselves. Scribe is a batch endpoint (no streaming),
+ * which means we declare `streaming: false` and rely on LiveKit's
+ * built-in StreamAdapter to segment incoming audio with VAD and call
+ * `_recognize` per utterance.
+ *
+ * Docs: https://elevenlabs.io/docs/api-reference/speech-to-text
+ *
+ * Supported languages: 99+ (model auto-detects; can be forced via
+ * `language`). Model id is `scribe_v1` (and faster `scribe_v1_experimental`
+ * variants — configurable per request).
+ */
+import { stt, mergeFrames } from "@livekit/agents";
+export class ElevenLabsSTT extends stt.STT {
+    label = "elevenlabs.STT";
+    opts;
+    constructor(opts) {
+        super({ streaming: false, interimResults: false });
+        this.opts = {
+            model: "scribe_v1",
+            baseURL: "https://api.elevenlabs.io/v1",
+            ...opts,
+        };
+    }
+    get model() {
+        return this.opts.model;
+    }
+    get provider() {
+        return "elevenlabs";
+    }
+    async _recognize(buffer, abortSignal) {
+        const frame = mergeFrames(buffer);
+        const wav = createWavBuffer(frame);
+        const form = new FormData();
+        form.set("model_id", this.opts.model);
+        form.set("file", new File([new Uint8Array(wav)], "audio.wav", { type: "audio/wav" }));
+        if (this.opts.language)
+            form.set("language_code", this.opts.language);
+        if (this.opts.timestamps)
+            form.set("timestamps_granularity", "word");
+        if (this.opts.diarize)
+            form.set("diarize", "true");
+        const resp = await fetch(`${this.opts.baseURL}/speech-to-text`, {
+            method: "POST",
+            headers: { "xi-api-key": this.opts.apiKey },
+            body: form,
+            ...(abortSignal ? { signal: abortSignal } : {}),
+        });
+        if (!resp.ok) {
+            const body = await resp.text();
+            throw new Error(`ElevenLabs STT ${resp.status}: ${body.slice(0, 200)}`);
+        }
+        const data = (await resp.json());
+        return {
+            type: stt.SpeechEventType.FINAL_TRANSCRIPT,
+            alternatives: [
+                {
+                    text: data.text ?? "",
+                    language: (data.language_code ?? this.opts.language ?? ""),
+                    startTime: 0,
+                    endTime: 0,
+                    confidence: 0,
+                },
+            ],
+        };
+    }
+    stream() {
+        throw new Error("ElevenLabs STT does not support streaming — use StreamAdapter");
+    }
+}
+/**
+ * Pack a raw PCM16 mono `AudioFrame` into a minimal WAV buffer. Must
+ * match the layout ElevenLabs expects when you upload `audio.wav`.
+ */
+function createWavBuffer(frame) {
+    const bitsPerSample = 16;
+    const byteRate = (frame.sampleRate * frame.channels * bitsPerSample) / 8;
+    const blockAlign = (frame.channels * bitsPerSample) / 8;
+    const header = Buffer.alloc(44);
+    header.write("RIFF", 0);
+    header.writeUInt32LE(36 + frame.data.byteLength, 4);
+    header.write("WAVE", 8);
+    header.write("fmt ", 12);
+    header.writeUInt32LE(16, 16);
+    header.writeUInt16LE(1, 20);
+    header.writeUInt16LE(frame.channels, 22);
+    header.writeUInt32LE(frame.sampleRate, 24);
+    header.writeUInt32LE(byteRate, 28);
+    header.writeUInt16LE(blockAlign, 32);
+    header.writeUInt16LE(16, 34);
+    header.write("data", 36);
+    header.writeUInt32LE(frame.data.byteLength, 40);
+    return Buffer.concat([header, Buffer.from(frame.data.buffer)]);
+}
+//# sourceMappingURL=ElevenLabsSTT.js.map

package/dist/voice/ElevenLabsSTT.js.map

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package/dist/voice/GroqTTS.d.ts

