@craftedxp/voice-js 0.4.0 → 0.4.1

package/dist/browser.js

@@ -875,9 +875,17 @@ var BrowserVoiceClient = class {
 
 // src/webrtc/createWebRtcCall.ts
 async function createWebRtcCall(opts) {
+  validateClientToolMap(opts.clientTools)
   const proto = createProtocolState()
   let muted = false
   let ended = false
+  const tools = opts.clientTools ?? {}
+  const sendControl = (frame) => {
+    if (dc?.readyState !== 'open') return
+    try {
+      dc.send(JSON.stringify(frame))
+    } catch {}
+  }
   const fireState = (next) => {
     if (proto.state === next) return
     proto.state = next
@@ -892,8 +900,14 @@ async function createWebRtcCall(opts) {
       onAgentTurnStart: () => opts.onAgentTurnStart?.(),
       onAgentTurnEnd: () => {},
       onCallEnd: () => teardown(),
-      onConnected: () => {},
-      onClientToolCall: () => {},
+      onConnected: () => {
+        if (Object.keys(tools).length > 0) {
+          sendControl(buildRegisterFrame(tools))
+        }
+      },
+      onClientToolCall: (frame) => {
+        dispatchClientToolCall(sendControl, tools, frame)
+      },
     })
   }
   fireState('connecting')
@@ -932,6 +946,11 @@ async function createWebRtcCall(opts) {
   dc.onerror = () => {
     opts.onError?.({ code: 'socket_error', message: 'control channel error' })
   }
+  dc.onopen = () => {
+    if (Object.keys(tools).length > 0) {
+      sendControl(buildRegisterFrame(tools))
+    }
+  }
   const gateway = opts.webrtcGatewayBase || ''
   const offerUrl = gateway
     ? `${gateway}/webrtc/offer?token=${encodeURIComponent(opts.token)}`
@@ -1073,6 +1092,7 @@ var BrowserVoiceFactory = class {
             : void 0,
           onInterrupt: options.onInterrupt,
           onAgentTurnStart: options.onAgentTurnStart,
+          clientTools: options.clientTools,
         })
       }
       const client = new BrowserVoiceClient({

package/dist/browser.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/browser.ts","../src/config.ts","../src/worklets/mic-downsampler.worklet.js","../src/AudioCapture.ts","../src/AudioPlayback.ts","../src/ReconnectingWebSocket.ts","../src/protocol.ts","../src/clientTools.ts","../src/ClientMarksBuffer.ts","../src/VoiceClient.ts","../src/webrtc/createWebRtcCall.ts"],"sourcesContent":["// Browser entry — `import { configureVoiceClient } from '@craftedxp/voice-js'`\n//\n// Pre-injects:\n//   - native `WebSocket` (globalThis.WebSocket)\n//   - browser audio adapter (AudioContext + AudioWorklet + getUserMedia)\n//\n// Electron renderer + browser tabs both pick this entry via the\n// `browser` condition in package.json `exports`.\n\nimport {\n  mergeStartCallContext,\n  normalizeConfig,\n  type Call,\n  type FetchTokenResult,\n  type StartCallOptions,\n  type VoiceClientConfig,\n  type VoiceClientFactory,\n} from './config'\nimport type { WebSocketFactory, WebSocketLike } from './ReconnectingWebSocket'\nimport { BrowserVoiceClient } from './VoiceClient'\nimport { createWebRtcCall } from './webrtc/createWebRtcCall'\n\n// Adapter from native browser WebSocket → our WebSocketLike contract.\n// One-line wrapper; the shapes already match.\nconst browserWsFactory: WebSocketFactory = (url) =>\n  new (globalThis as unknown as { WebSocket: new (u: string) => WebSocketLike }).WebSocket(url)\n\nclass BrowserVoiceFactory implements VoiceClientFactory {\n  readonly config: VoiceClientConfig\n\n  constructor(config: VoiceClientConfig) {\n    this.config = config\n  }\n\n  startCall = async (options: StartCallOptions): Promise<Call> => {\n    if (!options.agentId) {\n      throw new Error('startCall: agentId is required')\n    }\n\n    const { context, metadata } = mergeStartCallContext(this.config, options)\n    const fetchArgs = {\n      agentId: options.agentId,\n      userId: options.userId,\n      context,\n      metadata,\n    }\n\n    // Resolve the token + transport. Three input shapes:\n    //   1. options.token (test escape hatch) — always WS, current behaviour\n    //   2. fetchToken returns a bare string — always WS (back-compat)\n    //   3. fetchToken returns FetchTokenResult — server-chosen transport\n    let resolved: FetchTokenResult\n    if (options.token) {\n      resolved = { token: options.token, transport: 'ws' }\n    } else {\n      const r = await this.config.fetchToken(fetchArgs)\n      if (!r) {\n        throw new Error('configureVoiceClient.fetchToken returned empty token')\n      }\n      resolved = typeof r === 'string' ? { token: r, transport: 'ws' } : r\n      if (!resolved.token) {\n        throw new Error('configureVoiceClient.fetchToken returned an object without `token`')\n      }\n    }\n\n    if (resolved.transport === 'webrtc') {\n      return createWebRtcCall({\n        agentId: options.agentId,\n        apiBase: this.config.apiBase,\n        token: resolved.token,\n        webrtcGatewayBase: resolved.webrtcGatewayBase,\n        onStateChange: options.onStateChange,\n        onTranscript: options.onTranscript,\n        onError: options.onError,\n        // Synthesise a minimal CallEndEvent. WebRTC doesn't carry an end reason\n        // from the server yet — use 'agent_ended' as placeholder. durationMs is\n        // tracked at 0 until the followup lands (see spec Followups section).\n        onEnd: options.onEnd\n          ? () => options.onEnd!({ reason: 'agent_ended', durationMs: 0 })\n          : undefined,\n        onInterrupt: options.onInterrupt,\n        onAgentTurnStart: options.onAgentTurnStart,\n      })\n    }\n\n    const client = new BrowserVoiceClient({\n      config: this.config,\n      // Carry merged context/metadata through to startCall so server can\n      // see what the SDK saw.\n      options: { ...options, context, metadata },\n      token: resolved.token,\n      wsFactory: browserWsFactory,\n    })\n    await client.start()\n    return client\n  }\n}\n\n/**\n * One-time SDK setup. Returns a factory you call `startCall` on for\n * every voice call.\n *\n * Example:\n *   const voice = configureVoiceClient({\n *     apiBase: 'https://api.your-server.com',\n *     fetchToken: async ({ agentId }) => {\n *       const r = await fetch('/api/voice-token', {\n *         method: 'POST',\n *         body: JSON.stringify({ agentId }),\n *       })\n *       return (await r.json()).token\n *     },\n *   })\n *\n *   // Per call (typically inside a click handler):\n *   const call = await voice.startCall({\n *     agentId: 'agt_xxx',\n *     onTranscript: (entries) => render(entries),\n *     onEnd: ({ reason }) => log(reason),\n *   })\n *   call.mute()\n *   call.end()\n */\nexport function configureVoiceClient(config: VoiceClientConfig): VoiceClientFactory {\n  return new BrowserVoiceFactory(normalizeConfig(config))\n}\n\n// ---------------------------------------------------------------------------\n// Public re-exports — types + advanced primitives.\n// ---------------------------------------------------------------------------\n\nexport type {\n  Call,\n  ClientTool,\n  ClientToolMap,\n  FetchToken,\n  FetchTokenArgs,\n  FetchTokenResult,\n  StartCallOptions,\n  VoiceClientConfig,\n  VoiceClientFactory,\n} from './config'\n\nexport type {\n  CallEndEvent,\n  CallEndReason,\n  CallError,\n  CallErrorCode,\n  CallState,\n  TranscriptEntry,\n  VolumeEvent,\n} from './protocol'\n\n// Low-level primitives — exposed for advanced consumers who want to\n// drive just the mic, just the playback, or build a custom transport on\n// top of the protocol decoder. Most users should stick to\n// `configureVoiceClient().startCall()`.\nexport { createAudioCapture } from './AudioCapture'\nexport type { CaptureController, CaptureOptions, OnChunk, OnError, OnVolume } from './AudioCapture'\n\nexport { createAudioPlayback } from './AudioPlayback'\nexport type { PlaybackController, PlaybackOptions, OnAgentSpeakingChange } from './AudioPlayback'\n\nexport { createReconnectingWebSocket } from './ReconnectingWebSocket'\nexport type {\n  ReconnectingWebSocket,\n  RWSEvent,\n  RWSOptions,\n  WebSocketFactory,\n  WebSocketLike,\n} from './ReconnectingWebSocket'\n\nexport { handleServerMessage, createProtocolState, buildWsUrl } from './protocol'\nexport type { ProtocolState, ProtocolCallbacks, ServerMessage } from './protocol'\n","// Public configuration surface.\n//\n// `configureVoiceClient` returns a small factory that knows how to mint\n// `ct_` tokens via your callback. Per-call you call `factory.startCall`.\n//\n// Parity with @craftedxp/voice-rn:\n//   voice-rn calls `configureVoiceClient` once at app startup as a\n//   side-effect (singleton), because in RN there is exactly one host\n//   app and one running `<AgentCall>` at a time. In JS environments\n//   the same process can drive multiple clients (multi-tenant\n//   dashboards, terminal multiplexers, electron apps with several\n//   panels), so the JS SDK returns the factory rather than mutating a\n//   module-level singleton. Same option names, same callback shape.\n//\n// Auth model:\n//   - `apiKey` is INTENTIONALLY ABSENT from the public surface.\n//     Pre-0.2 had it; baking an `sk_` into a JS bundle ships your\n//     server-grade credentials to every client. The right pattern is\n//     `fetchToken` — your code asks YOUR backend for a short-lived\n//     `ct_` and your backend uses its `sk_` (via @craftedxp/sdk-node)\n//     to mint it.\n//   - For tests + local prototypes a bare `token` may be passed to\n//     `startCall` directly, bypassing `fetchToken`. Don't ship that to\n//     production — `ct_` lifetimes are short and you want the SDK to\n//     re-mint on expiry.\n\nimport type { CallEndEvent, CallError, CallState, TranscriptEntry, VolumeEvent } from './protocol'\nimport type { ClientToolMap } from './clientTools'\n\n// ---------------------------------------------------------------------------\n// fetchToken contract — matches voice-rn FetchTokenArgs verbatim.\n// ---------------------------------------------------------------------------\n\nexport interface FetchTokenArgs {\n  /** The agent the SDK is about to call. */\n  agentId: string\n  /**\n   * Optional consumer-side user identifier. Round-tripped to the server\n   * as `contactId` for Phase 11 contact memory. The SDK does not\n   * inspect this; your backend uses it to scope the token mint.\n   */\n  userId?: string\n  /**\n   * Per-call structured context lowered into the agent's effective\n   * system prompt server-side at session open. Opaque to the SDK.\n   */\n  context?: Record<string, unknown>\n  /**\n   * String key/value pairs round-tripped on the `call.ended` webhook.\n   * Capped at 1 KB total server-side. NOT lowered into the system prompt.\n   */\n  metadata?: Record<string, string>\n}\n\n/**\n * What `fetchToken` may return. The rich object form lets the server\n * choose the transport per call. Returning a bare string is backwards-\n * compatible — the SDK treats it as `{ token, transport: 'ws' }`.\n */\nexport interface FetchTokenResult {\n  /** Raw `ct_` to feed into the WS open / WebRTC offer. */\n  token: string\n  /** Server-selected transport. Default `'ws'` if absent. */\n  transport?: 'ws' | 'webrtc'\n  /** Required when `transport === 'webrtc'` AND the server uses a\n   *  separate signaling gateway. When omitted on a webrtc result, the\n   *  SDK falls back to the API base's Phase-1 routes (local dev). */\n  webrtcGatewayBase?: string\n}\n\nexport type FetchToken = (args: FetchTokenArgs) => Promise<string | FetchTokenResult>\n\n// ---------------------------------------------------------------------------\n// Factory configuration\n// ---------------------------------------------------------------------------\n\nexport interface VoiceClientConfig {\n  /**\n   * Full HTTPS URL of the Voissia server. The WebSocket scheme is\n   * derived: `https` → `wss`, `http` → `ws`. No trailing slash needed.\n   */\n  apiBase: string\n  /**\n   * Called by the SDK whenever it needs a fresh `ct_` token (initial\n   * connect; mid-call refresh on `token_expired`). Your implementation\n   * should hit YOUR backend, which holds the `sk_` API key and mints\n   * via `POST /v1/call-tokens` (or `client.callTokens.mint` from\n   * @craftedxp/sdk-node). Never embed `sk_` in JS code that ships to a\n   * client.\n   */\n  fetchToken: FetchToken\n  /**\n   * Optional metadata applied to EVERY startCall. Per-call `metadata`\n   * in `startCall` is merged on top (per-call wins on key conflicts).\n   * Useful for dashboard-wide tags like `{ surface: 'web', appVersion }`.\n   */\n  defaultMetadata?: Record<string, string>\n  /**\n   * Optional context applied to EVERY startCall. Per-call `context` in\n   * `startCall` is merged on top. Useful for cross-call invariants like\n   * the signed-in user's locale.\n   */\n  defaultContext?: Record<string, unknown>\n}\n\n// ---------------------------------------------------------------------------\n// Per-call options — startCall({ ... })\n// ---------------------------------------------------------------------------\n\nexport interface StartCallOptions {\n  /** The agent to call. */\n  agentId: string\n  /** Per-call user identifier. Round-tripped to fetchToken as `userId`. */\n  userId?: string\n  /**\n   * Per-call structured context. Merged on top of `defaultContext`\n   * configured at factory time.\n   */\n  context?: Record<string, unknown>\n  /**\n   * Per-call metadata. Merged on top of `defaultMetadata` configured\n   * at factory time.\n   */\n  metadata?: Record<string, string>\n  /**\n   * When false, the SDK + server stay full-duplex but barge-in is\n   * suppressed. Useful for alarm-style flows where the user shouldn't\n   * accidentally interrupt the script. Default true.\n   */\n  bargeIn?: boolean\n  /**\n   * Client-side tools the agent's LLM can call mid-conversation. Each\n   * tool's handler runs on the consumer's side; result is fed back to\n   * the LLM through the existing call WebSocket. Schema and handler\n   * colocate. Validated synchronously at startCall — bad input throws.\n   *\n   * See docs/integration-echocheck.md for the wire protocol and the\n   * server-side guarantees.\n   */\n  clientTools?: ClientToolMap\n  /**\n   * Test-only escape hatch — pass a pre-minted `ct_` directly and skip\n   * the `fetchToken` call. Don't use this in production code: tokens\n   * expire and the SDK can't re-mint without the callback.\n   */\n  token?: string\n\n  // Event callbacks — same shape as voice-rn UseVoiceCallOptions.\n  onStateChange?: (state: CallState) => void\n  onTranscript?: (entries: TranscriptEntry[]) => void\n  onError?: (err: CallError) => void\n  onEnd?: (end: CallEndEvent) => void\n  /** Volume-meter event for VU UIs. ~10 Hz cadence (browser bundle only). */\n  onVolume?: (vol: VolumeEvent) => void\n  /**\n   * Fires when the server signals barge-in (the user started talking\n   * mid-agent-turn). The browser bundle automatically flushes its\n   * built-in audio playback before this callback runs; the callback is\n   * fired regardless. Node / Electron consumers with custom playback\n   * should drain their audio queue here so the agent goes silent\n   * immediately.\n   */\n  onInterrupt?: () => void\n  /**\n   * Fires on `agent_turn_start` — the server has begun a new agent\n   * turn. The state-machine transition to `agent_speaking` happens at\n   * the same moment via `onStateChange`; use this when you want a\n   * precise turn anchor (e.g. \"agent has been speaking for N ms\" UIs)\n   * without diffing state.\n   */\n  onAgentTurnStart?: () => void\n}\n\n// ---------------------------------------------------------------------------\n// Call handle returned from startCall — matches voice-rn\n// VoiceCallController where it makes sense.\n// ---------------------------------------------------------------------------\n\nexport interface Call {\n  /** Current state. Snapshot — subscribe via onStateChange for live updates. */\n  readonly state: CallState\n  /** Full transcript so far. Snapshot — subscribe via onTranscript for live updates. */\n  readonly transcript: TranscriptEntry[]\n  /** True after `mute()` and before `unmute()`. */\n  readonly isMuted: boolean\n  /** End the call locally. Closes the WS, stops the mic, fires onEnd. Idempotent. */\n  end: () => void\n  /** Mute mic frames. Wire stays active so server endpointing doesn't false-positive. Idempotent. */\n  mute: () => void\n  /** Unmute mic frames. Idempotent. */\n  unmute: () => void\n}\n\n// ---------------------------------------------------------------------------\n// Factory contract — what configureVoiceClient returns. The actual\n// implementation differs between browser (audio-equipped) and node (raw\n// PCM) bundles; both satisfy this interface.\n// ---------------------------------------------------------------------------\n\nexport interface VoiceClientFactory {\n  /** Read back the resolved config (post trailing-slash normalisation). */\n  readonly config: VoiceClientConfig\n  /**\n   * Open a fresh call. Returns when the WS is open; rejects on\n   * pre-flight failure (missing config, fetchToken throw, etc). Mid-\n   * call failures arrive via the per-call `onError` callback — they\n   * don't reject this promise.\n   */\n  startCall: (options: StartCallOptions) => Promise<Call>\n}\n\n// ---------------------------------------------------------------------------\n// Pre-flight validation. Pulled out so both bundles share the exact\n// same \"missing field\" error messages.\n// ---------------------------------------------------------------------------\n\nexport function normalizeConfig(config: VoiceClientConfig): VoiceClientConfig {\n  if (!config) throw new Error('configureVoiceClient: config is required')\n  if ('apiKey' in (config as object)) {\n    throw new Error(\n      'configureVoiceClient: `apiKey` is no longer supported. Embedding sk_ in JS code ships server-grade credentials to every client. Pass `fetchToken: async ({ agentId }) => { /* call YOUR backend mint */ }` instead — see the @craftedxp/voice-js README for the migration recipe.',\n    )\n  }\n  if (!config.apiBase) {\n    throw new Error('configureVoiceClient: apiBase is required')\n  }\n  if (typeof config.fetchToken !== 'function') {\n    throw new Error('configureVoiceClient: fetchToken must be a function')\n  }\n  return {\n    ...config,\n    apiBase: config.apiBase.replace(/\\/+$/, ''),\n  }\n}\n\n// Merge factory-level defaults with per-call overrides. Per-call wins.\nexport function mergeStartCallContext(\n  factory: VoiceClientConfig,\n  call: StartCallOptions,\n): { context?: Record<string, unknown>; metadata?: Record<string, string> } {\n  const context =\n    factory.defaultContext || call.context\n      ? { ...(factory.defaultContext ?? {}), ...(call.context ?? {}) }\n      : undefined\n  const metadata =\n    factory.defaultMetadata || call.metadata\n      ? { ...(factory.defaultMetadata ?? {}), ...(call.metadata ?? {}) }\n      : undefined\n  return { context, metadata }\n}\n\nexport type { ClientTool, ClientToolMap } from './clientTools'\n","// AudioWorklet — runs off the main thread in the audio rendering graph.\n//\n// Input: Float32 samples at the AudioContext's native sampleRate (typically\n// 48000 Hz on desktop, 44100 Hz on some iOS devices).\n// Output: 16 kHz mono Int16 PCM, shipped to the main thread via\n// `port.postMessage(ArrayBuffer, [ArrayBuffer])` (transferred, not copied).\n//\n// Why AudioWorklet instead of ScriptProcessorNode: ScriptProcessorNode is\n// deprecated + main-thread-bound, so any JS jank produces audible audio\n// glitches. AudioWorklet's `process()` runs on the audio rendering thread\n// at the graph's block cadence (128 frames by default) and backpressures\n// via returning `true` / `false`.\n//\n// This file is loaded as text (see tsup.config.ts loader) and registered\n// at runtime via `audioWorklet.addModule(blobUrl)`.\n\nclass MicDownsampler extends AudioWorkletProcessor {\n  constructor() {\n    super()\n    // Target sample rate for STT. Matches Deepgram Nova-3 + the platform's\n    // server-side SAMPLE_RATE constant in AgentCallHandler.\n    this.targetRate = 16000\n    // Accumulator for the downsample. We collect incoming samples and emit\n    // an Int16 chunk when we've accumulated ~1024 target-rate samples\n    // (~64 ms at 16 kHz) — matches the mobile SDK's chunk size so both\n    // platforms have the same server-side framing.\n    this.outputFrames = 1024\n    this.acc = []\n    // Running index used for fractional resampling.\n    this.readCursor = 0\n  }\n\n  // `inputs[0][0]` = first channel of first input. 