@dialgem/client 0.0.0-stage → 1.0.0

package/dist/index.cjs.map

@@ -0,0 +1 @@
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A minimal typed event emitter. A handler that throws never stops the others or the call. */\nexport class Emitter<Events extends Record<string, unknown>> {\n  private handlers = new Map<keyof Events, Set<(payload: never) => void>>()\n\n  on<K extends keyof Events>(event: K, handler: (payload: Events[K]) => void): () => void {\n    let set = this.handlers.get(event)\n    if (!set) this.handlers.set(event, (set = new Set()))\n    set.add(handler as (payload: never) => void)\n    return () => this.off(event, handler)\n  }\n\n  off<K extends keyof Events>(event: K, handler: (payload: Events[K]) => void): void {\n    this.handlers.get(event)?.delete(handler as (payload: never) => void)\n  }\n\n  protected emit<K extends keyof Events>(event: K, payload: Events[K]): void {\n    for (const handler of [...(this.handlers.get(event) ?? [])]) {\n      try {\n        ;(handler as (payload: Events[K]) => void)(payload)\n      } catch (error) {\n        // Report it without breaking the call: a page's rendering bug is not a reason to drop the audio.\n        if (typeof reportError === \"function\") reportError(error)\n        else console.error(error)\n      }\n    }\n  }\n}\n","/**\n * What the call WebSocket speaks. Everything here is pure, so it is tested without a browser.\n *\n * Client to server: one JSON text frame to open the call, then binary frames of 16 kHz mono signed\n * 16-bit little-endian PCM, and the text frame `hangup` to end it.\n * Server to client: JSON text frames (events) and binary frames of 16 kHz mono signed 16-bit PCM\n * (the agent's voice), each sentence beginning with a 44-byte RIFF header.\n */\n\n/** The only address the SDK talks to unless a self-hosted deployment says otherwise. */\nexport const DEFAULT_BASE_URL = \"wss://api.dialgem.com\"\n\n/** Both directions of the call's audio. */\nexport const SAMPLE_RATE = 16_000\n\n/** Microphone audio is sent in chunks of this many samples: 40 ms, small enough not to delay speech detection. */\nexport const CAPTURE_CHUNK_SAMPLES = 640\n\nexport class UrlError extends Error {}\n\n/**\n * The WebSocket address for a call. The token rides in the query string because browsers cannot set\n * headers on a WebSocket; it is single-use and lives 60 seconds, which is what makes that acceptable.\n *\n * Plain `ws:` is refused except for a local address: a token sent in the clear can be replayed by\n * whoever reads it before the real page connects.\n */\nexport function buildCallUrl(baseURL: string, accessToken: string): string {\n  let url: URL\n  try {\n    url = new URL(baseURL)\n  } catch {\n    throw new UrlError(`baseURL is not a URL: ${baseURL}`)\n  }\n  if (url.protocol === \"https:\") url.protocol = \"wss:\"\n  if (url.protocol === \"http:\") url.protocol = \"ws:\"\n  if (url.protocol !== \"wss:\" && url.protocol !== \"ws:\") throw new UrlError(\"baseURL must be a wss:// or ws:// address\")\n  const local = url.hostname === \"localhost\" || url.hostname === \"127.0.0.1\" || url.hostname === \"[::1]\"\n  if (url.protocol === \"ws:\" && !local) throw new UrlError(\"baseURL must use wss:// except for localhost\")\n  if (!accessToken) throw new UrlError(\"accessToken is empty\")\n  // An organization API key (`zk_...`) grants full access to the account. A page that is handed one has\n  // already leaked it to every visitor, so refuse before it is put in a URL that gets logged.\n  if (/^zk_/i.test(accessToken.trim())) {\n    throw new UrlError(\n      \"That is an organization API key, not a call token. Never put an API key in a web page: create the call on your server with webCalls.create() and give the page its access_token.\",\n    )\n  }\n  const base = url.toString().replace(/\\/+$/, \"\")\n  const query = new URLSearchParams({ web_token: accessToken, live_transcript: \"1\" })\n  return `${base}/v1/agent/call?