@@ -0,0 +1,38 @@
+/**
+ * GroqTTS — a native Groq TTS plugin for the LiveKit voice pipeline.
+ *
+ * Background: the @livekit/agents-plugin-openai TTS class hardcodes
+ * `response_format: "pcm"` when it calls the upstream API. That works
+ * for OpenAI's real TTS endpoint, but Groq's OpenAI-compatible
+ * `/v1/audio/speech` endpoint rejects it — it only accepts `wav`. So
+ * every Groq TTS call via the openai plugin fails with a 400:
+ *
+ *   "response_format must be one of [wav]"
+ *
+ * This class talks to Groq directly with `response_format=wav`, strips
+ * the 44-byte WAV header, then feeds the raw PCM16 LE payload into
+ * `AudioByteStream` the same way the openai plugin would. Output sample
+ * rate is reported to LiveKit at construction time and must match the
+ * actual WAV rate; if Groq ever returns something else we fall back to
+ * the header-reported rate.
+ */
+import { type APIConnectOptions, tts } from "@livekit/agents";
+export declare function defaultVoiceForGroqModel(model: string): string;
+export interface GroqTTSOpts {
+    readonly apiKey: string;
+    readonly model: string;
+    readonly voice: string;
+    readonly baseURL?: string;
+    /** Override the inferred sample rate. Defaults look up the model. */
+    readonly sampleRate?: number;
+}
+export declare class GroqTTS extends tts.TTS {
+    label: string;
+    private readonly opts;
+    constructor(opts: GroqTTSOpts);
+    get model(): string;
+    get provider(): string;
+    synthesize(text: string, connOptions?: APIConnectOptions, abortSignal?: AbortSignal): tts.ChunkedStream;
+    stream(): tts.SynthesizeStream;
+}
+//# sourceMappingURL=GroqTTS.d.ts.map

package/dist/voice/GroqTTS.d.ts.map

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package/dist/voice/GroqTTS.js