128 Float32 samples per\n  // call at the context's sampleRate. Return true = keep processing.\n  process(inputs) {\n    const input = inputs[0]\n    if (!input || input.length === 0) return true\n    const channel = input[0]\n    if (!channel || channel.length === 0) return true\n\n    const ctxRate = sampleRate // global inside AudioWorkletProcessor\n    const ratio = ctxRate / this.targetRate\n\n    // Simple linear-interp downsample. For 48000 → 16000 that's 3:1, which\n    // linear handles fine for voice. Anti-alias filtering would be\n    // theoretically better but inaudible for speech.\n    for (let i = 0; i < channel.length; i++) {\n      this.acc.push(channel[i])\n    }\n\n    while (this.acc.length - this.readCursor >= ratio * this.outputFrames) {\n      const out = new Int16Array(this.outputFrames)\n      let readIdx = this.readCursor\n      for (let i = 0; i < this.outputFrames; i++) {\n        // Linear interp between floor(readIdx) and ceil(readIdx)\n        const low = Math.floor(readIdx)\n        const high = Math.min(low + 1, this.acc.length - 1)\n        const frac = readIdx - low\n        const sample = this.acc[low] * (1 - frac) + this.acc[high] * frac\n        // Clip + convert to int16\n        const clipped = Math.max(-1, Math.min(1, sample))\n        out[i] = clipped < 0 ? clipped * 0x8000 : clipped * 0x7fff\n        readIdx += ratio\n      }\n      // Transfer the ArrayBuffer (zero-copy) to the main thread.\n      this.port.postMessage(out.buffer, [out.buffer])\n      this.readCursor = readIdx\n    }\n\n    // Garbage-collect the consumed portion of `acc` every so often so it\n    // doesn't grow without bound. Leave ~one chunk of headroom.\n    if (this.readCursor > ratio * this.outputFrames) {\n      this.acc = this.acc.slice(Math.floor(this.readCursor))\n      this.readCursor -= Math.floor(this.readCursor)\n    }\n\n    return true\n  }\n}\n\nregisterProcessor('mic-downsampler', MicDownsampler)\n","// Mic capture pipeline:\n//   getUserMedia → AudioContext → AudioWorkletNode → Int16 PCM chunks\n//\n// We register the worklet from an inline Blob URL so consumers don't need\n// to host or bundle a separate .js file. The worklet source is loaded as\n// text at build time via tsup's custom loader (see tsup.config.ts).\n\n// eslint-disable-next-line @typescript-eslint/ban-ts-comment\n// @ts-ignore — text-loaded by tsup\nimport workletSource from './worklets/mic-downsampler.worklet.js'\n\nexport type OnChunk = (pcm: ArrayBuffer) => void\nexport type OnVolume = (rms01: number) => void\nexport type OnError = (err: Error) => void\n\nexport interface CaptureOptions {\n  onChunk: OnChunk\n  // Optional — when set we compute input RMS and call this at ~10Hz. Used\n  // for volume visualisers in consumer UIs.\n  onVolume?: OnVolume\n  onError?: OnError\n}\n\nexport interface CaptureController {\n  start: () => Promise<void>\n  stop: () => void\n  mute: (muted: boolean) => void\n  isCapturing: () => boolean\n}\n\n// Volume-meter cadence. 100ms = 10Hz updates; smoother than per-chunk\n// (16Hz) without flicker.\nconst VOLUME_INTERVAL_MS = 100\n\nexport const createAudioCapture = (options: CaptureOptions): CaptureController => {\n  let audioContext: AudioContext | null = null\n  let mediaStream: MediaStream | null = null\n  let sourceNode: MediaStreamAudioSourceNode | null = null\n  let workletNode: AudioWorkletNode | null = null\n  let analyser: AnalyserNode | null = null\n  let volumeTimer: ReturnType<typeof setInterval> | null = null\n  let muted = false\n  let capturing = false\n\n  const computeRms = (buf: Float32Array): number => {\n    let sum = 0\n    for (let i = 0; i < buf.length; i++) sum += buf[i] * buf[i]\n    const rms = Math.sqrt(sum / buf.length)\n    // Clamp to 0..1 — RMS of a full-scale signal is 0.707, so boost a bit\n    // so \"normal speech\" hits ~0.5 in the visualiser.\n    return Math.min(1, rms * 1.8)\n  }\n\n  const start = async () => {\n    if (capturing) return\n    try {\n      // Browsers only grant getUserMedia on secure contexts (https or\n      // localhost). The thrown DOMException is clear enough — we let it\n      // propagate with a wrapped error message for clarity.\n      mediaStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          // Hand tuning for voice agent use: we want the raw signal so the\n          // server-side STT can do its own noise handling. Disable browser\n          // AEC/AGC/NR — experimentally they fight with whatever processing\n          // the TTS playback path feeds back in over speakers.\n          echoCancellation: true,\n          noiseSuppression: true,\n          autoGainControl: true,\n          channelCount: 1,\n        },\n      })\n\n      audioContext = new AudioContext()\n      // Most browsers suspend AudioContext until a user gesture. connect()\n      // is idempotent; if we were constructed inside a click/touch handler\n      // it's already running.\n      if (audioContext.state === 'suspended') await audioContext.resume()\n\n      // Load + register the worklet from an inline Blob URL. One-time per\n      // context; subsequent `new AudioWorkletNode` calls are cheap.\n      const blob = new Blob([workletSource], { type: 'application/javascript' })\n      const url = URL.createObjectURL(blob)\n      try {\n        await audioContext.audioWorklet.addModule(url)\n      } finally {\n        URL.revokeObjectURL(url)\n      }\n\n      sourceNode = audioContext.createMediaStreamSource(mediaStream)\n      workletNode = new AudioWorkletNode(audioContext, 'mic-downsampler')\n\n      // Worklet → main thread: Int16 PCM chunks\n      workletNode.port.onmessage = (event: MessageEvent<ArrayBuffer>) => {\n        if (muted) return\n        options.onChunk(event.data)\n      }\n\n      // Analyser sits in parallel to the worklet for the volume meter. Not\n      // strictly necessary — we could compute RMS inside the worklet and\n      // post it too — but AnalyserNode avoids extra bridge chatter.\n      if (options.onVolume) {\n        analyser = audioContext.createAnalyser()\n        analyser.fftSize = 256\n        sourceNode.connect(analyser)\n        const buf = new Float32Array(analyser.fftSize)\n        volumeTimer = setInterval(() => {\n          if (!analyser) return\n          analyser.getFloatTimeDomainData(buf)\n          options.onVolume?.(computeRms(buf))\n        }, VOLUME_INTERVAL_MS)\n      }\n\n      sourceNode.connect(workletNode)\n      // The worklet doesn't need to feed into the destination — its only\n      // job is to emit messages. But the AudioContext graph keeps the\n      // worklet running only if it's connected somewhere; connect to a\n      // zero-gain node to park it.\n      const sink = audioContext.createGain()\n      sink.gain.value = 0\n      workletNode.connect(sink).connect(audioContext.destination)\n\n      capturing = true\n    } catch (err) {\n      const wrapped =\n        err instanceof Error ? err : new Error(typeof err === 'string' ? err : 'capture failed')\n      options.onError?.(wrapped)\n      throw wrapped\n    }\n  }\n\n  const stop = () => {\n    if (!capturing) return\n    capturing = false\n    if (volumeTimer) {\n      clearInterval(volumeTimer)\n      volumeTimer = null\n    }\n    try {\n      workletNode?.disconnect()\n      analyser?.disconnect()\n      sourceNode?.disconnect()\n    } catch {\n      // already disconnected\n    }\n    workletNode = null\n    analyser = null\n    sourceNode = null\n    if (mediaStream) {\n      for (const track of mediaStream.getTracks()) track.stop()\n      mediaStream = null\n    }\n    if (audioContext && audioContext.state !== 'closed') {\n      void audioContext.close().catch(() => undefined)\n    }\n    audioContext = null\n  }\n\n  return {\n    start,\n    stop,\n    mute: (v) => {\n      muted = v\n    },\n    isCapturing: () => capturing,\n  }\n}\n","// Playback pipeline:\n//   WebSocket binary frame (Int16 PCM @ 16 kHz) → AudioBufferSourceNode\n//   chained for gapless playback → AudioContext destination.\n//\n// We deliberately don't use MediaSource or a single streaming source —\n// each agent TTS chunk is scheduled as its own short AudioBuffer so we\n// can interrupt (flush the queue) the moment a barge-in lands. One chunk\n// ≈ 20–100ms of audio.\n\nexport type OnVolume = (rms01: number) => void\nexport type OnAgentSpeakingChange = (speaking: boolean) => void\n\nexport interface PlaybackOptions {\n  // Sample rate of the incoming PCM — matches AgentCallHandler.SAMPLE_RATE.\n  sampleRate?: number\n  onVolume?: OnVolume\n  onSpeakingChange?: OnAgentSpeakingChange\n}\n\nexport interface PlaybackController {\n  enqueue: (pcm: ArrayBuffer) => void\n  flush: () => void\n  close: () => void\n  resume: () => Promise<void>\n}\n\nconst DEFAULT_SAMPLE_RATE = 16_000\nconst VOLUME_INTERVAL_MS = 100\n\nexport const createAudioPlayback = (options: PlaybackOptions = {}): PlaybackController => {\n  const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE\n  let audioContext: AudioContext | null = null\n  let gainNode: GainNode | null = null\n  let analyser: AnalyserNode | null = null\n  let volumeTimer: ReturnType<typeof setInterval> | null = null\n  let nextStartTime = 0\n  let scheduledNodes: AudioBufferSourceNode[] = []\n  let speaking = false\n\n  const ensureContext = async () => {\n    if (audioContext) {\n      if (audioContext.state === 'suspended') await audioContext.resume()\n      return\n    }\n    // Pass the target sample rate as a hint. Browsers may choose a\n    // different rate (48000 is most common); `createBuffer(numChannels,\n    // length, sampleRate)` with a different rate than the context will\n    // trigger resampling on `start()`, which the browser handles natively.\n    audioContext = new AudioContext({ sampleRate })\n    gainNode = audioContext.createGain()\n    if (options.onVolume) {\n      analyser = audioContext.createAnalyser()\n      analyser.fftSize = 256\n      gainNode.connect(analyser)\n      const buf = new Float32Array(analyser.fftSize)\n      volumeTimer = setInterval(() => {\n        if (!analyser) return\n        analyser.getFloatTimeDomainData(buf)\n        let sum = 0\n        for (let i = 0; i < buf.length; i++) sum += buf[i] * buf[i]\n        const rms = Math.sqrt(sum / buf.length)\n        options.onVolume?.(Math.min(1, rms * 1.8))\n      }, VOLUME_INTERVAL_MS)\n    }\n    gainNode.connect(audioContext.destination)\n    nextStartTime = audioContext.currentTime\n  }\n\n  const setSpeaking = (v: boolean) => {\n    if (v === speaking) return\n    speaking = v\n    options.onSpeakingChange?.(v)\n  }\n\n  const pruneFinished = () => {\n    const now = audioContext?.currentTime ?? 0\n    scheduledNodes = scheduledNodes.filter((n) => {\n      const node = n as AudioBufferSourceNode & { _endsAt?: number }\n      return (node._endsAt ?? 0) > now\n    })\n    if (scheduledNodes.length === 0) setSpeaking(false)\n  }\n\n  const enqueue = (pcm: ArrayBuffer) => {\n    // We don't want to await here — WS onmessage is hot-path, and AudioContext\n    // resume is cheap once warmed. `void` the promise and proceed on next\n    // tick; if the context isn't ready, the first chunk may land a hair\n    // late but subsequent chunks catch up.\n    if (!audioContext) {\n      void ensureContext().then(() => enqueue(pcm))\n      return\n    }\n    if (!audioContext || !gainNode) return\n\n    const int16 = new Int16Array(pcm)\n    if (int16.length === 0) return\n\n    const audioBuffer = audioContext.createBuffer(1, int16.length, sampleRate)\n    const float32 = audioBuffer.getChannelData(0)\n    for (let i = 0; i < int16.length; i++) {\n      float32[i] = int16[i] / 0x8000\n    }\n\n    const node = audioContext.createBufferSource()\n    node.buffer = audioBuffer\n    node.connect(gainNode)\n\n    const now = audioContext.currentTime\n    // Schedule back-to-back. If we've fallen behind (mic lag, WS gap),\n    // re-anchor to now so we don't skew the whole queue further.\n    const startAt = Math.max(now, nextStartTime)\n    node.start(startAt)\n\n    const duration = int16.length / sampleRate\n    ;(node as AudioBufferSourceNode & { _endsAt?: number })._endsAt = startAt + duration\n    nextStartTime = startAt + duration\n\n    scheduledNodes.push(node)\n    setSpeaking(true)\n\n    // Prune after every enqueue so `speaking` collapses quickly when the\n    // queue drains. Cheap O(n) over a short list.\n    node.onended = () => pruneFinished()\n  }\n\n  // Barge-in: stop anything pending, reset the schedule pointer.\n  //\n  // We can't simply `.stop()` playing sources mid-buffer and expect the\n  // AudioContext's destination to be silent immediately — Safari in\n  // particular sometimes plays out a few ms of tail. Reliable approach:\n  // swap the gainNode out so nothing after this point is audible, then\n  // clean up the old one on the next tick.\n  const flush = () => {\n    if (!audioContext || !gainNode) return\n    for (const node of scheduledNodes) {\n      try {\n        node.stop()\n      } catch {\n        // already stopped\n      }\n    }\n    scheduledNodes = []\n    // Swap gainNode so even stubborn trailing samples on the old node\n    // don't reach the speakers.\n    gainNode.disconnect()\n    gainNode = audioContext.createGain()\n    if (analyser) {\n      analyser.disconnect()\n      gainNode.connect(analyser)\n    }\n    gainNode.connect(audioContext.destination)\n    nextStartTime = audioContext.currentTime\n    setSpeaking(false)\n  }\n\n  const close = () => {\n    flush()\n    if (volumeTimer) {\n      clearInterval(volumeTimer)\n      volumeTimer = null\n    }\n    if (audioContext && audioContext.state !== 'closed') {\n      void audioContext.close().catch(() => undefined)\n    }\n    audioContext = null\n    gainNode = null\n    analyser = null\n  }\n\n  const resume = async () => {\n    await ensureContext()\n  }\n\n  return { enqueue, flush, close, resume }\n}\n","// Minimal auto-reconnecting WebSocket wrapper.\n//\n// Scope: transparent reconnection on unexpected drops (network blip,\n// server restart). We deliberately do NOT try to preserve call state\n// across reconnects — the server's AgentCallHandler is session-scoped\n// and won't resume where we left off. If the consumer needs mid-call\n// resilience, that's a server-side resumable-session feature, not a\n// client-side retry.\n//\n// In v1 we reconnect only when `connect()` has been called and\n// `.close(1000, ...)` has NOT been called by the consumer (i.e., we drop\n// because of network, not because the user hung up). On unclean close\n// the consumer gets a fresh connection + a `reconnected` event so they\n// can re-issue any greeting / context.\n//\n// Transport agnostic — accepts a WebSocket-constructor factory so the\n// browser bundle uses native `WebSocket` and the node bundle injects the\n// `ws` package.\n\nexport type RWSEvent =\n  | { type: 'open' }\n  | { type: 'reconnected' }\n  | { type: 'message'; data: string | ArrayBuffer }\n  | { type: 'close'; code: number; reason: string; permanent: boolean }\n  | { type: 'error'; error: Error }\n\n// Minimal WebSocket-like contract. Both browser `WebSocket` and the\n// `ws` package's WebSocket satisfy it.\nexport interface WebSocketLike {\n  binaryType: string\n  readyState: number\n  onopen: ((ev: unknown) => void) | null\n  onmessage: ((ev: { data: string | ArrayBuffer }) => void) | null\n  onerror: ((ev: unknown) => void) | null\n  onclose: ((ev: { code: number; reason: string }) => void) | null\n  send: (data: string | ArrayBuffer | ArrayBufferView) => void\n  close: (code?: number, reason?: string) => void\n}\n\nexport type WebSocketFactory = (url: string) => WebSocketLike\n\nexport interface RWSOptions {\n  url: string\n  // Factory so we can swap browser-native WebSocket for the `ws` package\n  // in node. Browser entry pre-fills with a wrapper around globalThis.WebSocket.\n  wsFactory: WebSocketFactory\n  // Cap the number of auto-reconnect attempts. `0` disables retry entirely\n  // (closer to native WebSocket semantics). Default: 3.\n  maxRetries?: number\n  // Initial backoff; doubles up to maxBackoffMs.\n  initialBackoffMs?: number\n  maxBackoffMs?: number\n}\n\nconst READYSTATE_OPEN = 1\nconst READYSTATE_CLOSED = 3\n\nexport const createReconnectingWebSocket = (\n  options: RWSOptions,\n  onEvent: (ev: RWSEvent) => void,\n) => {\n  const maxRetries = options.maxRetries ?? 3\n  const initialBackoff = options.initialBackoffMs ?? 500\n  const maxBackoff = options.maxBackoffMs ?? 8000\n\n  let ws: WebSocketLike | null = null\n  let intentionalClose = false\n  let retries = 0\n  let backoff = initialBackoff\n  let reconnectTimer: ReturnType<typeof setTimeout> | null = null\n\n  const openOnce = () => {\n    ws = options.wsFactory(options.url)\n    // Binary frames arrive as Int16 PCM from the server. ArrayBuffer is\n    // more convenient than Blob (one fewer async read) — we `.arrayBuffer()`\n    // a Blob manually only if we receive one unexpectedly.\n    ws.binaryType = 'arraybuffer'\n    ws.onopen = () => {\n      if (retries === 0) onEvent({ type: 'open' })\n      else onEvent({ type: 'reconnected' })\n      retries = 0\n      backoff = initialBackoff\n    }\n    ws.onmessage = (ev) => {\n      onEvent({ type: 'message', data: ev.data as string | ArrayBuffer })\n    }\n    ws.onerror = () => {\n      onEvent({ type: 'error', error: new Error('WebSocket error') })\n    }\n    ws.onclose = (ev) => {\n      ws = null\n      const shouldRetry = !intentionalClose && retries < maxRetries\n      if (!shouldRetry) {\n        onEvent({\n          type: 'close',\n          code: ev.code,\n          reason: ev.reason,\n          permanent: true,\n        })\n        return\n      }\n      onEvent({\n        type: 'close',\n        code: ev.code,\n        reason: ev.reason,\n        permanent: false,\n      })\n      retries++\n      const delay = Math.min(backoff, maxBackoff)\n      backoff = Math.min(backoff * 2, maxBackoff)\n      reconnectTimer = setTimeout(openOnce, delay)\n    }\n  }\n\n  openOnce()\n\n  return {\n    send: (data: string | ArrayBuffer | ArrayBufferView) => {\n      if (ws && ws.readyState === READYSTATE_OPEN) ws.send(data)\n    },\n    close: (code = 1000, reason = 'client-requested') => {\n      intentionalClose = true\n      if (reconnectTimer) {\n        clearTimeout(reconnectTimer)\n        reconnectTimer = null\n      }\n      try {\n        ws?.close(code, reason)\n      } catch {\n        // already closed\n      }\n    },\n    readyState: () => ws?.readyState ?? READYSTATE_CLOSED,\n  }\n}\n\nexport type ReconnectingWebSocket = ReturnType<typeof createReconnectingWebSocket>\n","// Wire-protocol types + stateless transcript reducer.\n//\n// Shared between the browser and node bundles. The protocol surface is\n// matched (where it makes sense) to @craftedxp/voice-rn 0.3.x so a\n// consumer can write the same `onTranscript` / `onStateChange` /\n// `onError` / `onEnd` shape for both web and RN clients.\n\nimport type { ClientToolCallFrame } from './clientTools'\n\n// Re-export so consumers / dashboards can type the wire shape. The\n// `client_marks` frame itself is purely an internal observability\n// channel between the SDK and server — it's emitted by the SDK and\n// logged on the server, never surfaced to consumer code.\nexport type { ClientMarksFrame } from './ClientMarksBuffer'\n\n// ---------------------------------------------------------------------------\n// State machine\n// ---------------------------------------------------------------------------\n\nexport type CallState =\n  | 'idle'\n  | 'connecting'\n  | 'listening'\n  | 'user_speaking'\n  | 'agent_speaking'\n  | 'ended'\n  | 'error'\n\n// ---------------------------------------------------------------------------\n// Transcript model — matches voice-rn TranscriptEntry\n// ---------------------------------------------------------------------------\n\nexport type TranscriptEntry =\n  | { id: string; role: 'user'; text: string; committed: boolean }\n  | { id: string; role: 'agent'; text: string; interrupted?: boolean }\n  | { id: string; role: 'tool'; text: string }\n  | { id: string; role: 'system'; text: string }\n\n// ---------------------------------------------------------------------------\n// Stable error code contract — matches voice-rn CallErrorCode where the\n// failure modes overlap. Web-specific codes (mic_denied via getUserMedia\n// rejection, etc.) keep their voice-rn names so cross-platform consumers\n// can write one switch statement.