${query.toString()}`\n}\n\n/** The first frame. The server decides the agent and its configuration from the token and ignores this body. */\nexport const INIT_FRAME = JSON.stringify({ agent_config: {} })\n\nexport type ServerFrame =\n  | { type: \"session_started\"; sessionId: string }\n  | { type: \"transcript\"; role: \"user\" | \"agent\"; text: string }\n  | { type: \"transcript_partial\"; text: string }\n  | { type: \"agent_interrupted\" }\n  | { type: \"call_ended\" }\n  | { type: \"error\"; reason: string }\n  | { type: \"ignored\" }\n\n/** One text frame from the server. Anything unrecognized (including malformed JSON) is `ignored`, never an exception. */\nexport function parseServerFrame(text: string): ServerFrame {\n  let msg: unknown\n  try {\n    msg = JSON.parse(text)\n  } catch {\n    return { type: \"ignored\" }\n  }\n  if (typeof msg !== \"object\" || msg === null) return { type: \"ignored\" }\n  const m = msg as Record<string, unknown>\n  if (typeof m[\"error\"] === \"string\") {\n    return { type: \"error\", reason: typeof m[\"disconnection_reason\"] === \"string\" ? m[\"disconnection_reason\"] : m[\"error\"] }\n  }\n  switch (m[\"type\"]) {\n    case \"session_started\":\n      return { type: \"session_started\", sessionId: typeof m[\"session_id\"] === \"string\" ? m[\"session_id\"] : \"\" }\n    case \"transcript\":\n      if ((m[\"role\"] === \"user\" || m[\"role\"] === \"agent\") && typeof m[\"text\"] === \"string\") {\n        return { type: \"transcript\", role: m[\"role\"], text: m[\"text\"] }\n      }\n      return { type: \"ignored\" }\n    case \"transcript_partial\":\n      return m[\"role\"] === \"user\" && typeof m[\"text\"] === \"string\" ? { type: \"transcript_partial\", text: m[\"text\"] } : { type: \"ignored\" }\n    case \"agent_interrupted\":\n      return { type: \"agent_interrupted\" }\n    case \"call_ended\":\n      return { type: \"call_ended\" }\n    default:\n      return { type: \"ignored\" }\n  }\n}\n\n/**\n * Strips the 44-byte RIFF header the server puts at the start of each sentence. A header-only frame\n * yields nothing. Audio that does not start with one is returned as is.\n */\nexport function stripRiffHeader(chunk: ArrayBuffer): ArrayBuffer | null {\n  if (chunk.byteLength >= 4) {\n    const magic = new Uint8Array(chunk, 0, 4)\n    if (magic[0] === 0x52 && magic[1] === 0x49 && magic[2] === 0x46 && magic[3] === 0x46) {\n      return chunk.byteLength <= 44 ? null : chunk.slice(44)\n    }\n  }\n  return chunk\n}\n\n/**\n * What the server said when it turned the call away. Browsers cannot see the close code of a socket refused before\n * it is accepted, so the server accepts a refused call, sends this reason in one frame and then closes it.\n */\nexport function describeRefusal(reason: string): {\n  code: \"refused\" | \"busy\" | \"server_error\" | \"call_refused\"\n  message: string\n} {\n  switch (reason) {\n    case \"origin_not_allowed\":\n      return { code: \"refused\", message: \"This page's origin is not allowed to start calls. Add it under Web calls in your Dialgem settings.\" }\n    case \"token_expired\":\n      return { code: \"refused\", message: \"The access token expired before the call connected. Request a new one and start again.\" }\n    case \"token_already_used\":\n      return { code: \"refused\", message: \"The access token was already used. Each call needs its own token from your backend.\" }\n    case \"token_invalid\":\n    case \"bad_request\":\n      return { code: \"refused\", message: \"The access token is not valid for a call.\" }\n    case \"concurrency_limit_reached\":\n      return { code: \"busy\", message: \"Too many calls are running right now. Try again in a moment.\" }\n    case \"web_calls_unavailable\":\n      return { code: \"server_error\", message: \"Browser calls are not available right now.\" }\n    default:\n      return { code: \"call_refused\", message: `The call was declined: ${reason}.` }\n  }\n}\n\n/** What a closed socket means, from its close code. */\nexport function describeClose(code: number): { code: \"refused\" | \"busy\" | \"server_error\" | \"connection_failed\"; message: string } {\n  switch (code) {\n    case 1008:\n      return {\n        code: \"refused\",\n        message:\n          \"The call was refused. The access token was invalid, expired or already used, or this page's origin is not allowed to start calls.\",\n      }\n    case 1013:\n      return { code: \"busy\", message: \"Too many calls are running right now. Try again in a moment.