@@ -0,0 +1,208 @@
+/**
+ * GroqTTS — a native Groq TTS plugin for the LiveKit voice pipeline.
+ *
+ * Background: the @livekit/agents-plugin-openai TTS class hardcodes
+ * `response_format: "pcm"` when it calls the upstream API. That works
+ * for OpenAI's real TTS endpoint, but Groq's OpenAI-compatible
+ * `/v1/audio/speech` endpoint rejects it — it only accepts `wav`. So
+ * every Groq TTS call via the openai plugin fails with a 400:
+ *
+ *   "response_format must be one of [wav]"
+ *
+ * This class talks to Groq directly with `response_format=wav`, strips
+ * the 44-byte WAV header, then feeds the raw PCM16 LE payload into
+ * `AudioByteStream` the same way the openai plugin would. Output sample
+ * rate is reported to LiveKit at construction time and must match the
+ * actual WAV rate; if Groq ever returns something else we fall back to
+ * the header-reported rate.
+ */
+import { AudioByteStream, tts } from "@livekit/agents";
+/**
+ * Default per-model sample rates. Groq's `/v1/audio/speech` returns
+ * WAV at different rates depending on the model — PlayAI is 48 kHz,
+ * Orpheus is 24 kHz. If LiveKit is told the wrong rate, playback runs
+ * at the wrong speed (24 kHz played as 48 kHz = double-speed chipmunk).
+ */
+const SAMPLE_RATE_BY_MODEL = {
+    "playai-tts": 48_000,
+    "playai-tts-english": 48_000,
+    "playai-tts-arabic": 48_000,
+    "canopylabs/orpheus-v1-english": 24_000,
+    "orpheus-v1-english": 24_000,
+};
+const DEFAULT_SAMPLE_RATE = 24_000;
+const CHANNELS = 1;
+const WAV_HEADER_BYTES = 44;
+/**
+ * Default voice per Groq TTS model. Groq rejects requests without a voice
+ * with `400 voice is required`, and the right voice depends on the model
+ * family (PlayAI uses "Fritz-PlayAI" / "Celeste-PlayAI" / etc., Orpheus
+ * uses "tara" / "leah" / etc.). Picking the wrong family also 400s, so a
+ * single global default isn't safe — match by model.
+ */
+const DEFAULT_VOICE_BY_MODEL = {
+    "playai-tts": "Fritz-PlayAI",
+    "playai-tts-english": "Fritz-PlayAI",
+    "playai-tts-arabic": "Ahmad-PlayAI",
+    // Groq Orpheus accepts only [autumn, diana, hannah, austin, daniel, troy]
+    // — older "tara" / "leah" names from upstream Orpheus are rejected
+    // (400 invalid_request_error). Picking `troy` as a neutral male voice;
+    // override per-deploy via TTS_VOICE setting.
+    "canopylabs/orpheus-v1-english": "troy",
+    "orpheus-v1-english": "troy",
+};
+const FALLBACK_VOICE = "Fritz-PlayAI";
+/** Voices Groq's Orpheus model currently accepts. Used to validate TTS_VOICE. */
+const ORPHEUS_VOICES = new Set(["autumn", "diana", "hannah", "austin", "daniel", "troy"]);
+export function defaultVoiceForGroqModel(model) {
+    return DEFAULT_VOICE_BY_MODEL[model] ?? FALLBACK_VOICE;
+}
+export class GroqTTS extends tts.TTS {
+    label = "groq.TTS";
+    opts;
+    constructor(opts) {
+        const rate = opts.sampleRate ?? SAMPLE_RATE_BY_MODEL[opts.model] ?? DEFAULT_SAMPLE_RATE;
+        super(rate, CHANNELS, { streaming: false });
+        // Hard-fall back to a model-appropriate voice if none was passed —
+        // Groq returns `400 voice is required` and PlayAI/Orpheus voices are
+        // not interchangeable, so we have to pick one matched to the model.
+        // Also validate Orpheus voices: the upstream "tara"/"leah" names are
+        // rejected by Groq, so silently substitute the default if the user
+        // supplied one that's not in the accepted list.
+        let voice = (opts.voice && opts.voice.trim() !== "")
+            ? opts.voice.toLowerCase()
+            : defaultVoiceForGroqModel(opts.model);
+        if (/orpheus/.test(opts.model) && !ORPHEUS_VOICES.has(voice)) {
+            voice = defaultVoiceForGroqModel(opts.model);
+        }
+        this.opts = {
+            baseURL: "https://api.groq.com/openai/v1",
+            ...opts,
+            voice,
+            sampleRate: rate,
+        };
+    }
+    get model() {
+        return this.opts.model;
+    }
+    get provider() {
+        return "groq";
+    }
+    synthesize(text, connOptions, abortSignal) {
+        const audioPromise = fetchWavAsPcm(this.opts, text, abortSignal);
+        return new GroqChunkedStream(this, text, audioPromise, this.opts.sampleRate, connOptions, abortSignal);
+    }
+    stream() {
+        throw new Error("GroqTTS does not support streaming synthesis");
+    }
+}
+class GroqChunkedStream extends tts.ChunkedStream {
+    label = "groq.ChunkedStream";
+    pcmPromise;
+    sampleRate;
+    constructor(ttsRef, text, pcmPromise, sampleRate, connOptions, abortSignal) {
+        super(text, ttsRef, connOptions, abortSignal);
+        this.pcmPromise = pcmPromise;
+        this.sampleRate = sampleRate;
+    }
+    async run() {
+        const requestId = randomId();
+        const stream = new AudioByteStream(this.sampleRate, CHANNELS);
+        const queue = this.queue;
+        // Pre-emit ~120 ms of silence BEFORE awaiting the WAV. Groq's
+        // `/audio/speech` is non-streaming and Orpheus often takes 6-12 s
+        // to return; LiveKit's voice pipeline aborts the speak if no
+        // first frame arrives within 10 s, which is why daemora went
+        // silent for non-trivial responses. PCM16 silence = zeros at the
+        // configured sample rate; LiveKit emits it inaudibly while we
+        // wait for real audio.
+        const silenceMs = 120;
+        const silenceBytes = new ArrayBuffer(Math.floor((this.sampleRate * silenceMs) / 1000) * 2);
+        for (const f of stream.write(silenceBytes)) {
+            queue.put({ requestId, segmentId: "0", frame: f, final: false });
+        }
+        try {
+            const pcm = await this.pcmPromise;
+            const frames = stream.write(pcm);
+            let last;
+            const emit = (segmentId, final) => {
+                if (!last)
+                    return;
+                queue.put({ requestId, segmentId, frame: last, final });
+                last = undefined;
+            };
+            for (const frame of frames) {
+                emit("0", false);
+                last = frame;
+            }
+            emit("0", true);
+        }
+        catch (err) {
+            if (err instanceof Error && err.name === "AbortError")
+                return;
+            throw err;
+        }
+        finally {
+            queue.close();
+        }
+    }
+}
+async function fetchWavAsPcm(opts, text, abortSignal) {
+    // Pre-flight checks so misconfigurations turn into clear messages
+    // rather than an opaque "Groq TTS 400" trace from the API.
+    if (!opts.apiKey) {
+        throw new Error("Groq TTS: GROQ_API_KEY is not set. Add it in Settings → Secrets.");
+    }
+    if (!opts.voice) {
+        throw new Error(`Groq TTS: no voice configured for model "${opts.model}". ` +
+            `Set TTS_VOICE in Settings → Voice (e.g. "Fritz-PlayAI" for PlayAI models, "tara" for Orpheus).`);
+    }
+    if (!opts.model) {
+        throw new Error("Groq TTS: no model configured. Set TTS_MODEL in Settings → Voice.");
+    }
+    const resp = await fetch(`${opts.baseURL}/audio/speech`, {
+        method: "POST",
+        headers: {
+            Authorization: `Bearer ${opts.apiKey}`,
+            "Content-Type": "application/json",
+        },
+        body: JSON.stringify({
+            input: text,
+            model: opts.model,
+            voice: opts.voice,
+            response_format: "wav",
+        }),
+        ...(abortSignal ? { signal: abortSignal } : {}),
+    });
+    if (!resp.ok) {
+        const body = await resp.text();
+        // Surface the Groq error message verbatim — it's already informative
+        // ("voice is required", "model not found", etc.) and the worker logs
+        // it through the normal error pipeline.
+        throw new Error(`Groq TTS ${resp.status}: ${body.slice(0, 200)}`);
+    }
+    const buf = await resp.arrayBuffer();
+    // Strip WAV header. PCM16 LE starts at byte 44 for a standard 16-bit WAV.
+    // If the header is a different length (e.g. contains LIST chunks), search
+    // for the "data" marker instead.
+    return stripWavHeader(buf);
+}
+function stripWavHeader(buf) {
+    const view = new DataView(buf);
+    // Look for "data" (0x64 0x61 0x74 0x61) chunk marker within the first
+    // 128 bytes; its length is the 4 bytes following.
+    for (let i = 12; i < Math.min(buf.byteLength - 8, 256); i++) {
+        if (view.getUint8(i) === 0x64 &&
+            view.getUint8(i + 1) === 0x61 &&
+            view.getUint8(i + 2) === 0x74 &&
+            view.getUint8(i + 3) === 0x61) {
+            return buf.slice(i + 8);
+        }
+    }
+    // Fall back to the standard header length.
+    return buf.slice(WAV_HEADER_BYTES);
+}
+function randomId() {
+    return Math.random().toString(36).slice(2);
+}
+//# sourceMappingURL=GroqTTS.js.map