\n// ---------------------------------------------------------------------------\n\nexport type CallErrorCode =\n  // Programming errors — surface loudly to the host's developer.\n  | 'missing_credentials'\n  | 'forbidden'\n  // Browser audio failures.\n  | 'mic_denied'\n  | 'mic_start_failed'\n  | 'audio_session_failed'\n  // Auth lifecycle.\n  | 'token_expired'\n  | 'token_invalid'\n  | 'unauthorized'\n  // Network / connectivity.\n  | 'network_unreachable'\n  | 'socket_error'\n  // Business state.\n  | 'payment_required'\n  | 'not_found'\n  // End-of-call states surfaced via onError (also via onEnd reason='timeout').\n  | 'silence_timeout'\n  // Catch-all for unexpected server / 5xx / 1011.\n  | 'server_error'\n\nexport interface CallError {\n  code: CallErrorCode\n  message: string\n}\n\n// ---------------------------------------------------------------------------\n// End-of-call signal — matches voice-rn CallEndReason / CallEndEvent\n// ---------------------------------------------------------------------------\n\nexport type CallEndReason = 'agent_ended' | 'user_hangup' | 'timeout' | 'error'\n\nexport interface CallEndEvent {\n  reason: CallEndReason\n  // Present iff reason === 'error'. Mirrors the code from the most\n  // recent onError.\n  errorCode?: CallErrorCode\n  // Wallclock from start() resolving to the WS close. Useful for billing\n  // / \"you spoke for 1m23s\" UIs without forcing the host to track\n  // timestamps themselves.\n  durationMs: number\n}\n\n// ---------------------------------------------------------------------------\n// Volume meter event (browser bundle only — node bundle leaves volume to\n// the consumer if they're processing PCM themselves).\n// ---------------------------------------------------------------------------\n\nexport interface VolumeEvent {\n  // 0-1 RMS over the last ~100ms. Bind to a waveform / level meter.\n  input: number\n  output: number\n}\n\n// ---------------------------------------------------------------------------\n// Server → client message envelope. Loose typing — the server can add new\n// `type` values without breaking the SDK; unknown types are ignored by\n// the dispatch in handleServerMessage.\n// ---------------------------------------------------------------------------\n\nexport type ServerMessage = Record<string, unknown> & { type?: string }\n\n// ---------------------------------------------------------------------------\n// Stateless transcript reducer + state-machine helpers. Both the browser\n// and node clients call into these so the shape of the transcript stays\n// identical across environments.\n// ---------------------------------------------------------------------------\n\nexport interface ProtocolState {\n  state: CallState\n  transcript: TranscriptEntry[]\n  agentBubbleId: string | null\n  idCounter: number\n  // Reason latched from the server's call_end frame. Read by the\n  // surrounding client when the WS finally closes so onEnd fires with the\n  // right reason instead of falling back to user_hangup.\n  endReason: CallEndReason | null\n}\n\nexport const createProtocolState = (): ProtocolState => ({\n  state: 'idle',\n  transcript: [],\n  agentBubbleId: null,\n  idCounter: 0,\n  endReason: null,\n})\n\n// Side-effect callbacks the protocol layer fires as it processes server\n// frames. The surrounding client wires these up to event emitters\n// (browser) or to user-supplied callbacks (node).\nexport interface ProtocolCallbacks {\n  onState: (next: CallState) => void\n  onTranscript: (entries: TranscriptEntry[]) => void\n  onError: (err: CallError) => void\n  // Fires on `interrupt` — caller should flush its audio playback queue.\n  onInterrupt: () => void\n  // Fires on `agent_turn_start` — caller may want to reset its turn\n  // anchor for \"agent has been speaking N ms\" UIs. `seq` is the\n  // server-assigned turn ID; absent on legacy server builds that don't\n  // emit one (we treat that case as \"no correlation possible\").\n  onAgentTurnStart: (seq?: number) => void\n  // Fires on `agent_turn_end` — caller may use this to flush per-turn\n  // observability marks. `seq` matches the corresponding agent_turn_start.\n  onAgentTurnEnd: (seq?: number) => void\n  // Fires on `call_end` — caller closes its WS and resolves onEnd.\n  onCallEnd: (reason: CallEndReason) => void\n  // Fires on `connected` — caller should send the client_tools_register\n  // frame here so the server's first-turn grace window unblocks.\n  onConnected: () => void\n  // Fires on `client_tool_call` — caller dispatches to the matching\n  // tool handler and posts back a client_tool_result frame.\n  onClientToolCall: (frame: ClientToolCallFrame) => void\n}\n\n// Map server-supplied endReason strings onto our SDK-side CallEndReason.\nconst mapEndReason = (raw: string): CallEndReason => {\n  if (raw === 'agent_ended') return 'agent_ended'\n  if (raw === 'caller_hung_up') return 'user_hangup'\n  if (raw === 'silence_timeout' || raw === 'max_duration') return 'timeout'\n  return 'error'\n}\n\n// Pure-ish transcript reducer + dispatcher. Mutates `state` in place to\n// match the imperative pattern used by both clients; returns nothing.\n//\n// The \"agent_text\" / \"transcript\" interim handling is identical to\n// voice-rn's useVoiceCall — one growing user bubble while interim, one\n// growing agent bubble per turn.\nexport function handleServerMessage(\n  raw: string,\n  state: ProtocolState,\n  cb: ProtocolCallbacks,\n): void {\n  let msg: ServerMessage\n  try {\n    msg = JSON.parse(raw)\n  } catch {\n    return\n  }\n\n  switch (msg.type) {\n    case 'connected':\n      cb.onConnected()\n      setState(state, 'listening', cb)\n      return\n\n    case 'transcript': {\n      const text = (msg.text as string) ?? ''\n      if (!text) return\n      const isFinal = !!msg.isFinal\n      if (!isFinal) setState(state, 'user_speaking', cb)\n      upsertUserPartial(state, text, isFinal)\n      cb.onTranscript(state.transcript)\n      return\n    }\n\n    case 'agent_turn_start': {\n      const id = `m${state.idCounter++}`\n      state.agentBubbleId = id\n      state.transcript = [...state.transcript, { id, role: 'agent', text: '' }]\n      cb.onTranscript(state.transcript)\n      const seq = typeof msg.seq === 'number' ? (msg.seq as number) : undefined\n      cb.onAgentTurnStart(seq)\n      setState(state, 'agent_speaking', cb)\n      return\n    }\n\n    case 'agent_text': {\n      const delta = (msg.text as string) ?? ''\n      if (!delta || !state.agentBubbleId) return\n      const id = state.agentBubbleId\n      state.transcript = state.transcript.map((e) =>\n        e.id === id && e.role === 'agent' ? { ...e, text: e.text + delta } : e,\n      )\n      cb.onTranscript(state.transcript)\n      return\n    }\n\n    case 'agent_turn_end': {\n      state.agentBubbleId = null\n      const seq = typeof msg.seq === 'number' ? (msg.seq as number) : undefined\n      cb.onAgentTurnEnd(seq)\n      setState(state, 'listening', cb)\n      return\n    }\n\n    case 'interrupt':\n      cb.onInterrupt()\n      return\n\n    case 'agent_turn_abort': {\n      const committed = ((msg.committedText as string) ?? '').trim()\n      if (state.agentBubbleId) {\n        const id = state.agentBubbleId\n        if (committed) {\n          state.transcript = state.transcript.map((e) =>\n            e.id === id && e.role === 'agent' ? { ...e, text: committed, interrupted: true } : e,\n          )\n        } else {\n          state.transcript = state.transcript.filter((e) => e.id !== id)\n        }\n        cb.onTranscript(state.transcript)\n      }\n      state.agentBubbleId = null\n      return\n    }\n\n    case 'tool_call':\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'tool',\n          text: `→ ${String(msg.tool ?? '?')}(${msg.args ? JSON.stringify(msg.args) : ''})`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      return\n\n    case 'tool_result':\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'tool',\n          text: `${msg.ok ? '✓' : '✗'} ${String(msg.tool ?? '?')}`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      return\n\n    case 'client_tool_call': {\n      const toolCallId = String(msg.toolCallId ?? '')\n      const name = String(msg.name ?? '')\n      const args = (msg.args as Record<string, unknown>) ?? {}\n      if (!toolCallId || !name) return\n      cb.onClientToolCall({ toolCallId, name, args })\n      return\n    }\n\n    case 'call_end': {\n      const reasonRaw = String(msg.reason ?? '')\n      const reason = mapEndReason(reasonRaw)\n      state.endReason = reason\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'system',\n          text: `call ended${reasonRaw ? ` (${reasonRaw})` : ''}`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      cb.onCallEnd(reason)\n      return\n    }\n\n    case 'error': {\n      const code = (msg.code as CallErrorCode) ?? 'server_error'\n      const message = (msg.message as string) ?? 'server error'\n      cb.onError({ code, message })\n      return\n    }\n  }\n}\n\nconst setState = (state: ProtocolState, next: CallState, cb: ProtocolCallbacks) => {\n  if (state.state === next) return\n  // The consumer's setState (cb.onState) is responsible for both\n  // updating state.state and firing the user's onStateChange callback.\n  // We deliberately don't mutate here — if we did, the consumer's\n  // setState would early-return on the dedup check and never fire\n  // onStateChange for server-driven transitions.\n  cb.onState(next)\n}\n\n// Find the last uncommitted user bubble and grow it; or append a new\n// uncommitted bubble. Mirrors voice-rn upsertUserPartial.\nconst upsertUserPartial = (state: ProtocolState, text: string, isFinal: boolean) => {\n  let idx = -1\n  for (let i = state.transcript.length - 1; i >= 0; i--) {\n    const e = state.transcript[i]\n    if (e.role === 'user' && e.committed === false) {\n      idx = i\n      break\n    }\n  }\n  if (idx === -1) {\n    state.transcript = [\n      ...state.transcript,\n      { id: `m${state.idCounter++}`, role: 'user', text, committed: isFinal },\n    ]\n    return\n  }\n  const target = state.transcript[idx] as Extract<TranscriptEntry, { role: 'user' }>\n  const next = [...state.transcript]\n  next[idx] = { ...target, text, committed: isFinal }\n  state.transcript = next\n}\n\n// ---------------------------------------------------------------------------\n// WebSocket URL builder. Identical between browser + node bundles —\n// kept here so the rare server-side change to the path / query shape is\n// a one-line edit.\n// ---------------------------------------------------------------------------\n\nexport interface BuildWsUrlArgs {\n  apiBase: string\n  agentId: string\n  token: string\n  bargeIn?: boolean\n}\n\nexport function buildWsUrl(args: BuildWsUrlArgs): string {\n  const base = new URL(args.apiBase)\n  const proto = base.protocol === 'https:' ? 'wss:' : 'ws:'\n  const bargeQS = args.bargeIn === false ? '&barge=off' : ''\n  return `${proto}//${base.host}/v1/agents/${encodeURIComponent(args.agentId)}/call?token=${encodeURIComponent(args.token)}${bargeQS}`\n}\n","// Bundle-agnostic client-tools core for @craftedxp/voice-js.\n//\n// Imports neither `ws` nor DOM `WebSocket` — the dispatcher takes an\n// injected `send` callback so the same code runs in browsers and Node.\n//\n// Wire protocol (matches the server's Zod schemas in\n// server/src/agents/clientToolFrames.ts):\n//\n//   client → server  client_tools_register   { type, tools[] }\n//   server → client  client_tool_call        { type, toolCallId, name, args }\n//   client → server  client_tool_result      { type, toolCallId, result?, error? }\n\nexport interface ClientTool {\n  description: string\n  parameters: Record<string, unknown> // JSON Schema\n  usage?: string // ≤500 chars; appended to agent system prompt\n  timeoutMs?: number // server-enforced; (0, 30_000]\n  example?: string // surface-only UI hint; not sent on wire\n  handler: (args: Record<string, unknown>) => Promise<string | object> | string | object\n}\n\nexport type ClientToolMap = Record<string, ClientTool>\n\n// Server → client frame shape. Used by both protocol.ts (when surfacing\n// the frame to a callback) and dispatchClientToolCall (when consuming it).\nexport interface ClientToolCallFrame {\n  toolCallId: string\n  name: string\n  args: Record<string, unknown>\n}\n\nconst NAME_RE = /^[a-zA-Z_][a-zA-Z0-9_]*$/\nconst MAX_TOOLS = 64\nconst MAX_USAGE = 500\nconst MAX_TIMEOUT_MS = 30_000\n\n// Synchronous, throw-on-error validation. Called once at startCall time\n// before the WS opens, so bad input fails fast at the configuration\n// boundary instead of mid-call.\nexport const validateClientToolMap = (tools: ClientToolMap | undefined): void => {\n  if (tools === undefined) return\n  if (typeof tools !== 'object' || tools === null || Array.isArray(tools)) {\n    throw new Error('clientTools must be an object keyed by tool name')\n  }\n  const entries = Object.entries(tools)\n  if (entries.length > MAX_TOOLS) {\n    throw new Error(`clientTools may declare at most 64 tools (got ${entries.length})`)\n  }\n  for (const [name, def] of entries) {\n    if (!NAME_RE.test(name)) {\n      throw new Error(\n        `clientTools[\"${name}\"]: name must be a valid identifier (^[a-zA-Z_][a-zA-Z0-9_]*$)`,\n      )\n    }\n    if (!def || typeof def !== 'object') {\n      throw new Error(`clientTools[\"${name}\"]: must be an object`)\n    }\n    if (typeof def.description !== 'string' || def.description.length === 0) {\n      throw new Error(`clientTools[\"${name}\"]: must have a description`)\n    }\n    if (typeof def.handler !== 'function') {\n      throw new Error(`clientTools[\"${name}\"]: must have a handler function`)\n    }\n    if (def.usage !== undefined && def.usage.length > MAX_USAGE) {\n      throw new Error(`clientTools[\"${name}\"]: usage must be ≤500 chars`)\n    }\n    if (\n      def.timeoutMs !== undefined &&\n      (!Number.isFinite(def.timeoutMs) || def.timeoutMs <= 0 || def.timeoutMs > MAX_TIMEOUT_MS)\n    ) {\n      throw new Error(`clientTools[\"${name}\"]: timeoutMs must be in (0, 30000]`)\n    }\n  }\n}\n\n// Build the registration frame the SDK sends right after `connected`.\n// Strips runtime-only fields (handler, example) and only includes the\n// optional fields when set.\nexport const buildRegisterFrame = (tools: ClientToolMap): object => ({\n  type: 'client_tools_register',\n  tools: Object.entries(tools).map(([name, def]) => ({\n    name,\n    description: def.description,\n    parameters: def.parameters,\n    ...(def.usage !== undefined ? { usage: def.usage } : {}),\n    ...(def.timeoutMs !== undefined ? { timeoutMs: def.timeoutMs } : {}),\n  })),\n})\n\ntype FrameSender = (frame: object) => void\n\n// Run the matching handler asynchronously and post a result/error frame\n// back through `send`. Send-side throws are swallowed (the WS may have\n// closed mid-call; the server has already cancelled the in-flight\n// invocation on its side).\nexport const dispatchClientToolCall = (\n  send: FrameSender,\n  tools: ClientToolMap,\n  frame: ClientToolCallFrame,\n): void => {\n  const safeSend = (payload: object) => {\n    try {\n      send(payload)\n    } catch {\n      // WS likely closed mid-handler. Server-side already cancelled.\n    }\n  }\n\n  const tool = tools[frame.name]\n  if (!tool) {\n    safeSend({\n      type: 'client_tool_result',\n      toolCallId: frame.toolCallId,\n      error: `No handler for ${frame.name}`,\n    })\n    return\n  }\n\n  void (async () => {\n    try {\n      const out = await tool.handler(frame.args)\n      safeSend({\n        type: 'client_tool_result',\n        toolCallId: frame.toolCallId,\n        result: typeof out === 'string' ? out : JSON.stringify(out),\n      })\n    } catch (err) {\n      safeSend({\n        type: 'client_tool_result',\n        toolCallId: frame.toolCallId,\n        error: err instanceof Error ? err.message : String(err),\n      })\n    }\n  })()\n}\n","// ClientMarksBuffer — per-turn latency-mark correlator shared between the\n// browser and node SDK clients.\n//\n// Why this exists:\n//   The server mints a `seq` (turnId) only AFTER the user finishes\n//   speaking, at VoiceOrchestrator.handleTranscript. The client, however,\n//   captures its first user-perceived mark — `t_client_mic_to_first_frame`\n//   — before any server frame for the turn exists. We therefore buffer\n//   client-side marks under a \"pending\" slot and back-fill the seq when\n//   the server's `agent_turn_start { seq }` arrives. When `agent_turn_end\n//   { seq }` lands AND we've captured the inbound `client_first_audible`\n//   mark, the buffer flushes a `client_marks` frame to the server.\n//\n// Lifetime + bounds:\n//   - One pending bucket at a time. A new `agent_turn_start` flushes\n//     anything still pending (shouldn't happen in normal flow, but is\n//     safer than silently dropping marks).\n//   - If `agent_turn_end` arrives but we never captured first_audible\n//     (e.g. agent had no audio output), we still emit a `client_marks`\n//     frame with whatever marks are present. Missing marks are simply\n//     omitted from the wire payload.\n//   - The buffer is GC'd by replacement on every turn boundary; there is\n//     no long-term retention. A turn whose `agent_turn_end` never\n//     arrives (server crash mid-turn) is implicitly dropped on the next\n//     `agent_turn_start` or when the call ends. We do NOT add an\n//     explicit watchdog — observability marks aren't worth the\n//     complexity of a timer here.\n//\n// Time source:\n//   `performance.now()` in both browser and Node. Wall-clock-monotonic,\n//   sub-ms precision. Marks are stored as raw `performance.now()` values;\n//   only the deltas between them ever go on the wire.\n//\n// Wire shape (see protocol.ts):\n//   { type: 'client_marks', seq, marks: { ... }, clientNow }\n//\n// The CLAUDE.md / functional-style preference: this module is a small\n// factory returning a closure-based controller. No classes, no shared\n// mutable module state.\n\nexport interface ClientMarksFrame {\n  type: 'client_marks'\n  seq: number\n  marks: {\n    // The headline number: client-measured wall-clock between the first\n    // outbound audio frame for this turn and the first agent-audio\n    // chunk committed for playback. Includes mic-encode time, network\n    // RTT in both directions, server-side processing (STT + LLM + TTS),\n    // and any client-side scheduling delay. Absent if either endpoint\n    // wasn't captured.\n    client_mic_to_first_audible_ms?: number\n  }\n  // Wall-clock at send time, ms-since-epoch. Lets the analysis script\n  // estimate clock skew between client and server logs even though the\n  // marks themselves are deltas.\n  clientNow?: number\n}\n\n// What the consumer-facing controller exposes. The browser and node\n// clients both wire to this same surface.\nexport interface ClientMarksController {\n  // Outbound mark. Called on every outgoing audio chunk; the buffer\n  // gates so only the first chunk PER TURN is timestamped. `markTurnBoundary`\n  // resets the gate so the next chunk is treated as the start of the\n  // next turn's outbound stream.\n  markFirstOutboundAudio: () => void\n\n  // Inbound mark. Called every time the SDK commits an agent-audio\n  // chunk for playback (browser: AudioContext schedule; node: handed to\n  // consumer via onAudioChunk). Gated to first-per-turn.\n  markFirstAudibleOutput: () => void\n\n  // Called when the SDK receives `agent_turn_start { seq }`. Tags any\n  // accumulated outbound marks with this seq. Also resets the\n  // first-outbound gate so the NEXT turn's outbound mark can be\n  // captured again. (Outbound is captured for a turn before its seq is\n  // known; inbound is captured after, with the seq tagged.)\n  onAgentTurnStart: (seq: number) => void\n\n  // Called when the SDK receives `agent_turn_end { seq }`. Triggers\n  // emission of the buffered `client_marks` frame for that seq, with\n  // whatever marks are present (missing ones omitted).\n  onAgentTurnEnd: (seq: number) => void\n\n  // Force-flush any pending state. Called on call end / teardown so\n  // the last turn's marks aren't dropped on hangup.\n  flush: () => void\n}\n\n// `send` is the raw WS-send sink injected by the surrounding client.