\" }\n    case 1011:\n    case 1002:\n      return { code: \"server_error\", message: \"The call could not start because of a problem on Dialgem's side.\" }\n    default:\n      return { code: \"connection_failed\", message: \"Could not connect to the call.\" }\n  }\n}\n\n/** Float samples in [-1, 1] as little-endian 16-bit PCM. */\nexport function floatToPcm16(samples: Float32Array): ArrayBuffer {\n  const out = new Int16Array(samples.length)\n  for (let i = 0; i < samples.length; i++) {\n    const s = Math.max(-1, Math.min(1, samples[i] ?? 0))\n    out[i] = s < 0 ? s * 32768 : s * 32767\n  }\n  return out.buffer\n}\n","import { SAMPLE_RATE, stripRiffHeader } from \"./protocol.js\"\n\n/** The part of `AudioContext` the player uses, so a test can stand in for it. */\nexport interface PlaybackContext {\n  readonly currentTime: number\n  readonly destination: AudioNode\n  createBuffer(channels: number, length: number, sampleRate: number): AudioBuffer\n  createBufferSource(): AudioBufferSourceNode\n}\n\n/**\n * Plays the agent's voice as it arrives.\n *\n * Each binary frame is scheduled onto the audio clock right after the previous one, so speech starts within\n * a few milliseconds of the first frame instead of after a whole sentence has been buffered, and consecutive\n * frames join without a gap. The buffers are 16 kHz; the browser resamples them to whatever the output device\n * runs at.\n *\n * Network delivery is faster than playback, so the last byte of a sentence arrives while the speaker still has\n * a second of it to say. `remainingMs` is how much is still queued, which is what lets a call that ends\n * with a goodbye let the goodbye finish.\n */\nexport class PcmPlayer {\n  private nextTime = 0\n  private sources: AudioBufferSourceNode[] = []\n  /** A trailing odd byte: frames are not guaranteed to end on a sample boundary. */\n  private carry: Uint8Array | null = null\n\n  constructor(private readonly ctx: PlaybackContext) {}\n\n  feed(chunk: ArrayBuffer): void {\n    let data = stripRiffHeader(chunk)\n    if (data === null) return\n    if (this.carry) {\n      const joined = new Uint8Array(this.carry.length + data.byteLength)\n      joined.set(this.carry, 0)\n      joined.set(new Uint8Array(data), this.carry.length)\n      data = joined.buffer\n      this.carry = null\n    }\n    const usable = data.byteLength - (data.byteLength % 2)\n    if (usable !== data.byteLength) this.carry = new Uint8Array(data.slice(usable))\n    const samples = usable / 2\n    if (samples < 1) return\n\n    const pcm = new Int16Array(data.slice(0, usable))\n    const floats = new Float32Array(samples)\n    for (let i = 0; i < samples; i++) floats[i] = (pcm[i] ?? 0) / 32768\n\n    const buffer = this.ctx.createBuffer(1, samples, SAMPLE_RATE)\n    buffer.copyToChannel(floats, 0)\n    const source = this.ctx.createBufferSource()\n    source.buffer = buffer\n    source.connect(this.ctx.destination)\n    source.onended = () => {\n      const i = this.sources.indexOf(source)\n      if (i !== -1) this.sources.splice(i, 1)\n    }\n\n    const now = this.ctx.currentTime\n    // Behind (the first frame, or after an interruption): start a hair ahead of now so the first sample is not clipped.\n    if (this.nextTime < now) this.nextTime = now + 0.02\n    source.start(this.nextTime)\n    this.sources.push(source)\n    this.nextTime += buffer.duration\n  }\n\n  /** Stop everything queued, immediately: the caller started talking over the agent. */\n  interrupt(): void {\n    for (const s of this.sources) {\n      try {\n        s.stop(0)\n      } catch {\n        // already finished\n      }\n    }\n    this.sources = []\n    this.nextTime = 0\n    this.carry = null\n  }\n\n  /** Whether anything is queued or playing. */\n  active(): boolean {\n    return this.sources.length > 0\n  }\n\n  /** Milliseconds of audio still queued ahead of the audio clock. */\n  remainingMs(): number {\n    return Math.max(0, (this.nextTime - this.ctx.currentTime) * 1000)\n  }\n}\n","import { CAPTURE_CHUNK_SAMPLES, SAMPLE_RATE } from \"./protocol.js\"\n\nexport const WORKLET_NAME = \"dialgem-capture\"\n\n/**\n * The microphone capture processor, as source text.