package/dist/voice/GroqTTS.js.map

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

@@ -0,0 +1,30 @@
+/**
+ * LiveKitServer — manages a local livekit-server process.
+ *
+ * In dev mode, Daemora auto-starts livekit-server --dev on port 7880
+ * as a loopback SFU. The browser and voice agent both connect to it.
+ * In production, this points to a cloud LiveKit instance instead.
+ *
+ * Lifecycle:
+ *   1. Check if 7880 is already occupied → reuse
+ *   2. Find livekit-server binary on PATH
+ *   3. Spawn with --dev flag (generates devkey/secret automatically)
+ *   4. Wait for port to bind (8s timeout)
+ */
+export declare class LiveKitServer {
+    private child;
+    private readonly port;
+    constructor(port?: number);
+    get url(): string;
+    get isManaged(): boolean;
+    /**
+     * Ensure livekit-server is running. If already listening on the port,
+     * reuse it. Otherwise spawn a new --dev instance.
+     */
+    ensureRunning(): Promise<void>;
+    stop(): void;
+    private findBinary;
+    private isPortOpen;
+    private waitForPort;
+}
+//# sourceMappingURL=LiveKitServer.d.ts.map

package/dist/voice/LiveKitServer.d.ts.map

@@ -0,0 +1 @@
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package/dist/voice/LiveKitServer.js