\n// Stringified JSON in browser/node. Returns whether the send was\n// accepted (we don't surface failure beyond the boolean — observability\n// frames are best-effort).\nexport type ClientMarksSend = (frame: ClientMarksFrame) => void\n\n// `now` defaults to performance.now() but is injected so unit tests can\n// drive a fake clock deterministically.\nexport interface CreateClientMarksBufferArgs {\n  send: ClientMarksSend\n  now?: () => number\n}\n\nexport const createClientMarksBuffer = (\n  args: CreateClientMarksBufferArgs,\n): ClientMarksController => {\n  const now = args.now ?? (() => performance.now())\n\n  // Pending bucket — one in-flight turn at a time. We hold the OUTBOUND\n  // mark before we know `seq`, then graduate to a per-seq bucket on\n  // agent_turn_start so a second turn's inbound marks can't trample the\n  // first turn's outbound mark before flush.\n  let pendingFirstOutboundAt: number | null = null\n\n  // Once agent_turn_start { seq } arrives we move the pending outbound\n  // mark into a per-seq slot and start collecting the inbound mark.\n  // Map keyed by seq; in practice never holds more than 1-2 entries\n  // (one currently-flushing, one just-started).\n  const inFlight = new Map<\n    number,\n    {\n      firstOutboundAt: number | null\n      firstAudibleAt: number | null\n      ended: boolean\n    }\n  >()\n\n  const tryEmit = (seq: number) => {\n    const slot = inFlight.get(seq)\n    if (!slot) return\n    if (!slot.ended) return\n    // Build wire payload. Only include marks we actually captured.\n    const marks: ClientMarksFrame['marks'] = {}\n    if (slot.firstOutboundAt !== null && slot.firstAudibleAt !== null) {\n      marks.client_mic_to_first_audible_ms = slot.firstAudibleAt - slot.firstOutboundAt\n    }\n    args.send({\n      type: 'client_marks',\n      seq,\n      marks,\n      clientNow: Date.now(),\n    })\n    inFlight.delete(seq)\n  }\n\n  const markFirstOutboundAudio = () => {\n    if (pendingFirstOutboundAt !== null) return // already stamped this turn\n    pendingFirstOutboundAt = now()\n  }\n\n  const markFirstAudibleOutput = () => {\n    // Inbound is captured AFTER agent_turn_start, so we know which seq\n    // it belongs to: the most recent un-ended in-flight slot. If there's\n    // somehow none (race: audio before agent_turn_start), drop silently.\n    let target: { firstAudibleAt: number | null; ended: boolean } | undefined\n    for (const slot of inFlight.values()) {\n      if (!slot.ended) {\n        target = slot\n        // No break — last un-ended wins, matching the \"most recent\n        // turn\" intent if multiple are in flight.\n      }\n    }\n    if (!target) return\n    if (target.firstAudibleAt !== null) return // first-per-turn gate\n    target.firstAudibleAt = now()\n  }\n\n  const onAgentTurnStart = (seq: number) => {\n    inFlight.set(seq, {\n      firstOutboundAt: pendingFirstOutboundAt,\n      firstAudibleAt: null,\n      ended: false,\n    })\n    // Reset outbound gate so the NEXT turn captures cleanly. Note:\n    // pendingFirstOutboundAt is already moved into the slot above; we\n    // don't need to preserve it.\n    pendingFirstOutboundAt = null\n  }\n\n  const onAgentTurnEnd = (seq: number) => {\n    const slot = inFlight.get(seq)\n    if (!slot) {\n      // Got an end with no matching start (server bug, or restart). Best\n      // effort: emit an empty marks frame so the analysis script knows\n      // the client saw the turn at all.\n      args.send({ type: 'client_marks', seq, marks: {}, clientNow: Date.now() })\n      return\n    }\n    slot.ended = true\n    tryEmit(seq)\n  }\n\n  const flush = () => {\n    for (const seq of [...inFlight.keys()]) {\n      const slot = inFlight.get(seq)!\n      slot.ended = true\n      tryEmit(seq)\n    }\n    pendingFirstOutboundAt = null\n  }\n\n  return {\n    markFirstOutboundAudio,\n    markFirstAudibleOutput,\n    onAgentTurnStart,\n    onAgentTurnEnd,\n    flush,\n  }\n}\n","// Browser VoiceClient — drives one in-progress call.\n//\n// Created by `BrowserVoiceFactory.startCall` (see browser.ts). Most\n// consumers never touch this class directly — they receive a `Call`\n// handle that exposes `end`/`mute`/`unmute` and subscribe to events\n// via the `onState` / `onTranscript` / `onError` / `onEnd` callbacks\n// they passed to `startCall`.\n//\n// This is the rough equivalent of @craftedxp/voice-rn's `useVoiceCall`\n// hook, but environment-agnostic (no React).\n\nimport { createAudioCapture, type CaptureController } from './AudioCapture'\nimport { createAudioPlayback, type PlaybackController } from './AudioPlayback'\nimport {\n  createReconnectingWebSocket,\n  type ReconnectingWebSocket,\n  type WebSocketFactory,\n} from './ReconnectingWebSocket'\nimport {\n  buildWsUrl,\n  createProtocolState,\n  handleServerMessage,\n  type CallEndReason,\n  type CallError,\n  type ProtocolState,\n} from './protocol'\nimport type { Call, StartCallOptions, VoiceClientConfig } from './config'\nimport { buildRegisterFrame, dispatchClientToolCall, validateClientToolMap } from './clientTools'\nimport { createClientMarksBuffer, type ClientMarksController } from './ClientMarksBuffer'\n\ninterface VoiceClientArgs {\n  // Pre-resolved by the factory.\n  config: VoiceClientConfig\n  options: StartCallOptions\n  token: string\n  // Browser uses native WebSocket; node injects `ws`.\n  wsFactory: WebSocketFactory\n}\n\nexport class BrowserVoiceClient implements Call {\n  private readonly args: VoiceClientArgs\n  private readonly proto: ProtocolState\n\n  private rws: ReconnectingWebSocket | null = null\n  private capture: CaptureController | null = null\n  private playback: PlaybackController | null = null\n\n  private muted = false\n  private inputVolume = 0\n  private outputVolume = 0\n  private startedAt: number | null = null\n  private endedFired = false\n  private lastError: CallError | null = null\n  private marks: ClientMarksController\n\n  constructor(args: VoiceClientArgs) {\n    this.args = args\n    this.proto = createProtocolState()\n    validateClientToolMap(args.options.clientTools)\n    // Best-effort observability frame. Wraps WS send so a failed send\n    // (closed socket) is silently swallowed — these marks are never\n    // load-bearing for the call itself.\n    this.marks = createClientMarksBuffer({\n      send: (frame) => {\n        try {\n          this.rws?.send(JSON.stringify(frame))\n        } catch {\n          // WS already gone; drop the frame.\n        }\n      },\n    })\n  }\n\n  // ---------------------------------------------------------------\n  // Call interface\n  // ---------------------------------------------------------------\n\n  get state() {\n    return this.proto.state\n  }\n\n  get transcript() {\n    return this.proto.transcript.slice()\n  }\n\n  get isMuted() {\n    return this.muted\n  }\n\n  end = () => {\n    this.teardown('user_hangup')\n  }\n\n  mute = () => {\n    if (this.muted) return\n    this.muted = true\n    this.capture?.mute(true)\n  }\n\n  unmute = () => {\n    if (!this.muted) return\n    this.muted = false\n    this.capture?.mute(false)\n  }\n\n  // ---------------------------------------------------------------\n  // Lifecycle — called by the factory immediately after construction.\n  // Resolves once the WS is open and capture is starting; mid-call\n  // failures arrive via `onError`.\n  // ---------------------------------------------------------------\n\n  async start(): Promise<void> {\n    this.setState('connecting')\n    this.startedAt = Date.now()\n\n    const url = buildWsUrl({\n      apiBase: this.args.config.apiBase,\n      agentId: this.args.options.agentId,\n      token: this.args.token,\n      bargeIn: this.args.options.bargeIn,\n    })\n\n    // Playback context needs a user gesture before its AudioContext can\n    // produce sound. In practice startCall is invoked inside a click\n    // handler; resume() is cheap and idempotent.\n    this.playback = createAudioPlayback({\n      onVolume: (v) => {\n        this.outputVolume = v\n        this.args.options.onVolume?.({ input: this.inputVolume, output: v })\n      },\n    })\n    try {\n      await this.playback.resume()\n    } catch {\n      // AudioContext may refuse if we weren't called from a gesture.\n      // First enqueue() will retry the resume.\n    }\n\n    this.rws = createReconnectingWebSocket(\n      {\n        url,\n        wsFactory: this.args.wsFactory,\n        maxRetries: 3,\n      },\n      (ev) => this.handleSocketEvent(ev),\n    )\n  }\n\n  // ---------------------------------------------------------------\n  // Internal\n  // ---------------------------------------------------------------\n\n  private sendClientToolsRegister = () => {\n    const frame = buildRegisterFrame(this.args.options.clientTools ?? {})\n    this.rws?.send(JSON.stringify(frame))\n  }\n\n  private setState = (next: ProtocolState['state']) => {\n    if (this.proto.state === next) return\n    this.proto.state = next\n    this.args.options.onStateChange?.(next)\n  }\n\n  private emitError = (err: CallError) => {\n    this.lastError = err\n    this.args.options.onError?.(err)\n  }\n\n  private handleSocketEvent = (\n    ev:\n      | { type: 'open' }\n      | { type: 'reconnected' }\n      | { type: 'message'; data: string | ArrayBuffer }\n      | { type: 'close'; code: number; reason: string; permanent: boolean }\n      | { type: 'error'; error: Error },\n  ) => {\n    switch (ev.type) {\n      case 'open':\n        void this.startCapture()\n        break\n      case 'reconnected':\n        // Treat as a fresh call — server session is gone, no resume.\n        this.proto.transcript = []\n        this.proto.agentBubbleId = null\n        this.args.options.onTranscript?.(this.proto.transcript)\n        void this.startCapture()\n        this.setState('listening')\n        break\n      case 'message':\n        if (typeof ev.data === 'string') {\n          handleServerMessage(ev.data, this.proto, {\n            onState: this.setState,\n            onTranscript: (entries) => this.args.options.onTranscript?.(entries),\n            onError: this.emitError,\n            onInterrupt: () => {\n              this.playback?.flush()\n              this.args.options.onInterrupt?.()\n            },\n            onAgentTurnStart: (seq) => {\n              if (typeof seq === 'number') this.marks.onAgentTurnStart(seq)\n              this.args.options.onAgentTurnStart?.()\n            },\n            onAgentTurnEnd: (seq) => {\n              if (typeof seq === 'number') this.marks.onAgentTurnEnd(seq)\n            },\n            onCallEnd: (reason) => this.teardown(reason),\n            onConnected: () => this.sendClientToolsRegister(),\n            onClientToolCall: (frame) =>\n              dispatchClientToolCall(\n                (f) => this.rws?.send(JSON.stringify(f)),\n                this.args.options.clientTools ?? {},\n                frame,\n              ),\n          })\n        } else {\n          // Inbound mark: stamped at the moment we COMMIT this chunk for\n          // playback (enqueue schedules into AudioContext). We\n          // intentionally don't await — enqueue handles its own\n          // AudioContext warm-up; the mark captures the SDK's commit\n          // moment, which is the closest we can get to \"user heard it\"\n          // without instrumenting the AudioContext output graph.\n          this.marks.markFirstAudibleOutput()\n          this.playback?.enqueue(ev.data)\n        }\n        break\n      case 'close':\n        if (ev.permanent) {\n          // Pick the most informative reason: server-supplied wins, then\n          // last error, then default to user_hangup.\n          const reason: CallEndReason =\n            this.proto.endReason ?? (this.lastError ? 'error' : 'user_hangup')\n          this.teardown(reason)\n        }\n        break\n      case 'error':\n        this.emitError({ code: 'socket_error', message: ev.error.message })\n        break\n    }\n  }\n\n  private startCapture = async () => {\n    if (this.capture?.isCapturing()) return\n    this.capture = createAudioCapture({\n      onChunk: (pcm) => {\n        // Outbound mark — stamps the first frame leaving the SDK for\n        // each turn. The buffer gates so only the first chunk per\n        // pending-turn window is timestamped; subsequent chunks are\n        // no-ops until agent_turn_start moves the slot to a per-seq\n        // bucket and resets the gate. Semantic: this is \"moment audio\n        // is dispatched to the WS\", which in continuous open-mic mode\n        // is essentially the encode-and-handoff time per first frame\n        // after a turn boundary.\n        this.marks.markFirstOutboundAudio()\n        this.rws?.send(pcm)\n      },\n      onVolume: (v) => {\n        this.inputVolume = v\n        this.args.options.onVolume?.({ input: v, output: this.outputVolume })\n      },\n      onError: (err) => {\n        this.emitError({\n          code: err.name === 'NotAllowedError' ? 'mic_denied' : 'mic_start_failed',\n          message: err.message,\n        })\n      },\n    })\n    if (this.muted) this.capture.mute(true)\n    try {\n      await this.capture.start()\n    } catch {\n      // onError already emitted inside createAudioCapture.\n    }\n  }\n\n  private teardown = (reason: CallEndReason) => {\n    // Flush any in-flight observability marks BEFORE the WS closes so\n    // the final turn's client_marks frame doesn't get dropped on\n    // hangup.\n    try {\n      this.marks.flush()\n    } catch {\n      // never fatal\n    }\n    this.capture?.stop()\n    this.capture = null\n    this.playback?.close()\n    this.playback = null\n    try {\n      this.rws?.close(1000, reason)\n    } catch {\n      // already closed\n    }\n    this.rws = null\n    this.setState('ended')\n    this.fireEndOnce(reason)\n  }\n\n  private fireEndOnce = (reason: CallEndReason) => {\n    if (this.endedFired) return\n    this.endedFired = true\n    const startedAt = this.startedAt ?? Date.now()\n    this.args.options.onEnd?.({\n      reason,\n      errorCode: reason === 'error' ? this.lastError?.code : undefined,\n      durationMs: Date.now() - startedAt,\n    })\n  }\n}\n","// WebRTC transport for voice-js. Returns a `Call` with the same surface\n// as the WS path so consumers don't branch on transport. Signaling is\n// one-shot HTTP: POST /webrtc/offer (returns the SDP answer + callId);\n// trickled candidates go to POST /webrtc/ice. Audio flows over the\n// RTCPeerConnection; protocol frames (transcript, agent_turn_start, ...)\n// arrive on an in-band DataChannel named `control`.\n\nimport { createProtocolState, handleServerMessage, type ProtocolState } from '../protocol'\nimport type { CallError, CallState, TranscriptEntry } from '../protocol'\nimport type { Call } from '../config'\n\nexport interface CreateWebRtcCallOptions {\n  agentId: string\n  /** API base (`https://api.voissia.com` etc). Used as the signaling\n   *  fallback when `webrtcGatewayBase` is unset. */\n  apiBase: string\n  /** Pre-minted ct_. */\n  token: string\n  /** Gateway base from the mint response. When falsy, signaling goes to\n   *  `${apiBase}/v1/agents/:agentId/webrtc/{offer,ice}` (Phase-1 routes,\n   *  local dev). */\n  webrtcGatewayBase?: string\n  onStateChange?: (s: CallState) => void\n  onTranscript?: (entries: TranscriptEntry[]) => void\n  onError?: (e: CallError) => void\n  onEnd?: () => void\n  onInterrupt?: () => void\n  onAgentTurnStart?: () => void\n}\n\nexport async function createWebRtcCall(opts: CreateWebRtcCallOptions): Promise<Call> {\n  const proto: ProtocolState = createProtocolState()\n  let muted = false\n  let ended = false\n\n  const fireState = (next: CallState) => {\n    if (proto.state === next) return\n    proto.state = next\n    opts.onStateChange?.(next)\n  }\n\n  const dispatch = (raw: string) => {\n    handleServerMessage(raw, proto, {\n      onState: fireState,\n      onTranscript: (entries) => opts.onTranscript?.(entries),\n      onError: (err) => opts.onError?.(err),\n      onInterrupt: () => opts.onInterrupt?.(),\n      onAgentTurnStart: () => opts.onAgentTurnStart?.(),\n      onAgentTurnEnd: () => {},\n      onCallEnd: () => teardown(),\n      onConnected: () => {},\n      onClientToolCall: () => {},\n    })\n  }\n\n  fireState('connecting')\n\n  const pc = new RTCPeerConnection({\n    iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n  })\n\n  const audioEl = document.createElement('audio')\n  audioEl.autoplay = true\n  audioEl.style.display = 'none'\n  document.body.appendChild(audioEl)\n  pc.ontrack = (event) => {\n    audioEl.srcObject = event.streams[0] ?? new MediaStream([event.track])\n  }\n\n  let mic: MediaStream\n  try {\n    mic = await navigator.mediaDevices.getUserMedia({ audio: true })\n  } catch (err) {\n    const code =\n      err instanceof DOMException && err.name === 'NotAllowedError'\n        ? 'mic_denied'\n        : 'mic_start_failed'\n    opts.onError?.({\n      code,\n      message: err instanceof Error ? err.message : 'getUserMedia failed',\n    })\n    fireState('error')\n    pc.close()\n    audioEl.remove()\n    throw err\n  }\n  for (const track of mic.getAudioTracks()) pc.addTrack(track, mic)\n\n  const dc = pc.createDataChannel('control', { ordered: true })\n  dc.onmessage = (e) => {\n    if (typeof e.data === 'string') dispatch(e.data)\n  }\n  dc.onerror = () => {\n    opts.onError?.({ code: 'socket_error', message: 'control channel error' })\n  }\n\n  const gateway = opts.webrtcGatewayBase || ''\n  const offerUrl = gateway\n    ? `${gateway}/webrtc/offer?token=${encodeURIComponent(opts.token)}`\n    : `${opts.apiBase}/v1/agents/${encodeURIComponent(opts.agentId)}/webrtc/offer?token=${encodeURIComponent(opts.token)}`\n  const iceUrl = gateway\n    ? `${gateway}/webrtc/ice?token=${encodeURIComponent(opts.token)}`\n    : `${opts.apiBase}/v1/agents/${encodeURIComponent(opts.agentId)}/webrtc/ice?token=${encodeURIComponent(opts.token)}`\n\n  await pc.setLocalDescription(await pc.createOffer())\n\n  let callId: string\n  try {\n    const offerRes = await fetch(offerUrl, {\n      method: 'POST',\n      headers: { 'content-type': 'application/json' },\n      body: JSON.stringify({ sdp: pc.localDescription!.sdp, type: 'offer', agentId: opts.agentId }),\n    })\n    if (!offerRes.ok) {\n      const code = offerRes.status === 401 ? 'unauthorized' : 'server_error'\n      opts.onError?.({ code, message: `signaling failed: HTTP ${offerRes.status}` })\n      fireState('error')\n      mic.getTracks().forEach((t) => t.stop())\n      pc.close()\n      audioEl.remove()\n      throw new Error(`webrtc offer failed: ${offerRes.status}`)\n    }\n    const body = (await offerRes.json()) as { sdp: string; type: 'answer'; callId: string }\n    callId = body.callId\n    await pc.setRemoteDescription({ type: 'answer', sdp: body.sdp })\n  } catch (err) {\n    if (!ended) {\n      opts.onError?.({\n        code: 'network_unreachable',\n        message: err instanceof Error ? err.message : 'signaling failed',\n      })\n      fireState('error')\n      mic.getTracks().forEach((t) => t.stop())\n      pc.close()\n      audioEl.remove()\n    }\n    throw err\n  }\n\n  pc.onicecandidate = (e) => {\n    if (!e.candidate) return\n    void fetch(iceUrl, {\n      method: 'POST',\n      headers: { 'content-type': 'application/json' },\n      body: JSON.stringify({ callId, candidate: e.candidate }),\n    }).catch(() => {\n      /* best-effort */\n    })\n  }\n\n  pc.onconnectionstatechange = () => {\n    const s = pc.connectionState\n    if (s === 'connected') fireState('listening')\n    if (s === 'failed' || s === 'disconnected') {\n      opts.onError?.({ code: 'socket_error', message: `webrtc connection ${s}` })\n      teardown()\n    }\n    if (s === 'closed' && !ended) teardown()\n  }\n\n  const teardown = () => {\n    if (ended) return\n    ended = true\n    try {\n      mic.getTracks().forEach((t) => t.stop())\n    } catch {\n      /* already stopped */\n    }\n    try {\n      pc.close()\n    } catch {\n      /* already closed */\n    }\n    try {\n      audioEl.remove()\n    } catch {\n      /* already removed */\n    }\n    fireState('ended')\n    opts.onEnd?.