\n *\n * It runs in the audio thread at the device's own sample rate (44.1 or 48 kHz, whatever the hardware\n * does) and turns that into the 16 kHz mono 16-bit PCM the call needs, 640 samples at a time. Capturing at\n * the device rate and converting here, rather than asking the browser for a 16 kHz context, is what\n * keeps this working in Firefox, which refuses to connect a microphone to a context at another rate.\n *\n * Each output sample is the mean of the input samples that fall in its slot. That is a box filter: crude, but\n * it removes most of what would otherwise fold into the speech band, and speech recognition does not need more.\n * A rate that is not a whole multiple of 16 kHz (44.1 kHz) is handled by a fractional boundary, not by rounding.\n *\n * It is text, not an imported function, because `audioWorklet.addModule` takes a URL; the SDK serves it from a\n * Blob. The tests run this exact text, so what ships is what is tested.\n */\nexport const WORKLET_SOURCE = `\nclass DialgemCapture extends AudioWorkletProcessor {\n  constructor(options) {\n    super()\n    const o = (options && options.processorOptions) || {}\n    this.chunk = o.chunkSamples || ${CAPTURE_CHUNK_SAMPLES}\n    this.ratio = sampleRate / (o.targetRate || ${SAMPLE_RATE})\n    this.pos = 0\n    this.boundary = this.ratio\n    this.sum = 0\n    this.count = 0\n    this.last = 0\n    this.out = new Int16Array(this.chunk)\n    this.n = 0\n    this.peak = 0\n  }\n\n  emit(v) {\n    const a = v < 0 ? -v : v\n    if (a > this.peak) this.peak = a\n    const s = v > 1 ? 1 : v < -1 ? -1 : v\n    this.out[this.n++] = s < 0 ? s * 32768 : s * 32767\n    if (this.n === this.chunk) {\n      const pcm = this.out.buffer\n      this.port.postMessage({ pcm: pcm, peak: this.peak }, [pcm])\n      this.out = new Int16Array(this.chunk)\n      this.n = 0\n      this.peak = 0\n    }\n  }\n\n  process(inputs) {\n    const input = inputs[0] && inputs[0][0]\n    if (!input) return true\n    for (let i = 0; i < input.length; i++) {\n      this.sum += input[i]\n      this.count++\n      this.pos++\n      while (this.pos >= this.boundary) {\n        const v = this.count > 0 ? this.sum / this.count : this.last\n        this.last = v\n        this.sum = 0\n        this.count = 0\n        this.boundary += this.ratio\n        this.emit(v)\n      }\n    }\n    return true\n  }\n}\nregisterProcessor(${JSON.stringify(WORKLET_NAME)}, DialgemCapture)\n`\n","import { Emitter } from \"./emitter.js\"\nimport { PcmPlayer } from \"./player.js\"\nimport {\n  CAPTURE_CHUNK_SAMPLES,\n  DEFAULT_BASE_URL,\n  INIT_FRAME,\n  SAMPLE_RATE,\n  UrlError,\n  buildCallUrl,\n  describeClose,\n  describeRefusal,\n  parseServerFrame,\n} from \"./protocol.js\"\nimport { WORKLET_NAME, WORKLET_SOURCE } from \"./worklet.js\"\n\nexport type CallState = \"idle\" | \"connecting\" | \"active\" | \"ended\" | \"error\"\n\n/** How a call that started came to an end. */\nexport type EndReason =\n  /** `stop()` was called. */\n  | \"hangup\"\n  /** The agent ended the call, or a call limit (maximum length, silence) did. */\n  | \"agent_ended\"\n  /** The connection dropped. The call cannot be resumed: start a new one. */\n  | \"connection_lost\"\n\nexport type ErrorCode =\n  /** The browser lacks something the call needs (WebSocket, Web Audio, AudioWorklet, a microphone API). */\n  | \"unsupported\"\n  /** The visitor refused microphone access, or it is blocked for this page. */\n  | \"mic_denied\"\n  /** No microphone, or it could not be opened. */\n  | \"mic_unavailable\"\n  /** `accessToken` was a function and it threw or returned nothing. */\n  | \"token_unavailable\"\n  /** The token was invalid, expired or already used, or this page's origin is not allowed. */\n  | \"refused\"\n  /** The organization has too many calls running. */\n  | \"busy\"\n  /** The server declined the call; `reason` says why (for example `insufficient_credit`). */\n  | \"call_refused\"\n  | \"server_error\"\n  | \"connection_failed\"\n\nexport interface CallError {\n  code: ErrorCode\n  message: string\n  /** For `call_refused`: the server's reason. */\n  reason?: string | undefined\n}\n\nexport interface TranscriptEvent {\n  role: \"user\" | \"agent\"\n  text: string\n  /** False while the caller is still speaking (the words so far); true once the line is