@@ -0,0 +1,119 @@
+/**
+ * LiveKitServer — manages a local livekit-server process.
+ *
+ * In dev mode, Daemora auto-starts livekit-server --dev on port 7880
+ * as a loopback SFU. The browser and voice agent both connect to it.
+ * In production, this points to a cloud LiveKit instance instead.
+ *
+ * Lifecycle:
+ *   1. Check if 7880 is already occupied → reuse
+ *   2. Find livekit-server binary on PATH
+ *   3. Spawn with --dev flag (generates devkey/secret automatically)
+ *   4. Wait for port to bind (8s timeout)
+ */
+import { spawn, execSync } from "node:child_process";
+import { createConnection } from "node:net";
+import { createLogger } from "../util/logger.js";
+const log = createLogger("voice.livekit");
+const DEFAULT_PORT = 7880;
+export class LiveKitServer {
+    child = null;
+    port;
+    constructor(port) {
+        this.port = port ?? DEFAULT_PORT;
+    }
+    get url() {
+        return `ws://127.0.0.1:${this.port}`;
+    }
+    get isManaged() {
+        return this.child !== null && !this.child.killed;
+    }
+    /**
+     * Ensure livekit-server is running. If already listening on the port,
+     * reuse it. Otherwise spawn a new --dev instance.
+     */
+    async ensureRunning() {
+        // Already listening? Reuse.
+        if (await this.isPortOpen(200)) {
+            log.info({ port: this.port }, "livekit-server already running");
+            return;
+        }
+        // Find the binary
+        const binary = this.findBinary();
+        if (!binary) {
+            log.warn("livekit-server not found on PATH — voice will not work");
+            log.warn("Install: brew install livekit (macOS) or curl -sSL https://get.livekit.io | bash");
+            return;
+        }
+        log.info({ binary, port: this.port }, "spawning livekit-server --dev");
+        this.child = spawn(binary, ["--dev", "--bind", "127.0.0.1", "--port", String(this.port)], {
+            stdio: ["ignore", "pipe", "pipe"],
+            detached: false,
+        });
+        this.child.stdout?.on("data", (chunk) => {
+            const s = chunk.toString().trim();
+            if (s && /(starting|listening|error|ready)/i.test(s)) {
+                log.info(`[livekit] ${s.slice(0, 200)}`);
+            }
+        });
+        this.child.stderr?.on("data", (chunk) => {
+            const s = chunk.toString().trim();
+            if (s && /(starting|listening|error|ready)/i.test(s)) {
+                log.info(`[livekit] ${s.slice(0, 200)}`);
+            }
+        });
+        this.child.on("exit", (code, signal) => {
+            log.info({ code, signal }, "livekit-server exited");
+            this.child = null;
+        });
+        // Wait for port to bind
+        const up = await this.waitForPort(8_000);
+        if (!up) {
+            this.stop();
+            throw new Error(`livekit-server did not bind 127.0.0.1:${this.port} within 8s`);
+        }
+        log.info({ port: this.port }, "livekit-server ready");
+    }
+    stop() {
+        if (!this.child || this.child.killed)
+            return;
+        this.child.kill("SIGTERM");
+        setTimeout(() => {
+            if (this.child && !this.child.killed)
+                this.child.kill("SIGKILL");
+        }, 3000).unref();
+    }
+    findBinary() {
+        const candidates = [
+            "livekit-server",
+            "/opt/homebrew/bin/livekit-server",
+            "/usr/local/bin/livekit-server",
+        ];
+        for (const c of candidates) {
+            try {
+                execSync(`command -v ${c}`, { stdio: "ignore" });
+                return c;
+            }
+            catch { /* not found */ }
+        }
+        return null;
+    }
+    isPortOpen(timeoutMs) {
+        return new Promise((resolve) => {
+            const socket = createConnection({ host: "127.0.0.1", port: this.port });
+            const timer = setTimeout(() => { socket.destroy(); resolve(false); }, timeoutMs);
+            socket.once("connect", () => { clearTimeout(timer); socket.destroy(); resolve(true); });
+            socket.once("error", () => { clearTimeout(timer); socket.destroy(); resolve(false); });
+        });
+    }
+    async waitForPort(timeoutMs) {
+        const deadline = Date.now() + timeoutMs;
+        while (Date.now() < deadline) {
+            if (await this.isPortOpen(500))
+                return true;
+            await new Promise((r) => setTimeout(r, 200));
+        }
+        return false;
+    }
+}
+//# sourceMappingURL=LiveKitServer.js.map