()\n  }\n\n  return {\n    get state() {\n      return proto.state\n    },\n    get transcript() {\n      return proto.transcript.slice()\n    },\n    get 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+{"version":3,"sources":["../src/browser.ts","../src/config.ts","../src/worklets/mic-downsampler.worklet.js","../src/AudioCapture.ts","../src/AudioPlayback.ts","../src/ReconnectingWebSocket.ts","../src/protocol.ts","../src/clientTools.ts","../src/ClientMarksBuffer.ts","../src/VoiceClient.ts","../src/webrtc/createWebRtcCall.ts"],"sourcesContent":["// Browser entry — `import { configureVoiceClient } from '@craftedxp/voice-js'`\n//\n// Pre-injects:\n//   - native `WebSocket` (globalThis.WebSocket)\n//   - browser audio adapter (AudioContext + AudioWorklet + getUserMedia)\n//\n// Electron renderer + browser tabs both pick this entry via the\n// `browser` condition in package.json `exports`.\n\nimport {\n  mergeStartCallContext,\n  normalizeConfig,\n  type Call,\n  type FetchTokenResult,\n  type StartCallOptions,\n  type VoiceClientConfig,\n  type VoiceClientFactory,\n} from './config'\nimport type { WebSocketFactory, WebSocketLike } from './ReconnectingWebSocket'\nimport { BrowserVoiceClient } from './VoiceClient'\nimport { createWebRtcCall } from './webrtc/createWebRtcCall'\n\n// Adapter from native browser WebSocket → our WebSocketLike contract.\n// One-line wrapper; the shapes already match.\nconst browserWsFactory: WebSocketFactory = (url) =>\n  new (globalThis as unknown as { WebSocket: new (u: string) => WebSocketLike }).WebSocket(url)\n\nclass BrowserVoiceFactory implements VoiceClientFactory {\n  readonly config: VoiceClientConfig\n\n  constructor(config: VoiceClientConfig) {\n    this.config = config\n  }\n\n  startCall = async (options: StartCallOptions): Promise<Call> => {\n    if (!options.agentId) {\n      throw new Error('startCall: agentId is required')\n    }\n\n    const { context, metadata } = mergeStartCallContext(this.config, options)\n    const fetchArgs = {\n      agentId: options.agentId,\n      userId: options.userId,\n      context,\n      metadata,\n    }\n\n    // Resolve the token + transport. Three input shapes:\n    //   1. options.token (test escape hatch) — always WS, current behaviour\n    //   2. fetchToken returns a bare string — always WS (back-compat)\n    //   3. fetchToken returns FetchTokenResult — server-chosen transport\n    let resolved: FetchTokenResult\n    if (options.token) {\n      resolved = { token: options.token, transport: 'ws' }\n    } else {\n      const r = await this.config.fetchToken(fetchArgs)\n      if (!r) {\n        throw new Error('configureVoiceClient.fetchToken returned empty token')\n      }\n      resolved = typeof r === 'string' ? { token: r, transport: 'ws' } : r\n      if (!resolved.token) {\n        throw new Error('configureVoiceClient.fetchToken returned an object without `token`')\n      }\n    }\n\n    if (resolved.transport === 'webrtc') {\n      return createWebRtcCall({\n        agentId: options.agentId,\n        apiBase: this.config.apiBase,\n        token: resolved.token,\n        webrtcGatewayBase: resolved.webrtcGatewayBase,\n        onStateChange: options.onStateChange,\n        onTranscript: options.onTranscript,\n        onError: options.onError,\n        // Synthesise a minimal CallEndEvent. WebRTC doesn't carry an end reason\n        // from the server yet — use 'agent_ended' as placeholder. durationMs is\n        // tracked at 0 until the followup lands (see spec Followups section).\n        onEnd: options.onEnd\n          ? () => options.onEnd!({ reason: 'agent_ended', durationMs: 0 })\n          : undefined,\n        onInterrupt: options.onInterrupt,\n        onAgentTurnStart: options.onAgentTurnStart,\n        clientTools: options.clientTools,\n      })\n    }\n\n    const client = new BrowserVoiceClient({\n      config: this.config,\n      // Carry merged context/metadata through to startCall so server can\n      // see what the SDK saw.\n      options: { ...options, context, metadata },\n      token: resolved.token,\n      wsFactory: browserWsFactory,\n    })\n    await client.start()\n    return client\n  }\n}\n\n/**\n * One-time SDK setup. Returns a factory you call `startCall` on for\n * every voice call.\n *\n * Example:\n *   const voice = configureVoiceClient({\n *     apiBase: 'https://api.your-server.com',\n *     fetchToken: async ({ agentId }) => {\n *       const r = await fetch('/api/voice-token', {\n *         method: 'POST',\n *         body: JSON.stringify({ agentId }),\n *       })\n *       return (await r.json()).token\n *     },\n *   })\n *\n *   // Per call (typically inside a click handler):\n *   const call = await voice.startCall({\n *     agentId: 'agt_xxx',\n *     onTranscript: (entries) => render(entries),\n *     onEnd: ({ reason }) => log(reason),\n *   })\n *   call.mute()\n *   call.end()\n */\nexport function configureVoiceClient(config: VoiceClientConfig): VoiceClientFactory {\n  return new BrowserVoiceFactory(normalizeConfig(config))\n}\n\n// ---------------------------------------------------------------------------\n// Public re-exports — types + advanced primitives.\n// ---------------------------------------------------------------------------\n\nexport type {\n  Call,\n  ClientTool,\n  ClientToolMap,\n  FetchToken,\n  FetchTokenArgs,\n  FetchTokenResult,\n  StartCallOptions,\n  VoiceClientConfig,\n  VoiceClientFactory,\n} from './config'\n\nexport type {\n  CallEndEvent,\n  CallEndReason,\n  CallError,\n  CallErrorCode,\n  CallState,\n  TranscriptEntry,\n  VolumeEvent,\n} from './protocol'\n\n// Low-level primitives — exposed for advanced consumers who want to\n// drive just the mic, just the playback, or build a custom transport on\n// top of the protocol decoder. Most users should stick to\n// `configureVoiceClient().startCall()`.\nexport { createAudioCapture } from './AudioCapture'\nexport type { CaptureController, CaptureOptions, OnChunk, OnError, OnVolume } from './AudioCapture'\n\nexport { createAudioPlayback } from './AudioPlayback'\nexport type { PlaybackController, PlaybackOptions, OnAgentSpeakingChange } from './AudioPlayback'\n\nexport { createReconnectingWebSocket } from './ReconnectingWebSocket'\nexport type {\n  ReconnectingWebSocket,\n  RWSEvent,\n  RWSOptions,\n  WebSocketFactory,\n  WebSocketLike,\n} from './ReconnectingWebSocket'\n\nexport { handleServerMessage, createProtocolState, buildWsUrl } from './protocol'\nexport type { ProtocolState, ProtocolCallbacks, ServerMessage } from './protocol'\n","// Public configuration surface.\n//\n// `configureVoiceClient` returns a small factory that knows how to mint\n// `ct_` tokens via your callback. Per-call you call `factory.startCall`.\n//\n// Parity with @craftedxp/voice-rn:\n//   voice-rn calls `configureVoiceClient` once at app startup as a\n//   side-effect (singleton), because in RN there is exactly one host\n//   app and one running `<AgentCall>` at a time. In JS environments\n//   the same process can drive multiple clients (multi-tenant\n//   dashboards, terminal multiplexers, electron apps with several\n//   panels), so the JS SDK returns the factory rather than mutating a\n//   module-level singleton. Same option names, same callback shape.\n//\n// Auth model:\n//   - `apiKey` is INTENTIONALLY ABSENT from the public surface.\n//     Pre-0.2 had it; baking an `sk_` into a JS bundle ships your\n//     server-grade credentials to every client. The right pattern is\n//     `fetchToken` — your code asks YOUR backend for a short-lived\n//     `ct_` and your backend uses its `sk_` (via @craftedxp/sdk-node)\n//     to mint it.\n//   - For tests + local prototypes a bare `token` may be passed to\n//     `startCall` directly, bypassing `fetchToken`. Don't ship that to\n//     production — `ct_` lifetimes are short and you want the SDK to\n//     re-mint on expiry.\n\nimport type { CallEndEvent, CallError, CallState, TranscriptEntry, VolumeEvent } from './protocol'\nimport type { ClientToolMap } from './clientTools'\n\n// ---------------------------------------------------------------------------\n// fetchToken contract — matches voice-rn FetchTokenArgs verbatim.\n// ---------------------------------------------------------------------------\n\nexport interface FetchTokenArgs {\n  /** The agent the SDK is about to call. */\n  agentId: string\n  /**\n   * Optional consumer-side user identifier. Round-tripped to the server\n   * as `contactId` for Phase 11 contact memory. The SDK does not\n   * inspect this; your backend uses it to scope the token mint.\n   */\n  userId?: string\n  /**\n   * Per-call structured context lowered into the agent's effective\n   * system prompt server-side at session open. Opaque to the SDK.\n   */\n  context?: Record<string, unknown>\n  /**\n   * String key/value pairs round-tripped on the `call.ended` webhook.\n   * Capped at 1 KB total server-side. NOT lowered into the system prompt.\n   */\n  metadata?: Record<string, string>\n}\n\n/**\n * What `fetchToken` may return. The rich object form lets the server\n * choose the transport per call. Returning a bare string is backwards-\n * compatible — the SDK treats it as `{ token, transport: 'ws' }`.\n */\nexport interface FetchTokenResult {\n  /** Raw `ct_` to feed into the WS open / WebRTC offer. */\n  token: string\n  /** Server-selected transport. Default `'ws'` if absent. */\n  transport?: 'ws' | 'webrtc'\n  /** Required when `transport === 'webrtc'` AND the server uses a\n   *  separate signaling gateway. When omitted on a webrtc result, the\n   *  SDK falls back to the API base's Phase-1 routes (local dev). */\n  webrtcGatewayBase?: string\n}\n\nexport type FetchToken = (args: FetchTokenArgs) => Promise<string | FetchTokenResult>\n\n// ---------------------------------------------------------------------------\n// Factory configuration\n// ---------------------------------------------------------------------------\n\nexport interface VoiceClientConfig {\n  /**\n   * Full HTTPS URL of the Voissia server. The WebSocket scheme is\n   * derived: `https` → `wss`, `http` → `ws`. No trailing slash needed.\n   */\n  apiBase: string\n  /**\n   * Called by the SDK whenever it needs a fresh `ct_` token (initial\n   * connect; mid-call refresh on `token_expired`). Your implementation\n   * should hit YOUR backend, which holds the `sk_` API key and mints\n   * via `POST /v1/call-tokens` (or `client.callTokens.mint` from\n   * @craftedxp/sdk-node). Never embed `sk_` in JS code that ships to a\n   * client.\n   */\n  fetchToken: FetchToken\n  /**\n   * Optional metadata applied to EVERY startCall. Per-call `metadata`\n   * in `startCall` is merged on top (per-call wins on key conflicts).\n   * Useful for dashboard-wide tags like `{ surface: 'web', appVersion }`.\n   */\n  defaultMetadata?: Record<string, string>\n  /**\n   * Optional context applied to EVERY startCall. Per-call `context` in\n   * `startCall` is merged on top. Useful for cross-call invariants like\n   * the signed-in user's locale.\n   */\n  defaultContext?: Record<string, unknown>\n}\n\n// ---------------------------------------------------------------------------\n// Per-call options — startCall({ ... })\n// ---------------------------------------------------------------------------\n\nexport interface StartCallOptions {\n  /** The agent to call. */\n  agentId: string\n  /** Per-call user identifier. Round-tripped to fetchToken as `userId`. */\n  userId?: string\n  /**\n   * Per-call structured context. Merged on top of `defaultContext`\n   * configured at factory time.\n   */\n  context?: Record<string, unknown>\n  /**\n   * Per-call metadata. Merged on top of `defaultMetadata` configured\n   * at factory time.\n   */\n  metadata?: Record<string, string>\n  /**\n   * When false, the SDK + server stay full-duplex but barge-in is\n   * suppressed. Useful for alarm-style flows where the user shouldn't\n   * accidentally interrupt the script. Default true.\n   */\n  bargeIn?: boolean\n  /**\n   * Client-side tools the agent's LLM can call mid-conversation. Each\n   * tool's handler runs on the consumer's side; result is fed back to\n   * the LLM through the existing call WebSocket. Schema and handler\n   * colocate. Validated synchronously at startCall — bad input throws.\n   *\n   * See docs/integration-echocheck.md for the wire protocol and the\n   * server-side guarantees.\n   */\n  clientTools?: ClientToolMap\n  /**\n   * Test-only escape hatch — pass a pre-minted `ct_` directly and skip\n   * the `fetchToken` call. Don't use this in production code: tokens\n   * expire and the SDK can't re-mint without the callback.\n   */\n  token?: string\n\n  // Event callbacks — same shape as voice-rn UseVoiceCallOptions.\n  onStateChange?: (state: CallState) => void\n  onTranscript?: (entries: TranscriptEntry[]) => void\n  onError?: (err: CallError) => void\n  onEnd?: (end: CallEndEvent) => void\n  /** Volume-meter event for VU UIs. ~10 Hz cadence (browser bundle only). */\n  onVolume?: (vol: VolumeEvent) => void\n  /**\n   * Fires when the server signals barge-in (the user started talking\n   * mid-agent-turn). The browser bundle automatically flushes its\n   * built-in audio playback before this callback runs; the callback is\n   * fired regardless. Node / Electron consumers with custom playback\n   * should drain their audio queue here so the agent goes silent\n   * immediately.\n   */\n  onInterrupt?: () => void\n  /**\n   * Fires on `agent_turn_start` — the server has begun a new agent\n   * turn. The state-machine transition to `agent_speaking` happens at\n   * the same moment via `onStateChange`; use this when you want a\n   * precise turn anchor (e.g. \"agent has been speaking for N ms\" UIs)\n   * without diffing state.\n   */\n  onAgentTurnStart?: () => void\n}\n\n// ---------------------------------------------------------------------------\n// Call handle returned from startCall — matches voice-rn\n// VoiceCallController where it makes sense.\n// ---------------------------------------------------------------------------\n\nexport interface Call {\n  /** Current state. Snapshot — subscribe via onStateChange for live updates. */\n  readonly state: CallState\n  /** Full transcript so far. Snapshot — subscribe via onTranscript for live updates. */\n  readonly transcript: TranscriptEntry[]\n  /** True after `mute()` and before `unmute()`. */\n  readonly isMuted: boolean\n  /** End the call locally. Closes the WS, stops the mic, fires onEnd. Idempotent. */\n  end: () => void\n  /** Mute mic frames. Wire stays active so server endpointing doesn't false-positive. Idempotent. */\n  mute: () => void\n  /** Unmute mic frames. Idempotent. */\n  unmute: () => void\n}\n\n// ---------------------------------------------------------------------------\n// Factory contract — what configureVoiceClient returns. The actual\n// implementation differs between browser (audio-equipped) and node (raw\n// PCM) bundles; both satisfy this interface.\n// ---------------------------------------------------------------------------\n\nexport interface VoiceClientFactory {\n  /** Read back the resolved config (post trailing-slash normalisation). */\n  readonly config: VoiceClientConfig\n  /**\n   * Open a fresh call. Returns when the WS is open; rejects on\n   * pre-flight failure (missing config, fetchToken throw, etc). Mid-\n   * call failures arrive via the per-call `onError` callback — they\n   * don't reject this promise.\n   */\n  startCall: (options: StartCallOptions) => Promise<Call>\n}\n\n// ---------------------------------------------------------------------------\n// Pre-flight validation. Pulled out so both bundles share the exact\n// same \"missing field\" error messages.\n// ---------------------------------------------------------------------------\n\nexport function normalizeConfig(config: VoiceClientConfig): VoiceClientConfig {\n  if (!config) throw new Error('configureVoiceClient: config is required')\n  if ('apiKey' in (config as object)) {\n    throw new Error(\n      'configureVoiceClient: `apiKey` is no longer supported. Embedding sk_ in JS code ships server-grade credentials to every client. Pass `fetchToken: async ({ agentId }) => { /* call YOUR backend mint */ }` instead — see the @craftedxp/voice-js README for the migration recipe.',\n    )\n  }\n  if (!config.apiBase) {\n    throw new Error('configureVoiceClient: apiBase is required')\n  }\n  if (typeof config.fetchToken !== 'function') {\n    throw new Error('configureVoiceClient: fetchToken must be a function')\n  }\n  return {\n    ...config,\n    apiBase: config.apiBase.replace(/\\/+$/, ''),\n  }\n}\n\n// Merge factory-level defaults with per-call overrides. Per-call wins.\nexport function mergeStartCallContext(\n  factory: VoiceClientConfig,\n  call: StartCallOptions,\n): { context?: Record<string, unknown>; metadata?: Record<string, string> } {\n  const context =\n    factory.defaultContext || call.context\n      ? { ...(factory.defaultContext ?? {}), ...(call.context ?? {}) }\n      : undefined\n  const metadata =\n    factory.defaultMetadata || call.metadata\n      ? { ...(factory.defaultMetadata ?? {}), ...(call.metadata ?? {}) }\n      : undefined\n  return { context, metadata }\n}\n\nexport type { ClientTool, ClientToolMap } from './clientTools'\n","// AudioWorklet — runs off the main thread in the audio rendering graph.\n//\n// Input: Float32 samples at the AudioContext's native sampleRate (typically\n// 48000 Hz on desktop, 44100 Hz on some iOS devices).\n// Output: 16 kHz mono Int16 PCM, shipped to the main thread via\n// `port.postMessage(ArrayBuffer, [ArrayBuffer])` (transferred, not copied).\n//\n// Why AudioWorklet instead of ScriptProcessorNode: ScriptProcessorNode is\n// deprecated + main-thread-bound, so any JS jank produces audible audio\n// glitches. AudioWorklet's `process()` runs on the audio rendering thread\n// at the graph's block cadence (128 frames by default) and backpressures\n// via returning `true` / `false`.\n//\n// This file is loaded as text (see tsup.config.ts loader) and registered\n// at runtime via `audioWorklet.addModule(blobUrl)`.\n\nclass MicDownsampler extends AudioWorkletProcessor {\n  constructor() {\n    super()\n    // Target sample rate for STT. Matches Deepgram Nova-3 + the platform's\n    // server-side SAMPLE_RATE constant in AgentCallHandler.\n    this.targetRate = 16000\n    // Accumulator for the downsample. We collect incoming samples and emit\n    // an Int16 chunk when we've accumulated ~1024 target-rate samples\n    // (~64 ms at 16 kHz) — matches the mobile SDK's chunk size so both\n    // platforms have the same server-side framing.\n    this.outputFrames = 1024\n    this.acc = []\n    // Running index used for fractional resampling.\n    this.readCursor = 0\n  }\n\n  // `inputs[0][0]` = first channel of first input. 128 Float32 samples per\n  // call at the context's sampleRate. Return true = keep processing.\n  process(inputs) {\n    const input = inputs[0]\n    if (!input || input.length === 0) return true\n    const channel = input[0]\n    if (!channel || channel.length === 0) return true\n\n    const ctxRate = sampleRate // global inside AudioWorkletProcessor\n    const ratio = ctxRate / this.targetRate\n\n    // Simple linear-interp downsample. For 48000 → 16000 that's 3:1, which\n    // linear handles fine for voice. Anti-alias filtering would be\n    // theoretically better but inaudible for speech.\n    for (let i = 0; i < channel.length; i++) {\n      this.acc.push(channel[i])\n    }\n\n    while (this.acc.length - this.readCursor >= ratio * this.outputFrames) {\n      const out = new Int16Array(this.outputFrames)\n      let readIdx = this.readCursor\n      for (let i = 0; i < this.outputFrames; i++) {\n        // Linear interp between floor(readIdx) and ceil(readIdx)\n        const low = Math.floor(readIdx)\n        const high = Math.min(low + 1, this.acc.length - 1)\n        const frac = readIdx - low\n        const sample = this.acc[low] * (1 - frac) + this.acc[high] * frac\n        // Clip + convert to int16\n        const clipped = Math.max(-1, Math.min(1, sample))\n        out[i] = clipped < 0 ? clipped * 0x8000 : clipped * 0x7fff\n        readIdx += ratio\n      }\n      // Transfer the ArrayBuffer (zero-copy) to the main thread.\n      this.port.postMessage(out.buffer, [out.buffer])\n      this.readCursor = readIdx\n    }\n\n    // Garbage-collect the consumed portion of `acc` every so often so it\n    // doesn't grow without bound. Leave ~one chunk of headroom.\n    if (this.readCursor > ratio * this.outputFrames) {\n      this.acc = this.acc.slice(Math.floor(this.readCursor))\n      this.readCursor -= Math.floor(this.readCursor)\n    }\n\n    return true\n  }\n}\n\nregisterProcessor('mic-downsampler', MicDownsampler)\n","// Mic capture pipeline:\n//   getUserMedia → AudioContext → AudioWorkletNode → Int16 PCM chunks\n//\n// We register the worklet from an inline Blob URL so consumers don't need\n// to host or bundle a separate .js file. The worklet source is loaded as\n// text at build time via tsup's custom loader (see tsup.config.ts).\n\n// eslint-disable-next-line @typescript-eslint/ban-ts-comment\n// @ts-ignore — text-loaded by tsup\nimport workletSource from './worklets/mic-downsampler.worklet.js'\n\nexport type OnChunk = (pcm: ArrayBuffer) => void\nexport type OnVolume = (rms01: number) => void\nexport type OnError = (err: Error) => void\n\nexport interface CaptureOptions {\n  onChunk: OnChunk\n  // Optional — when set we compute input RMS and call this at ~10Hz. Used\n  // for volume visualisers in consumer UIs.\n  onVolume?: OnVolume\n  onError?: OnError\n}\n\nexport interface CaptureController {\n  start: () => Promise<void>\n  stop: () => void\n  mute: (muted: boolean) => void\n  isCapturing: () => boolean\n}\n\n// Volume-meter cadence. 100ms = 10Hz updates; smoother than per-chunk\n// (16Hz) without flicker.\nconst VOLUME_INTERVAL_MS = 100\n\nexport const createAudioCapture = (options: CaptureOptions): CaptureController => {\n  let audioContext: AudioContext | null = null\n  let mediaStream: MediaStream | null = null\n  let sourceNode: MediaStreamAudioSourceNode | null = null\n  let workletNode: AudioWorkletNode | null = null\n  let analyser: AnalyserNode | null = null\n  let volumeTimer: ReturnType<typeof setInterval> | null = null\n  let muted = false\n  let capturing = false\n\n  const computeRms = (buf: Float32Array): number => {\n    let sum = 0\n    for (let i = 0; i < buf.length; i++) sum += buf[i] * buf[i]\n    const rms = Math.sqrt(sum / buf.length)\n    // Clamp to 0..1 — RMS of a full-scale signal is 0.707, so boost a bit\n    // so \"normal speech\" hits ~0.5 in the visualiser.\n    return Math.min(1, rms * 1.8)\n  }\n\n  const start = async () => {\n    if (capturing) return\n    try {\n      // Browsers only grant getUserMedia on secure contexts (https or\n      // localhost). The thrown DOMException is clear enough — we let it\n      // propagate with a wrapped error message for clarity.\n      mediaStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          // Hand tuning for voice agent use: we want the raw signal so the\n          // server-side STT can do its own noise handling. Disable browser\n          // AEC/AGC/NR — experimentally they fight with whatever processing\n          // the TTS playback path feeds back in over speakers.\n          echoCancellation: true,\n          noiseSuppression: true,\n          autoGainControl: true,\n          channelCount: 1,\n        },\n      })\n\n      audioContext = new AudioContext()\n      // Most browsers suspend AudioContext until a user gesture. connect()\n      // is idempotent; if we were constructed inside a click/touch handler\n      // it's already running.\n      if (audioContext.state === 'suspended') await audioContext.resume()\n\n      // Load + register the worklet from an inline Blob URL. One-time per\n      // context; subsequent `new AudioWorkletNode` calls are cheap.\n      const blob = new Blob([workletSource], { type: 'application/javascript' })\n      const url = URL.createObjectURL(blob)\n      try {\n        await audioContext.audioWorklet.addModule(url)\n      } finally {\n        URL.revokeObjectURL(url)\n      }\n\n      sourceNode = audioContext.createMediaStreamSource(mediaStream)\n      workletNode = new AudioWorkletNode(audioContext, 'mic-downsampler')\n\n      // Worklet → main thread: Int16 PCM chunks\n      workletNode.port.onmessage = (event: MessageEvent<ArrayBuffer>) => {\n        if (muted) return\n        options.onChunk(event.data)\n      }\n\n      // Analyser sits in parallel to the worklet for the volume meter. Not\n      // strictly necessary — we could compute RMS inside the worklet and\n      // post it too — but AnalyserNode avoids extra bridge chatter.\n      if (options.onVolume) {\n        analyser = audioContext.createAnalyser()\n        analyser.fftSize = 256\n        sourceNode.connect(analyser)\n        const buf = new Float32Array(analyser.fftSize)\n        volumeTimer = setInterval(() => {\n          if (!analyser) return\n          analyser.getFloatTimeDomainData(buf)\n          options.onVolume?.(computeRms(buf))\n        }, VOLUME_INTERVAL_MS)\n      }\n\n      sourceNode.connect(workletNode)\n      // The worklet doesn't need to feed into the destination — its only\n      // job is to emit messages. But the AudioContext graph keeps the\n      // worklet running only if it's connected somewhere; connect to a\n      // zero-gain node to park it.\n      const sink = audioContext.createGain()\n      sink.gain.value = 0\n      workletNode.connect(sink).connect(audioContext.destination)\n\n      capturing = true\n    } catch (err) {\n      const wrapped =\n        err instanceof Error ? err : new Error(typeof err === 'string' ? err : 'capture failed')\n      options.onError?.(wrapped)\n      throw wrapped\n    }\n  }\n\n  const stop = () => {\n    if (!capturing) return\n    capturing = false\n    if (volumeTimer) {\n      clearInterval(volumeTimer)\n      volumeTimer = null\n    }\n    try {\n      workletNode?.disconnect()\n      analyser?.disconnect()\n      sourceNode?.disconnect()\n    } catch {\n      // already disconnected\n    }\n    workletNode = null\n    analyser = null\n    sourceNode = null\n    if (mediaStream) {\n      for (const track of mediaStream.getTracks()) track.stop()\n      mediaStream = null\n    }\n    if (audioContext && audioContext.state !== 'closed') {\n      void audioContext.close().catch(() => undefined)\n    }\n    audioContext = null\n  }\n\n  return {\n    start,\n    stop,\n    mute: (v) => {\n      muted = v\n    },\n    isCapturing: () => capturing,\n  }\n}\n","// Playback pipeline:\n//   WebSocket binary frame (Int16 PCM @ 16 kHz) → AudioBufferSourceNode\n//   chained for gapless playback → AudioContext destination.\n//\n// We deliberately don't use MediaSource or a single streaming source —\n// each agent TTS chunk is scheduled as its own short AudioBuffer so we\n// can interrupt (flush the queue) the moment a barge-in lands. One chunk\n// ≈ 20–100ms of audio.\n\nexport type OnVolume = (rms01: number) => void\nexport type OnAgentSpeakingChange = (speaking: boolean) => void\n\nexport interface PlaybackOptions {\n  // Sample rate of the incoming PCM — matches AgentCallHandler.SAMPLE_RATE.\n  sampleRate?: number\n  onVolume?: OnVolume\n  onSpeakingChange?: OnAgentSpeakingChange\n}\n\nexport interface PlaybackController {\n  enqueue: (pcm: ArrayBuffer) => void\n  flush: () => void\n  close: () => void\n  resume: () => Promise<void>\n}\n\nconst DEFAULT_SAMPLE_RATE = 16_000\nconst VOLUME_INTERVAL_MS = 100\n\nexport const createAudioPlayback = (options: PlaybackOptions = {}): PlaybackController => {\n  const sampleRate = options.sampleRate ?? DEFAULT_SAMPLE_RATE\n  let audioContext: AudioContext | null = null\n  let gainNode: GainNode | null = null\n  let analyser: AnalyserNode | null = null\n  let volumeTimer: ReturnType<typeof setInterval> | null = null\n  let nextStartTime = 0\n  let scheduledNodes: AudioBufferSourceNode[] = []\n  let speaking = false\n\n  const ensureContext = async () => {\n    if (audioContext) {\n      if (audioContext.state === 'suspended') await audioContext.resume()\n      return\n    }\n    // Pass the target sample rate as a hint. Browsers may choose a\n    // different rate (48000 is most common); `createBuffer(numChannels,\n    // length, sampleRate)` with a different rate than the context will\n    // trigger resampling on `start()`, which the browser handles natively.\n    audioContext = new AudioContext({ sampleRate })\n    gainNode = audioContext.createGain()\n    if (options.onVolume) {\n      analyser = audioContext.createAnalyser()\n      analyser.fftSize = 256\n      gainNode.connect(analyser)\n      const buf = new Float32Array(analyser.fftSize)\n      volumeTimer = setInterval(() => {\n        if (!analyser) return\n        analyser.getFloatTimeDomainData(buf)\n        let sum = 0\n        for (let i = 0; i < buf.length; i++) sum += buf[i] * buf[i]\n        const rms = Math.sqrt(sum / buf.length)\n        options.onVolume?.(Math.min(1, rms * 1.8))\n      }, VOLUME_INTERVAL_MS)\n    }\n    gainNode.connect(audioContext.destination)\n    nextStartTime = audioContext.currentTime\n  }\n\n  const setSpeaking = (v: boolean) => {\n    if (v === speaking) return\n    speaking = v\n    options.onSpeakingChange?.(v)\n  }\n\n  const pruneFinished = () => {\n    const now = audioContext?.currentTime ?? 0\n    scheduledNodes = scheduledNodes.filter((n) => {\n      const node = n as AudioBufferSourceNode & { _endsAt?: number }\n      return (node._endsAt ?? 0) > now\n    })\n    if (scheduledNodes.length === 0) setSpeaking(false)\n  }\n\n  const enqueue = (pcm: ArrayBuffer) => {\n    // We don't want to await here — WS onmessage is hot-path, and AudioContext\n    // resume is cheap once warmed. `void` the promise and proceed on next\n    // tick; if the context isn't ready, the first chunk may land a hair\n    // late but subsequent chunks catch up.\n    if (!audioContext) {\n      void ensureContext().then(() => enqueue(pcm))\n      return\n    }\n    if (!audioContext || !gainNode) return\n\n    const int16 = new Int16Array(pcm)\n    if (int16.length === 0) return\n\n    const audioBuffer = audioContext.createBuffer(1, int16.length, sampleRate)\n    const float32 = audioBuffer.getChannelData(0)\n    for (let i = 0; i < int16.length; i++) {\n      float32[i] = int16[i] / 0x8000\n    }\n\n    const node = audioContext.createBufferSource()\n    node.buffer = audioBuffer\n    node.connect(gainNode)\n\n    const now = audioContext.currentTime\n    // Schedule back-to-back. If we've fallen behind (mic lag, WS gap),\n    // re-anchor to now so we don't skew the whole queue further.\n    const startAt = Math.max(now, nextStartTime)\n    node.start(startAt)\n\n    const duration = int16.length / sampleRate\n    ;(node as AudioBufferSourceNode & { _endsAt?: number })._endsAt = startAt + duration\n    nextStartTime = startAt + duration\n\n    scheduledNodes.push(node)\n    setSpeaking(true)\n\n    // Prune after every enqueue so `speaking` collapses quickly when the\n    // queue drains. Cheap O(n) over a short list.\n    node.onended = () => pruneFinished()\n  }\n\n  // Barge-in: stop anything pending, reset the schedule pointer.\n  //\n  // We can't simply `.stop()` playing sources mid-buffer and expect the\n  // AudioContext's destination to be silent immediately — Safari in\n  // particular sometimes plays out a few ms of tail. Reliable approach:\n  // swap the gainNode out so nothing after this point is audible, then\n  // clean up the old one on the next tick.\n  const flush = () => {\n    if (!audioContext || !gainNode) return\n    for (const node of scheduledNodes) {\n      try {\n        node.stop()\n      } catch {\n        // already stopped\n      }\n    }\n    scheduledNodes = []\n    // Swap gainNode so even stubborn trailing samples on the old node\n    // don't reach the speakers.\n    gainNode.disconnect()\n    gainNode = audioContext.createGain()\n    if (analyser) {\n      analyser.disconnect()\n      gainNode.connect(analyser)\n    }\n    gainNode.connect(audioContext.destination)\n    nextStartTime = audioContext.currentTime\n    setSpeaking(false)\n  }\n\n  const close = () => {\n    flush()\n    if (volumeTimer) {\n      clearInterval(volumeTimer)\n      volumeTimer = null\n    }\n    if (audioContext && audioContext.state !== 'closed') {\n      void audioContext.close().catch(() => undefined)\n    }\n    audioContext = null\n    gainNode = null\n    analyser = null\n  }\n\n  const resume = async () => {\n    await ensureContext()\n  }\n\n  return { enqueue, flush, close, resume }\n}\n","// Minimal auto-reconnecting WebSocket wrapper.\n//\n// Scope: transparent reconnection on unexpected drops (network blip,\n// server restart). We deliberately do NOT try to preserve call state\n// across reconnects — the server's AgentCallHandler is session-scoped\n// and won't resume where we left off. If the consumer needs mid-call\n// resilience, that's a server-side resumable-session feature, not a\n// client-side retry.\n//\n// In v1 we reconnect only when `connect()` has been called and\n// `.close(1000, ...)` has NOT been called by the consumer (i.e., we drop\n// because of network, not because the user hung up). On unclean close\n// the consumer gets a fresh connection + a `reconnected` event so they\n// can re-issue any greeting / context.\n//\n// Transport agnostic — accepts a WebSocket-constructor factory so the\n// browser bundle uses native `WebSocket` and the node bundle injects the\n// `ws` package.\n\nexport type RWSEvent =\n  | { type: 'open' }\n  | { type: 'reconnected' }\n  | { type: 'message'; data: string | ArrayBuffer }\n  | { type: 'close'; code: number; reason: string; permanent: boolean }\n  | { type: 'error'; error: Error }\n\n// Minimal WebSocket-like contract. Both browser `WebSocket` and the\n// `ws` package's WebSocket satisfy it.\nexport interface WebSocketLike {\n  binaryType: string\n  readyState: number\n  onopen: ((ev: unknown) => void) | null\n  onmessage: ((ev: { data: string | ArrayBuffer }) => void) | null\n  onerror: ((ev: unknown) => void) | null\n  onclose: ((ev: { code: number; reason: string }) => void) | null\n  send: (data: string | ArrayBuffer | ArrayBufferView) => void\n  close: (code?: number, reason?: string) => void\n}\n\nexport type WebSocketFactory = (url: string) => WebSocketLike\n\nexport interface RWSOptions {\n  url: string\n  // Factory so we can swap browser-native WebSocket for the `ws` package\n  // in node. Browser entry pre-fills with a wrapper around globalThis.WebSocket.\n  wsFactory: WebSocketFactory\n  // Cap the number of auto-reconnect attempts. `0` disables retry entirely\n  // (closer to native WebSocket semantics). Default: 3.\n  maxRetries?: number\n  // Initial backoff; doubles up to maxBackoffMs.\n  initialBackoffMs?: number\n  maxBackoffMs?: number\n}\n\nconst READYSTATE_OPEN = 1\nconst READYSTATE_CLOSED = 3\n\nexport const createReconnectingWebSocket = (\n  options: RWSOptions,\n  onEvent: (ev: RWSEvent) => void,\n) => {\n  const maxRetries = options.maxRetries ?? 3\n  const initialBackoff = options.initialBackoffMs ?? 500\n  const maxBackoff = options.maxBackoffMs ?? 8000\n\n  let ws: WebSocketLike | null = null\n  let intentionalClose = false\n  let retries = 0\n  let backoff = initialBackoff\n  let reconnectTimer: ReturnType<typeof setTimeout> | null = null\n\n  const openOnce = () => {\n    ws = options.wsFactory(options.url)\n    // Binary frames arrive as Int16 PCM from the server. ArrayBuffer is\n    // more convenient than Blob (one fewer async read) — we `.arrayBuffer()`\n    // a Blob manually only if we receive one unexpectedly.\n    ws.binaryType = 'arraybuffer'\n    ws.onopen = () => {\n      if (retries === 0) onEvent({ type: 'open' })\n      else onEvent({ type: 'reconnected' })\n      retries = 0\n      backoff = initialBackoff\n    }\n    ws.onmessage = (ev) => {\n      onEvent({ type: 'message', data: ev.data as string | ArrayBuffer })\n    }\n    ws.onerror = () => {\n      onEvent({ type: 'error', error: new Error('WebSocket error') })\n    }\n    ws.onclose = (ev) => {\n      ws = null\n      const shouldRetry = !intentionalClose && retries < maxRetries\n      if (!shouldRetry) {\n        onEvent({\n          type: 'close',\n          code: ev.code,\n          reason: ev.reason,\n          permanent: true,\n        })\n        return\n      }\n      onEvent({\n        type: 'close',\n        code: ev.code,\n        reason: ev.reason,\n        permanent: false,\n      })\n      retries++\n      const delay = Math.min(backoff, maxBackoff)\n      backoff = Math.min(backoff * 2, maxBackoff)\n      reconnectTimer = setTimeout(openOnce, delay)\n    }\n  }\n\n  openOnce()\n\n  return {\n    send: (data: string | ArrayBuffer | ArrayBufferView) => {\n      if (ws && ws.readyState === READYSTATE_OPEN) ws.send(data)\n    },\n    close: (code = 1000, reason = 'client-requested') => {\n      intentionalClose = true\n      if (reconnectTimer) {\n        clearTimeout(reconnectTimer)\n        reconnectTimer = null\n      }\n      try {\n        ws?.close(code, reason)\n      } catch {\n        // already closed\n      }\n    },\n    readyState: () => ws?.readyState ?? READYSTATE_CLOSED,\n  }\n}\n\nexport type ReconnectingWebSocket = ReturnType<typeof createReconnectingWebSocket>\n","// Wire-protocol types + stateless transcript reducer.\n//\n// Shared between the browser and node bundles. The protocol surface is\n// matched (where it makes sense) to @craftedxp/voice-rn 0.3.x so a\n// consumer can write the same `onTranscript` / `onStateChange` /\n// `onError` / `onEnd` shape for both web and RN clients.\n\nimport type { ClientToolCallFrame } from './clientTools'\n\n// Re-export so consumers / dashboards can type the wire shape. The\n// `client_marks` frame itself is purely an internal observability\n// channel between the SDK and server — it's emitted by the SDK and\n// logged on the server, never surfaced to consumer code.\nexport type { ClientMarksFrame } from './ClientMarksBuffer'\n\n// ---------------------------------------------------------------------------\n// State machine\n// ---------------------------------------------------------------------------\n\nexport type CallState =\n  | 'idle'\n  | 'connecting'\n  | 'listening'\n  | 'user_speaking'\n  | 'agent_speaking'\n  | 'ended'\n  | 'error'\n\n// ---------------------------------------------------------------------------\n// Transcript model — matches voice-rn TranscriptEntry\n// ---------------------------------------------------------------------------\n\nexport type TranscriptEntry =\n  | { id: string; role: 'user'; text: string; committed: boolean }\n  | { id: string; role: 'agent'; text: string; interrupted?: boolean }\n  | { id: string; role: 'tool'; text: string }\n  | { id: string; role: 'system'; text: string }\n\n// ---------------------------------------------------------------------------\n// Stable error code contract — matches voice-rn CallErrorCode where the\n// failure modes overlap. Web-specific codes (mic_denied via getUserMedia\n// rejection, etc.) keep their voice-rn names so cross-platform consumers\n// can write one switch statement.\n// ---------------------------------------------------------------------------\n\nexport type CallErrorCode =\n  // Programming errors — surface loudly to the host's developer.\n  | 'missing_credentials'\n  | 'forbidden'\n  // Browser audio failures.\n  | 'mic_denied'\n  | 'mic_start_failed'\n  | 'audio_session_failed'\n  // Auth lifecycle.\n  | 'token_expired'\n  | 'token_invalid'\n  | 'unauthorized'\n  // Network / connectivity.\n  | 'network_unreachable'\n  | 'socket_error'\n  // Business state.\n  | 'payment_required'\n  | 'not_found'\n  // End-of-call states surfaced via onError (also via onEnd reason='timeout').\n  | 'silence_timeout'\n  // Catch-all for unexpected server / 5xx / 1011.\n  | 'server_error'\n\nexport interface CallError {\n  code: CallErrorCode\n  message: string\n}\n\n// ---------------------------------------------------------------------------\n// End-of-call signal — matches voice-rn CallEndReason / CallEndEvent\n// ---------------------------------------------------------------------------\n\nexport type CallEndReason = 'agent_ended' | 'user_hangup' | 'timeout' | 'error'\n\nexport interface CallEndEvent {\n  reason: CallEndReason\n  // Present iff reason === 'error'. Mirrors the code from the most\n  // recent onError.\n  errorCode?: CallErrorCode\n  // Wallclock from start() resolving to the WS close. Useful for billing\n  // / \"you spoke for 1m23s\" UIs without forcing the host to track\n  // timestamps themselves.\n  durationMs: number\n}\n\n// ---------------------------------------------------------------------------\n// Volume meter event (browser bundle only — node bundle leaves volume to\n// the consumer if they're processing PCM themselves).\n// ---------------------------------------------------------------------------\n\nexport interface VolumeEvent {\n  // 0-1 RMS over the last ~100ms. Bind to a waveform / level meter.