package/dist/voice/LiveKitServer.js.map

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

@@ -0,0 +1,25 @@
+/**
+ * LiveKit token generation using the official livekit-server-sdk.
+ *
+ * Generates access tokens for:
+ *   - Browser clients (join room, publish mic, subscribe to agent audio)
+ *   - Voice agent workers (join room, publish audio, subscribe to user)
+ */
+export interface LiveKitTokenOpts {
+    identity: string;
+    room: string;
+    apiKey: string;
+    apiSecret: string;
+    ttlSeconds?: number;
+}
+export declare function generateLiveKitToken(opts: LiveKitTokenOpts): Promise<string>;
+/**
+ * Generate a token for the voice agent worker.
+ * Has room admin permissions for managing tracks.
+ */
+export declare function generateAgentToken(opts: {
+    apiKey: string;
+    apiSecret: string;
+    room: string;
+}): Promise<string>;
+//# sourceMappingURL=LiveKitToken.d.ts.map

package/dist/voice/LiveKitToken.d.ts.map

@@ -0,0 +1 @@
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package/dist/voice/LiveKitToken.js

@@ -0,0 +1,44 @@
+/**
+ * LiveKit token generation using the official livekit-server-sdk.
+ *
+ * Generates access tokens for:
+ *   - Browser clients (join room, publish mic, subscribe to agent audio)
+ *   - Voice agent workers (join room, publish audio, subscribe to user)
+ */
+import { AccessToken } from "livekit-server-sdk";
+export async function generateLiveKitToken(opts) {
+    const at = new AccessToken(opts.apiKey, opts.apiSecret, {
+        identity: opts.identity,
+        ttl: opts.ttlSeconds ?? 6 * 60 * 60,
+    });
+    at.addGrant({
+        room: opts.room,
+        roomJoin: true,
+        canPublish: true,
+        canSubscribe: true,
+        canPublishData: true,
+        canUpdateOwnMetadata: true,
+    });
+    return await at.toJwt();
+}
+/**
+ * Generate a token for the voice agent worker.
+ * Has room admin permissions for managing tracks.
+ */
+export async function generateAgentToken(opts) {
+    const at = new AccessToken(opts.apiKey, opts.apiSecret, {
+        identity: "daemora-agent",
+        ttl: 24 * 60 * 60,
+    });
+    at.addGrant({
+        room: opts.room,
+        roomJoin: true,
+        roomAdmin: true,
+        canPublish: true,
+        canSubscribe: true,
+        canPublishData: true,
+        canUpdateOwnMetadata: true,
+    });
+    return await at.toJwt();
+}
+//# sourceMappingURL=LiveKitToken.js.map

package/dist/voice/LiveKitToken.js.map

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

@@ -0,0 +1,39 @@
+/**
+ * SidecarSupervisor — manages the Python voice + desktop sidecar process.
+ *
+ * The sidecar runs as a child process with:
+ *   - A random per-spawn DAEMORA_SIDECAR_TOKEN for auth
+ *   - All vault secrets inherited via process.env
+ *   - LiveKit connection env (URL, API key/secret)
+ *
+ * Security: the token is generated fresh on every spawn. Only Daemora
+ * and the one sidecar process know it. All HTTP calls between them
+ * require X-Daemora-Token header.
+ */
+export interface SidecarStatus {
+    running: boolean;
+    pid: number | null;
+    url: string;
+    lastError: string | null;
+}
+export declare class SidecarSupervisor {
+    private readonly sidecarDir;
+    private readonly livekitUrl;
+    private readonly livekitApiKey;
+    private readonly livekitApiSecret;
+    private child;
+    private token;
+    private lastError;
+    private starting;
+    constructor(sidecarDir: string, livekitUrl: string, livekitApiKey: string, livekitApiSecret: string);
+    get url(): string;
+    get sidecarToken(): string | null;
+    status(): SidecarStatus;
+    start(): Promise<SidecarStatus>;
+    stop(): Promise<void>;
+    /** Proxy a request to the sidecar with auth token. */
+    fetch(path: string, init?: RequestInit): Promise<Response>;
+    private waitForHealth;
+    private killStale;
+}
+//# sourceMappingURL=SidecarSupervisor.d.ts.map

package/dist/voice/SidecarSupervisor.d.ts.map

@@ -0,0 +1 @@
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