\n  input: number\n  output: number\n}\n\n// ---------------------------------------------------------------------------\n// Server → client message envelope. Loose typing — the server can add new\n// `type` values without breaking the SDK; unknown types are ignored by\n// the dispatch in handleServerMessage.\n// ---------------------------------------------------------------------------\n\nexport type ServerMessage = Record<string, unknown> & { type?: string }\n\n// ---------------------------------------------------------------------------\n// Stateless transcript reducer + state-machine helpers. Both the browser\n// and node clients call into these so the shape of the transcript stays\n// identical across environments.\n// ---------------------------------------------------------------------------\n\nexport interface ProtocolState {\n  state: CallState\n  transcript: TranscriptEntry[]\n  agentBubbleId: string | null\n  idCounter: number\n  // Reason latched from the server's call_end frame. Read by the\n  // surrounding client when the WS finally closes so onEnd fires with the\n  // right reason instead of falling back to user_hangup.\n  endReason: CallEndReason | null\n}\n\nexport const createProtocolState = (): ProtocolState => ({\n  state: 'idle',\n  transcript: [],\n  agentBubbleId: null,\n  idCounter: 0,\n  endReason: null,\n})\n\n// Side-effect callbacks the protocol layer fires as it processes server\n// frames. The surrounding client wires these up to event emitters\n// (browser) or to user-supplied callbacks (node).\nexport interface ProtocolCallbacks {\n  onState: (next: CallState) => void\n  onTranscript: (entries: TranscriptEntry[]) => void\n  onError: (err: CallError) => void\n  // Fires on `interrupt` — caller should flush its audio playback queue.\n  onInterrupt: () => void\n  // Fires on `agent_turn_start` — caller may want to reset its turn\n  // anchor for \"agent has been speaking N ms\" UIs. `seq` is the\n  // server-assigned turn ID; absent on legacy server builds that don't\n  // emit one (we treat that case as \"no correlation possible\").\n  onAgentTurnStart: (seq?: number) => void\n  // Fires on `agent_turn_end` — caller may use this to flush per-turn\n  // observability marks. `seq` matches the corresponding agent_turn_start.\n  onAgentTurnEnd: (seq?: number) => void\n  // Fires on `call_end` — caller closes its WS and resolves onEnd.\n  onCallEnd: (reason: CallEndReason) => void\n  // Fires on `connected` — caller should send the client_tools_register\n  // frame here so the server's first-turn grace window unblocks.\n  onConnected: () => void\n  // Fires on `client_tool_call` — caller dispatches to the matching\n  // tool handler and posts back a client_tool_result frame.\n  onClientToolCall: (frame: ClientToolCallFrame) => void\n}\n\n// Map server-supplied endReason strings onto our SDK-side CallEndReason.\nconst mapEndReason = (raw: string): CallEndReason => {\n  if (raw === 'agent_ended') return 'agent_ended'\n  if (raw === 'caller_hung_up') return 'user_hangup'\n  if (raw === 'silence_timeout' || raw === 'max_duration') return 'timeout'\n  return 'error'\n}\n\n// Pure-ish transcript reducer + dispatcher. Mutates `state` in place to\n// match the imperative pattern used by both clients; returns nothing.\n//\n// The \"agent_text\" / \"transcript\" interim handling is identical to\n// voice-rn's useVoiceCall — one growing user bubble while interim, one\n// growing agent bubble per turn.\nexport function handleServerMessage(\n  raw: string,\n  state: ProtocolState,\n  cb: ProtocolCallbacks,\n): void {\n  let msg: ServerMessage\n  try {\n    msg = JSON.parse(raw)\n  } catch {\n    return\n  }\n\n  switch (msg.type) {\n    case 'connected':\n      cb.onConnected()\n      setState(state, 'listening', cb)\n      return\n\n    case 'transcript': {\n      const text = (msg.text as string) ?? ''\n      if (!text) return\n      const isFinal = !!msg.isFinal\n      if (!isFinal) setState(state, 'user_speaking', cb)\n      upsertUserPartial(state, text, isFinal)\n      cb.onTranscript(state.transcript)\n      return\n    }\n\n    case 'agent_turn_start': {\n      const id = `m${state.idCounter++}`\n      state.agentBubbleId = id\n      state.transcript = [...state.transcript, { id, role: 'agent', text: '' }]\n      cb.onTranscript(state.transcript)\n      const seq = typeof msg.seq === 'number' ? (msg.seq as number) : undefined\n      cb.onAgentTurnStart(seq)\n      setState(state, 'agent_speaking', cb)\n      return\n    }\n\n    case 'agent_text': {\n      const delta = (msg.text as string) ?? ''\n      if (!delta || !state.agentBubbleId) return\n      const id = state.agentBubbleId\n      state.transcript = state.transcript.map((e) =>\n        e.id === id && e.role === 'agent' ? { ...e, text: e.text + delta } : e,\n      )\n      cb.onTranscript(state.transcript)\n      return\n    }\n\n    case 'agent_turn_end': {\n      state.agentBubbleId = null\n      const seq = typeof msg.seq === 'number' ? (msg.seq as number) : undefined\n      cb.onAgentTurnEnd(seq)\n      setState(state, 'listening', cb)\n      return\n    }\n\n    case 'interrupt':\n      cb.onInterrupt()\n      return\n\n    case 'agent_turn_abort': {\n      const committed = ((msg.committedText as string) ?? '').trim()\n      if (state.agentBubbleId) {\n        const id = state.agentBubbleId\n        if (committed) {\n          state.transcript = state.transcript.map((e) =>\n            e.id === id && e.role === 'agent' ? { ...e, text: committed, interrupted: true } : e,\n          )\n        } else {\n          state.transcript = state.transcript.filter((e) => e.id !== id)\n        }\n        cb.onTranscript(state.transcript)\n      }\n      state.agentBubbleId = null\n      return\n    }\n\n    case 'tool_call':\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'tool',\n          text: `→ ${String(msg.tool ?? '?')}(${msg.args ? JSON.stringify(msg.args) : ''})`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      return\n\n    case 'tool_result':\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'tool',\n          text: `${msg.ok ? '✓' : '✗'} ${String(msg.tool ?? '?')}`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      return\n\n    case 'client_tool_call': {\n      const toolCallId = String(msg.toolCallId ?? '')\n      const name = String(msg.name ?? '')\n      const args = (msg.args as Record<string, unknown>) ?? {}\n      if (!toolCallId || !name) return\n      cb.onClientToolCall({ toolCallId, name, args })\n      return\n    }\n\n    case 'call_end': {\n      const reasonRaw = String(msg.reason ?? '')\n      const reason = mapEndReason(reasonRaw)\n      state.endReason = reason\n      state.transcript = [\n        ...state.transcript,\n        {\n          id: `m${state.idCounter++}`,\n          role: 'system',\n          text: `call ended${reasonRaw ? ` (${reasonRaw})` : ''}`,\n        },\n      ]\n      cb.onTranscript(state.transcript)\n      cb.onCallEnd(reason)\n      return\n    }\n\n    case 'error': {\n      const code = (msg.code as CallErrorCode) ?? 'server_error'\n      const message = (msg.message as string) ?? 'server error'\n      cb.onError({ code, message })\n      return\n    }\n  }\n}\n\nconst setState = (state: ProtocolState, next: CallState, cb: ProtocolCallbacks) => {\n  if (state.state === next) return\n  // The consumer's setState (cb.onState) is responsible for both\n  // updating state.state and firing the user's onStateChange callback.\n  // We deliberately don't mutate here — if we did, the consumer's\n  // setState would early-return on the dedup check and never fire\n  // onStateChange for server-driven transitions.\n  cb.onState(next)\n}\n\n// Find the last uncommitted user bubble and grow it; or append a new\n// uncommitted bubble. Mirrors voice-rn upsertUserPartial.\nconst upsertUserPartial = (state: ProtocolState, text: string, isFinal: boolean) => {\n  let idx = -1\n  for (let i = state.transcript.length - 1; i >= 0; i--) {\n    const e = state.transcript[i]\n    if (e.role === 'user' && e.committed === false) {\n      idx = i\n      break\n    }\n  }\n  if (idx === -1) {\n    state.transcript = [\n      ...state.transcript,\n      { id: `m${state.idCounter++}`, role: 'user', text, committed: isFinal },\n    ]\n    return\n  }\n  const target = state.transcript[idx] as Extract<TranscriptEntry, { role: 'user' }>\n  const next = [...state.transcript]\n  next[idx] = { ...target, text, committed: isFinal }\n  state.transcript = next\n}\n\n// ---------------------------------------------------------------------------\n// WebSocket URL builder. Identical between browser + node bundles —\n// kept here so the rare server-side change to the path / query shape is\n// a one-line edit.\n// ---------------------------------------------------------------------------\n\nexport interface BuildWsUrlArgs {\n  apiBase: string\n  agentId: string\n  token: string\n  bargeIn?: boolean\n}\n\nexport function buildWsUrl(args: BuildWsUrlArgs): string {\n  const base = new URL(args.apiBase)\n  const proto = base.protocol === 'https:' ? 'wss:' : 'ws:'\n  const bargeQS = args.bargeIn === false ? '&barge=off' : ''\n  return `${proto}//${base.host}/v1/agents/${encodeURIComponent(args.agentId)}/call?token=${encodeURIComponent(args.token)}${bargeQS}`\n}\n","// Bundle-agnostic client-tools core for @craftedxp/voice-js.\n//\n// Imports neither `ws` nor DOM `WebSocket` — the dispatcher takes an\n// injected `send` callback so the same code runs in browsers and Node.\n//\n// Wire protocol (matches the server's Zod schemas in\n// server/src/agents/clientToolFrames.ts):\n//\n//   client → server  client_tools_register   { type, tools[] }\n//   server → client  client_tool_call        { type, toolCallId, name, args }\n//   client → server  client_tool_result      { type, toolCallId, result?, error? }\n\nexport interface ClientTool {\n  description: string\n  parameters: Record<string, unknown> // JSON Schema\n  usage?: string // ≤500 chars; appended to agent system prompt\n  timeoutMs?: number // server-enforced; (0, 30_000]\n  example?: string // surface-only UI hint; not sent on wire\n  handler: (args: Record<string, unknown>) => Promise<string | object> | string | object\n}\n\nexport type ClientToolMap = Record<string, ClientTool>\n\n// Server → client frame shape. Used by both protocol.ts (when surfacing\n// the frame to a callback) and dispatchClientToolCall (when consuming it).\nexport interface ClientToolCallFrame {\n  toolCallId: string\n  name: string\n  args: Record<string, unknown>\n}\n\nconst NAME_RE = /^[a-zA-Z_][a-zA-Z0-9_]*$/\nconst MAX_TOOLS = 64\nconst MAX_USAGE = 500\nconst MAX_TIMEOUT_MS = 30_000\n\n// Synchronous, throw-on-error validation. Called once at startCall time\n// before the WS opens, so bad input fails fast at the configuration\n// boundary instead of mid-call.\nexport const validateClientToolMap = (tools: ClientToolMap | undefined): void => {\n  if (tools === undefined) return\n  if (typeof tools !== 'object' || tools === null || Array.isArray(tools)) {\n    throw new Error('clientTools must be an object keyed by tool name')\n  }\n  const entries = Object.entries(tools)\n  if (entries.length > MAX_TOOLS) {\n    throw new Error(`clientTools may declare at most 64 tools (got ${entries.length})`)\n  }\n  for (const [name, def] of entries) {\n    if (!NAME_RE.test(name)) {\n      throw new Error(\n        `clientTools[\"${name}\"]: name must be a valid identifier (^[a-zA-Z_][a-zA-Z0-9_]*$)`,\n      )\n    }\n    if (!def || typeof def !== 'object') {\n      throw new Error(`clientTools[\"${name}\"]: must be an object`)\n    }\n    if (typeof def.description !== 'string' || def.description.length === 0) {\n      throw new Error(`clientTools[\"${name}\"]: must have a description`)\n    }\n    if (typeof def.handler !== 'function') {\n      throw new Error(`clientTools[\"${name}\"]: must have a handler function`)\n    }\n    if (def.usage !== undefined && def.usage.length > MAX_USAGE) {\n      throw new Error(`clientTools[\"${name}\"]: usage must be ≤500 chars`)\n    }\n    if (\n      def.timeoutMs !== undefined &&\n      (!Number.isFinite(def.timeoutMs) || def.timeoutMs <= 0 || def.timeoutMs > MAX_TIMEOUT_MS)\n    ) {\n      throw new Error(`clientTools[\"${name}\"]: timeoutMs must be in (0, 30000]`)\n    }\n  }\n}\n\n// Build the registration frame the SDK sends right after `connected`.\n// Strips runtime-only fields (handler, example) and only includes the\n// optional fields when set.\nexport const buildRegisterFrame = (tools: ClientToolMap): object => ({\n  type: 'client_tools_register',\n  tools: Object.entries(tools).map(([name, def]) => ({\n    name,\n    description: def.description,\n    parameters: def.parameters,\n    ...(def.usage !== undefined ? { usage: def.usage } : {}),\n    ...(def.timeoutMs !== undefined ? { timeoutMs: def.timeoutMs } : {}),\n  })),\n})\n\ntype FrameSender = (frame: object) => void\n\n// Run the matching handler asynchronously and post a result/error frame\n// back through `send`. Send-side throws are swallowed (the WS may have\n// closed mid-call; the server has already cancelled the in-flight\n// invocation on its side).\nexport const dispatchClientToolCall = (\n  send: FrameSender,\n  tools: ClientToolMap,\n  frame: ClientToolCallFrame,\n): void => {\n  const safeSend = (payload: object) => {\n    try {\n      send(payload)\n    } catch {\n      // WS likely closed mid-handler. Server-side already cancelled.\n    }\n  }\n\n  const tool = tools[frame.name]\n  if (!tool) {\n    safeSend({\n      type: 'client_tool_result',\n      toolCallId: frame.toolCallId,\n      error: `No handler for ${frame.name}`,\n    })\n    return\n  }\n\n  void (async () => {\n    try {\n      const out = await tool.handler(frame.args)\n      safeSend({\n        type: 'client_tool_result',\n        toolCallId: frame.toolCallId,\n        result: typeof out === 'string' ? out : JSON.stringify(out),\n      })\n    } catch (err) {\n      safeSend({\n        type: 'client_tool_result',\n        toolCallId: frame.toolCallId,\n        error: err instanceof Error ? err.message : String(err),\n      })\n    }\n  })()\n}\n","// ClientMarksBuffer — per-turn latency-mark correlator shared between the\n// browser and node SDK clients.\n//\n// Why this exists:\n//   The server mints a `seq` (turnId) only AFTER the user finishes\n//   speaking, at VoiceOrchestrator.handleTranscript. The client, however,\n//   captures its first user-perceived mark — `t_client_mic_to_first_frame`\n//   — before any server frame for the turn exists. We therefore buffer\n//   client-side marks under a \"pending\" slot and back-fill the seq when\n//   the server's `agent_turn_start { seq }` arrives. When `agent_turn_end\n//   { seq }` lands AND we've captured the inbound `client_first_audible`\n//   mark, the buffer flushes a `client_marks` frame to the server.\n//\n// Lifetime + bounds:\n//   - One pending bucket at a time. A new `agent_turn_start` flushes\n//     anything still pending (shouldn't happen in normal flow, but is\n//     safer than silently dropping marks).\n//   - If `agent_turn_end` arrives but we never captured first_audible\n//     (e.g. agent had no audio output), we still emit a `client_marks`\n//     frame with whatever marks are present. Missing marks are simply\n//     omitted from the wire payload.\n//   - The buffer is GC'd by replacement on every turn boundary; there is\n//     no long-term retention. A turn whose `agent_turn_end` never\n//     arrives (server crash mid-turn) is implicitly dropped on the next\n//     `agent_turn_start` or when the call ends. We do NOT add an\n//     explicit watchdog — observability marks aren't worth the\n//     complexity of a timer here.\n//\n// Time source:\n//   `performance.now()` in both browser and Node. Wall-clock-monotonic,\n//   sub-ms precision. Marks are stored as raw `performance.now()` values;\n//   only the deltas between them ever go on the wire.\n//\n// Wire shape (see protocol.ts):\n//   { type: 'client_marks', seq, marks: { ... }, clientNow }\n//\n// The CLAUDE.md / functional-style preference: this module is a small\n// factory returning a closure-based controller. No classes, no shared\n// mutable module state.\n\nexport interface ClientMarksFrame {\n  type: 'client_marks'\n  seq: number\n  marks: {\n    // The headline number: client-measured wall-clock between the first\n    // outbound audio frame for this turn and the first agent-audio\n    // chunk committed for playback. Includes mic-encode time, network\n    // RTT in both directions, server-side processing (STT + LLM + TTS),\n    // and any client-side scheduling delay. Absent if either endpoint\n    // wasn't captured.\n    client_mic_to_first_audible_ms?: number\n  }\n  // Wall-clock at send time, ms-since-epoch. Lets the analysis script\n  // estimate clock skew between client and server logs even though the\n  // marks themselves are deltas.\n  clientNow?: number\n}\n\n// What the consumer-facing controller exposes. The browser and node\n// clients both wire to this same surface.\nexport interface ClientMarksController {\n  // Outbound mark. Called on every outgoing audio chunk; the buffer\n  // gates so only the first chunk PER TURN is timestamped. `markTurnBoundary`\n  // resets the gate so the next chunk is treated as the start of the\n  // next turn's outbound stream.\n  markFirstOutboundAudio: () => void\n\n  // Inbound mark. Called every time the SDK commits an agent-audio\n  // chunk for playback (browser: AudioContext schedule; node: handed to\n  // consumer via onAudioChunk). Gated to first-per-turn.\n  markFirstAudibleOutput: () => void\n\n  // Called when the SDK receives `agent_turn_start { seq }`. Tags any\n  // accumulated outbound marks with this seq. Also resets the\n  // first-outbound gate so the NEXT turn's outbound mark can be\n  // captured again. (Outbound is captured for a turn before its seq is\n  // known; inbound is captured after, with the seq tagged.)\n  onAgentTurnStart: (seq: number) => void\n\n  // Called when the SDK receives `agent_turn_end { seq }`. Triggers\n  // emission of the buffered `client_marks` frame for that seq, with\n  // whatever marks are present (missing ones omitted).\n  onAgentTurnEnd: (seq: number) => void\n\n  // Force-flush any pending state. Called on call end / teardown so\n  // the last turn's marks aren't dropped on hangup.\n  flush: () => void\n}\n\n// `send` is the raw WS-send sink injected by the surrounding client.\n// Stringified JSON in browser/node. Returns whether the send was\n// accepted (we don't surface failure beyond the boolean — observability\n// frames are best-effort).\nexport type ClientMarksSend = (frame: ClientMarksFrame) => void\n\n// `now` defaults to performance.now() but is injected so unit tests can\n// drive a fake clock deterministically.\nexport interface CreateClientMarksBufferArgs {\n  send: ClientMarksSend\n  now?: () => number\n}\n\nexport const createClientMarksBuffer = (\n  args: CreateClientMarksBufferArgs,\n): ClientMarksController => {\n  const now = args.now ?? (() => performance.now())\n\n  // Pending bucket — one in-flight turn at a time. We hold the OUTBOUND\n  // mark before we know `seq`, then graduate to a per-seq bucket on\n  // agent_turn_start so a second turn's inbound marks can't trample the\n  // first turn's outbound mark before flush.\n  let pendingFirstOutboundAt: number | null = null\n\n  // Once agent_turn_start { seq } arrives we move the pending outbound\n  // mark into a per-seq slot and start collecting the inbound mark.\n  // Map keyed by seq; in practice never holds more than 1-2 entries\n  // (one currently-flushing, one just-started).\n  const inFlight = new Map<\n    number,\n    {\n      firstOutboundAt: number | null\n      firstAudibleAt: number | null\n      ended: boolean\n    }\n  >()\n\n  const tryEmit = (seq: number) => {\n    const slot = inFlight.get(seq)\n    if (!slot) return\n    if (!slot.ended) return\n    // Build wire payload. Only include marks we actually captured.\n    const marks: ClientMarksFrame['marks'] = {}\n    if (slot.firstOutboundAt !== null && slot.firstAudibleAt !== null) {\n      marks.client_mic_to_first_audible_ms = slot.firstAudibleAt - slot.firstOutboundAt\n    }\n    args.send({\n      type: 'client_marks',\n      seq,\n      marks,\n      clientNow: Date.now(),\n    })\n    inFlight.delete(seq)\n  }\n\n  const markFirstOutboundAudio = () => {\n    if (pendingFirstOutboundAt !== null) return // already stamped this turn\n    pendingFirstOutboundAt = now()\n  }\n\n  const markFirstAudibleOutput = () => {\n    // Inbound is captured AFTER agent_turn_start, so we know which seq\n    // it belongs to: the most recent un-ended in-flight slot. If there's\n    // somehow none (race: audio before agent_turn_start), drop silently.\n    let target: { firstAudibleAt: number | null; ended: boolean } | undefined\n    for (const slot of inFlight.values()) {\n      if (!slot.ended) {\n        target = slot\n        // No break — last un-ended wins, matching the \"most recent\n        // turn\" intent if multiple are in flight.\n      }\n    }\n    if (!target) return\n    if (target.firstAudibleAt !== null) return // first-per-turn gate\n    target.firstAudibleAt = now()\n  }\n\n  const onAgentTurnStart = (seq: number) => {\n    inFlight.set(seq, {\n      firstOutboundAt: pendingFirstOutboundAt,\n      firstAudibleAt: null,\n      ended: false,\n    })\n    // Reset outbound gate so the NEXT turn captures cleanly. Note:\n    // pendingFirstOutboundAt is already moved into the slot above; we\n    // don't need to preserve it.\n    pendingFirstOutboundAt = null\n  }\n\n  const onAgentTurnEnd = (seq: number) => {\n    const slot = inFlight.get(seq)\n    if (!slot) {\n      // Got an end with no matching start (server bug, or restart). Best\n      // effort: emit an empty marks frame so the analysis script knows\n      // the client saw the turn at all.\n      args.send({ type: 'client_marks', seq, marks: {}, clientNow: Date.now() })\n      return\n    }\n    slot.ended = true\n    tryEmit(seq)\n  }\n\n  const flush = () => {\n    for (const seq of [...inFlight.keys()]) {\n      const slot = inFlight.get(seq)!\n      slot.ended = true\n      tryEmit(seq)\n    }\n    pendingFirstOutboundAt = null\n  }\n\n  return {\n    markFirstOutboundAudio,\n    markFirstAudibleOutput,\n    onAgentTurnStart,\n    onAgentTurnEnd,\n    flush,\n  }\n}\n","// Browser VoiceClient — drives one in-progress call.\n//\n// Created by `BrowserVoiceFactory.startCall` (see browser.ts). Most\n// consumers never touch this class directly — they receive a `Call`\n// handle that exposes `end`/`mute`/`unmute` and subscribe to events\n// via the `onState` / `onTranscript` / `onError` / `onEnd` callbacks\n// they passed to `startCall`.\n//\n// This is the rough equivalent of @craftedxp/voice-rn's `useVoiceCall`\n// hook, but environment-agnostic (no React).\n\nimport { createAudioCapture, type CaptureController } from './AudioCapture'\nimport { createAudioPlayback, type PlaybackController } from './AudioPlayback'\nimport {\n  createReconnectingWebSocket,\n  type ReconnectingWebSocket,\n  type WebSocketFactory,\n} from './ReconnectingWebSocket'\nimport {\n  buildWsUrl,\n  createProtocolState,\n  handleServerMessage,\n  type CallEndReason,\n  type CallError,\n  type ProtocolState,\n} from './protocol'\nimport type { Call, StartCallOptions, VoiceClientConfig } from './config'\nimport { buildRegisterFrame, dispatchClientToolCall, validateClientToolMap } from './clientTools'\nimport { createClientMarksBuffer, type ClientMarksController } from './ClientMarksBuffer'\n\ninterface VoiceClientArgs {\n  // Pre-resolved by the factory.\n  config: VoiceClientConfig\n  options: StartCallOptions\n  token: string\n  // Browser uses native WebSocket; node injects `ws`.\n  wsFactory: WebSocketFactory\n}\n\nexport class BrowserVoiceClient implements Call {\n  private readonly args: VoiceClientArgs\n  private readonly proto: ProtocolState\n\n  private rws: ReconnectingWebSocket | null = null\n  private capture: CaptureController | null = null\n  private playback: PlaybackController | null = null\n\n  private muted = false\n  private inputVolume = 0\n  private outputVolume = 0\n  private startedAt: number | null = null\n  private endedFired = false\n  private lastError: CallError | null = null\n  private marks: ClientMarksController\n\n  constructor(args: VoiceClientArgs) {\n    this.args = args\n    this.proto = createProtocolState()\n    validateClientToolMap(args.options.clientTools)\n    // Best-effort observability frame. Wraps WS send so a failed send\n    // (closed socket) is silently swallowed — these marks are never\n    // load-bearing for the call itself.\n    this.marks = createClientMarksBuffer({\n      send: (frame) => {\n        try {\n          this.rws?.send(JSON.stringify(frame))\n        } catch {\n          // WS already gone; drop the frame.\n        }\n      },\n    })\n  }\n\n  // ---------------------------------------------------------------\n  // Call interface\n  // ---------------------------------------------------------------\n\n  get state() {\n    return this.proto.state\n  }\n\n  get transcript() {\n    return this.proto.transcript.slice()\n  }\n\n  get isMuted() {\n    return this.muted\n  }\n\n  end = () => {\n    this.teardown('user_hangup')\n  }\n\n  mute = () => {\n    if (this.muted) return\n    this.muted = true\n    this.capture?.mute(true)\n  }\n\n  unmute = () => {\n    if (!this.muted) return\n    this.muted = false\n    this.capture?.mute(false)\n  }\n\n  // ---------------------------------------------------------------\n  // Lifecycle — called by the factory immediately after construction.\n  // Resolves once the WS is open and capture is starting; mid-call\n  // failures arrive via `onError`.\n  // ---------------------------------------------------------------\n\n  async start(): Promise<void> {\n    this.setState('connecting')\n    this.startedAt = Date.now()\n\n    const url = buildWsUrl({\n      apiBase: this.args.config.apiBase,\n      agentId: this.args.options.agentId,\n      token: this.args.token,\n      bargeIn: this.args.options.bargeIn,\n    })\n\n    // Playback context needs a user gesture before its AudioContext can\n    // produce sound. In practice startCall is invoked inside a click\n    // handler; resume() is cheap and idempotent.\n    this.playback = createAudioPlayback({\n      onVolume: (v) => {\n        this.outputVolume = v\n        this.args.options.onVolume?.({ input: this.inputVolume, output: v })\n      },\n    })\n    try {\n      await this.playback.resume()\n    } catch {\n      // AudioContext may refuse if we weren't called from a gesture.\n      // First enqueue() will retry the resume.\n    }\n\n    this.rws = createReconnectingWebSocket(\n      {\n        url,\n        wsFactory: this.args.wsFactory,\n        maxRetries: 3,\n      },\n      (ev) => this.handleSocketEvent(ev),\n    )\n  }\n\n  // ---------------------------------------------------------------\n  // Internal\n  // ---------------------------------------------------------------\n\n  private sendClientToolsRegister = () => {\n    const frame = buildRegisterFrame(this.args.options.clientTools ?? {})\n    this.rws?.send(JSON.stringify(frame))\n  }\n\n  private setState = (next: ProtocolState['state']) => {\n    if (this.proto.state === next) return\n    this.proto.state = next\n    this.args.options.onStateChange?.(next)\n  }\n\n  private emitError = (err: CallError) => {\n    this.lastError = err\n    this.args.options.onError?.(err)\n  }\n\n  private handleSocketEvent = (\n    ev:\n      | { type: 'open' }\n      | { type: 'reconnected' }\n      | { type: 'message'; data: string | ArrayBuffer }\n      | { type: 'close'; code: number; reason: string; permanent: boolean }\n      | { type: 'error'; error: Error },\n  ) => {\n    switch (ev.type) {\n      case 'open':\n        void this.startCapture()\n        break\n      case 'reconnected':\n        // Treat as a fresh call — server session is gone, no resume.\n        this.proto.transcript = []\n        this.proto.agentBubbleId = null\n        this.args.options.onTranscript?.(this.proto.transcript)\n        void this.startCapture()\n        this.setState('listening')\n        break\n      case 'message':\n        if (typeof ev.data === 'string') {\n          handleServerMessage(ev.data, this.proto, {\n            onState: this.setState,\n            onTranscript: (entries) => this.args.options.onTranscript?.(entries),\n            onError: this.emitError,\n            onInterrupt: () => {\n              this.playback?.flush()\n              this.args.options.onInterrupt?.()\n            },\n            onAgentTurnStart: (seq) => {\n              if (typeof seq === 'number') this.marks.onAgentTurnStart(seq)\n              this.args.options.onAgentTurnStart?.()\n            },\n            onAgentTurnEnd: (seq) => {\n              if (typeof seq === 'number') this.marks.onAgentTurnEnd(seq)\n            },\n            onCallEnd: (reason) => this.teardown(reason),\n            onConnected: () => this.sendClientToolsRegister(),\n            onClientToolCall: (frame) =>\n              dispatchClientToolCall(\n                (f) => this.rws?.send(JSON.stringify(f)),\n                this.args.options.clientTools ?? {},\n                frame,\n              ),\n          })\n        } else {\n          // Inbound mark: stamped at the moment we COMMIT this chunk for\n          // playback (enqueue schedules into AudioContext). We\n          // intentionally don't await — enqueue handles its own\n          // AudioContext warm-up; the mark captures the SDK's commit\n          // moment, which is the closest we can get to \"user heard it\"\n          // without instrumenting the AudioContext output graph.\n          this.marks.markFirstAudibleOutput()\n          this.playback?.enqueue(ev.data)\n        }\n        break\n      case 'close':\n        if (ev.permanent) {\n          // Pick the most informative reason: server-supplied wins, then\n          // last error, then default to user_hangup.\n          const reason: CallEndReason =\n            this.proto.endReason ?? (this.lastError ? 'error' : 'user_hangup')\n          this.teardown(reason)\n        }\n        break\n      case 'error':\n        this.emitError({ code: 'socket_error', message: ev.error.message })\n        break\n    }\n  }\n\n  private startCapture = async () => {\n    if (this.capture?.isCapturing()) return\n    this.capture = createAudioCapture({\n      onChunk: (pcm) => {\n        // Outbound mark — stamps the first frame leaving the SDK for\n        // each turn. The buffer gates so only the first chunk per\n        // pending-turn window is timestamped; subsequent chunks are\n        // no-ops until agent_turn_start moves the slot to a per-seq\n        // bucket and resets the gate. Semantic: this is \"moment audio\n        // is dispatched to the WS\", which in continuous open-mic mode\n        // is essentially the encode-and-handoff time per first frame\n        // after a turn boundary.\n        this.marks.markFirstOutboundAudio()\n        this.rws?.send(pcm)\n      },\n      onVolume: (v) => {\n        this.inputVolume = v\n        this.args.options.onVolume?.({ input: v, output: this.outputVolume })\n      },\n      onError: (err) => {\n        this.emitError({\n          code: err.name === 'NotAllowedError' ? 'mic_denied' : 'mic_start_failed',\n          message: err.message,\n        })\n      },\n    })\n    if (this.muted) this.capture.mute(true)\n    try {\n      await this.capture.start()\n    } catch {\n      // onError already emitted inside createAudioCapture.\n    }\n  }\n\n  private teardown = (reason: CallEndReason) => {\n    // Flush any in-flight observability marks BEFORE the WS closes so\n    // the final turn's client_marks frame doesn't get dropped on\n    // hangup.\n    try {\n      this.marks.flush()\n    } catch {\n      // never fatal\n    }\n    this.capture?.stop()\n    this.capture = null\n    this.playback?.close()\n    this.playback = null\n    try {\n      this.rws?.close(1000, reason)\n    } catch {\n      // already closed\n    }\n    this.rws = null\n    this.setState('ended')\n    this.fireEndOnce(reason)\n  }\n\n  private fireEndOnce = (reason: CallEndReason) => {\n    if (this.endedFired) return\n    this.endedFired = true\n    const startedAt = this.startedAt ?? Date.now()\n    this.args.options.onEnd?.({\n      reason,\n      errorCode: reason === 'error' ? this.lastError?.code : undefined,\n      durationMs: Date.now() - startedAt,\n    })\n  }\n}\n","// WebRTC transport for voice-js. Returns a `Call` with the same surface\n// as the WS path so consumers don't branch on transport. Signaling is\n// one-shot HTTP: POST /webrtc/offer (returns the SDP answer + callId);\n// trickled candidates go to POST /webrtc/ice. Audio flows over the\n// RTCPeerConnection; protocol frames (transcript, agent_turn_start, ...)\n// arrive on an in-band DataChannel named `control`.\n\nimport { createProtocolState, handleServerMessage, type ProtocolState } from '../protocol'\nimport type { CallError, CallState, TranscriptEntry } from '../protocol'\nimport type { Call } from '../config'\nimport {\n  buildRegisterFrame,\n  dispatchClientToolCall,\n  validateClientToolMap,\n  type ClientToolMap,\n} from '../clientTools'\n\nexport interface CreateWebRtcCallOptions {\n  agentId: string\n  /** API base (`https://api.voissia.com` etc). Used as the signaling\n   *  fallback when `webrtcGatewayBase` is unset. */\n  apiBase: string\n  /** Pre-minted ct_. */\n  token: string\n  /** Gateway base from the mint response. When falsy, signaling goes to\n   *  `${apiBase}/v1/agents/:agentId/webrtc/{offer,ice}` (Phase-1 routes,\n   *  local dev). */\n  webrtcGatewayBase?: string\n  onStateChange?: (s: CallState) => void\n  onTranscript?: (entries: TranscriptEntry[]) => void\n  onError?: (e: CallError) => void\n  onEnd?: () => void\n  onInterrupt?: () => void\n  onAgentTurnStart?: () => void\n  /** Tool palette the agent's LLM may invoke during the call. Registered\n   *  with the server on DataChannel open and dispatched on incoming\n   *  client_tool_call frames. Same contract as the WS path. */\n  clientTools?: ClientToolMap\n}\n\nexport async function createWebRtcCall(opts: CreateWebRtcCallOptions): Promise<Call> {\n  // Validate up front so a malformed tool map fails before mic prompt.\n  validateClientToolMap(opts.clientTools)\n\n  const proto: ProtocolState = createProtocolState()\n  let muted = false\n  let ended = false\n  const tools: ClientToolMap = opts.clientTools ?? {}\n\n  // Send a control-channel frame as JSON. Used by the tool register on\n  // open + by dispatchClientToolCall to ship results back. Drops the\n  // send if the channel isn't open yet (the server will retry the tool\n  // call if it doesn't get a result).\n  const sendControl = (frame: object): void => {\n    if (dc?.readyState !== 'open') return\n    try {\n      dc.send(JSON.stringify(frame))\n    } catch {\n      // channel closed mid-send\n    }\n  }\n\n  const fireState = (next: CallState) => {\n    if (proto.state === next) return\n    proto.state = next\n    opts.onStateChange?.(next)\n  }\n\n  const dispatch = (raw: string) => {\n    handleServerMessage(raw, proto, {\n      onState: fireState,\n      onTranscript: (entries) => opts.onTranscript?.(entries),\n      onError: (err) => opts.onError?.(err),\n      onInterrupt: () => opts.onInterrupt?.(),\n      onAgentTurnStart: () => opts.onAgentTurnStart?.(),\n      onAgentTurnEnd: () => {},\n      onCallEnd: () => teardown(),\n      onConnected: () => {\n        // Server's `connected` frame is our cue to register the tool\n        // palette. Mirrors the WS path's behaviour in VoiceClient.ts.\n        if (Object.keys(tools).length > 0) {\n          sendControl(buildRegisterFrame(tools))\n        }\n      },\n      onClientToolCall: (frame) => {\n        dispatchClientToolCall(sendControl, tools, frame)\n      },\n    })\n  }\n\n  fireState('connecting')\n\n  const pc = new RTCPeerConnection({\n    iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n  })\n\n  const audioEl = document.createElement('audio')\n  audioEl.autoplay = true\n  audioEl.style.display = 'none'\n  document.body.appendChild(audioEl)\n  pc.ontrack = (event) => {\n    audioEl.srcObject = event.streams[0] ?? new MediaStream([event.track])\n  }\n\n  let mic: MediaStream\n  try {\n    mic = await navigator.mediaDevices.getUserMedia({ audio: true })\n  } catch (err) {\n    const code =\n      err instanceof DOMException && err.name === 'NotAllowedError'\n        ? 'mic_denied'\n        : 'mic_start_failed'\n    opts.onError?.({\n      code,\n      message: err instanceof Error ? err.message : 'getUserMedia failed',\n    })\n    fireState('error')\n    pc.close()\n    audioEl.remove()\n    throw err\n  }\n  for (const track of mic.getAudioTracks()) pc.addTrack(track, mic)\n\n  const dc: RTCDataChannel = pc.createDataChannel('control', { ordered: true })\n  dc.onmessage = (e) => {\n    if (typeof e.data === 'string') dispatch(e.data)\n  }\n  dc.onerror = () => {\n    opts.onError?.({ code: 'socket_error', message: 'control channel error' })\n  }\n  // Defensive: if the server never emits a `connected` protocol frame\n  // but the channel opens (older server builds or future protocol\n  // skew), still register the tools so they're available the moment\n  // the LLM tries to call one.\n  dc.onopen = () => {\n    if (Object.keys(tools).length > 0) {\n      sendControl(buildRegisterFrame(tools))\n    }\n  }\n\n  const gateway = opts.webrtcGatewayBase || ''\n  const offerUrl = gateway\n    ? `${gateway}/webrtc/offer?token=${encodeURIComponent(opts.token)}`\n    : `${opts.apiBase}/v1/agents/${encodeURIComponent(opts.agentId)}/webrtc/offer?token=${encodeURIComponent(opts.token)}`\n  const iceUrl = gateway\n    ? `${gateway}/webrtc/ice?token=${encodeURIComponent(opts.token)}`\n    : `${opts.apiBase}/v1/agents/${encodeURIComponent(opts.agentId)}/webrtc/ice?token=${encodeURIComponent(opts.token)}`\n\n  await pc.setLocalDescription(await pc.createOffer())\n\n  let callId: string\n  try {\n    const offerRes = await fetch(offerUrl, {\n      method: 'POST',\n      headers: { 'content-type': 'application/json' },\n      body: JSON.stringify({ sdp: pc.localDescription!.sdp, type: 'offer', agentId: opts.agentId }),\n    })\n    if (!offerRes.ok) {\n      const code = offerRes.status === 401 ? 'unauthorized' : 'server_error'\n      opts.onError?.({ code, message: `signaling failed: HTTP ${offerRes.status}` })\n      fireState('error')\n      mic.getTracks().forEach((t) => t.stop())\n      pc.close()\n      audioEl.remove()\n      throw new Error(`webrtc offer failed: ${offerRes.status}`)\n    }\n    const body = (await offerRes.json()) as { sdp: string; type: 'answer'; callId: string }\n    callId = body.callId\n    await pc.setRemoteDescription({ type: 'answer', sdp: body.sdp })\n  } catch (err) {\n    if (!ended) {\n      opts.onError?.({\n        code: 'network_unreachable',\n        message: err instanceof Error ? err.message : 'signaling failed',\n      })\n      fireState('error')\n      mic.getTracks().forEach((t) => t.stop())\n      pc.close()\n      audioEl.remove()\n    }\n    throw err\n  }\n\n  pc.onicecandidate = (e) => {\n    if (!e.candidate) return\n    void fetch(iceUrl, {\n      method: 'POST',\n      headers: { 'content-type': 'application/json' },\n      body: JSON.stringify({ callId, candidate: e.candidate }),\n    }).catch(() => {\n      /* best-effort */\n    })\n  }\n\n  pc.onconnectionstatechange = () => {\n    const s = pc.connectionState\n    if (s === 'connected') fireState('listening')\n    if (s === 'failed' || s === 'disconnected') {\n      opts.onError?.({ code: 'socket_error', message: `webrtc connection ${s}` })\n      teardown()\n    }\n    if (s === 'closed' && !ended) teardown()\n  }\n\n  const teardown = () => {\n    if (ended) return\n    ended = true\n    try {\n      mic.getTracks().forEach((t) => t.stop())\n    } catch {\n      /* already stopped */\n    }\n    try {\n      pc.close()\n    } catch {\n      /* already closed */\n    }\n    try {\n      audioEl.remove()\n    } catch {\n      /* already removed */\n    }\n    fireState('ended')\n    opts.onEnd?.()\n  }\n\n  return {\n    get state() {\n      return proto.state\n    },\n    get transcript() {\n      return proto.transcript.slice()\n    },\n    get isMuted() {\n      return muted\n    },\n    end: () => teardown(),\n    mute: () => {\n      if (muted) return\n      muted = true\n      mic.getAudioTracks().forEach((t) => (t.enabled = false))\n    },\n    unmute: () => {\n      if (!muted) return\n      muted = false\n      mic.getAudioTracks().forEach((t) => (t.enabled = true))\n    },\n  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