settled. */\n  final: boolean\n}\n\nexport interface CallEvents extends Record<string, unknown> {\n  state: CallState\n  started: { callId: string }\n  transcript: TranscriptEvent\n  agent_speaking: { speaking: boolean }\n  /** The caller talked over the agent and its speech was cut off. */\n  interrupted: Record<string, never>\n  /** Microphone level, 0 to 1, about ten times a second. Not emitted while muted. */\n  level: { level: number }\n  ended: { reason: EndReason }\n  error: CallError\n}\n\nexport interface MicrophoneOptions {\n  deviceId?: string | undefined\n  /** Default true: without it the agent's own voice comes back through the microphone and it answers itself. */\n  echoCancellation?: boolean | undefined\n  noiseSuppression?: boolean | undefined\n  autoGainControl?: boolean | undefined\n}\n\nexport interface DialgemCallOptions {\n  /**\n   * The access token from your backend's `POST /v1/calls/web`, or a function that returns it. A function\n   * is called **after** the visitor has allowed the microphone, because the token only lives 60 seconds and\n   * the permission prompt can take longer than that.\n   */\n  accessToken: string | (() => string | Promise<string>)\n  /** Only for a self-hosted deployment. Default `wss://api.dialgem.com`. */\n  baseURL?: string | undefined\n  microphone?: MicrophoneOptions | undefined\n  /** @internal Stand-ins for the browser, for tests. */\n  platform?: Partial<Platform> | undefined\n}\n\n/** What the call needs from the browser. */\nexport interface Platform {\n  WebSocket: typeof WebSocket\n  AudioContext: typeof AudioContext\n  AudioWorkletNode: typeof AudioWorkletNode\n  getUserMedia(constraints: MediaStreamConstraints): Promise<MediaStream>\n  createObjectURL(blob: Blob): string\n  revokeObjectURL(url: string): void\n  Blob: typeof Blob\n}\n\nconst LEVEL_INTERVAL_MS = 100\n/** A closing line is let finish playing, but never for longer than this. */\nconst MAX_DRAIN_MS = 10_000\n\nfunction browser(): Partial<Platform> {\n  const g = globalThis as unknown as Record<string, unknown>\n  const nav = g[\"navigator\"] as Navigator | undefined\n  const url = g[\"URL\"] as typeof URL | undefined\n  const out: Partial<Platform> = {}\n  if (g[\"WebSocket\"]) out.WebSocket = g[\"WebSocket\"] as Platform[\"WebSocket\"]\n  const audio = g[\"AudioContext\"] ?? g[\"webkitAudioContext\"]\n  if (audio) out.AudioContext = audio as Platform[\"AudioContext\"]\n  if (g[\"AudioWorkletNode\"]) out.AudioWorkletNode = g[\"AudioWorkletNode\"] as Platform[\"AudioWorkletNode\"]\n  if (g[\"Blob\"]) out.Blob = g[\"Blob\"] as Platform[\"Blob\"]\n  if (nav?.mediaDevices?.getUserMedia) out.getUserMedia = (c) => nav.mediaDevices.getUserMedia(c)\n  if (url?.createObjectURL) out.createObjectURL = (b) => url.createObjectURL(b)\n  if (url?.revokeObjectURL) out.revokeObjectURL = (u) => url.revokeObjectURL(u)\n  return out\n}\n\n/**\n * One conversation with one agent.\n *\n *   const call = new DialgemCall({ accessToken: () => fetch(\"/api/start-call\").then((r) => r.text()) })\n *   call.on(\"transcript\", (t) => ...)\n *   await call.start()      // from a click: browsers only let a page open the microphone and play sound after one\n *   ...\n *   call.stop()\n *\n * A call object is used once. Its token is single-use, so a second call needs a new object (and a new token).\n * There is no reconnecting: the conversation lives on the server for as long as the socket does, and a call\n * that drops has ended.\n */\nexport class DialgemCall extends Emitter<CallEvents> {\n  private _state: CallState = \"idle\"\n  private _muted = false\n  private _callId: string | null = null\n  private readonly platform: Platform\n\n  private ctx: AudioContext | null = null\n  private stream: MediaStream | null = null\n  private ws: WebSocket | null = null\n  private player: PcmPlayer | null = null\n  private source: MediaStreamAudioSourceNode | null = null\n  private node: AudioWorkletNode | null = null\n  private sink: GainNode | null = null\n  private moduleUrl: string | null = null\n  private speakingTimer: ReturnType<typeof setTimeout> | null = null\n  private drainTimer: ReturnType<typeof setTimeout> | null = null\n  private ending = false\n  private pendingRefusal: string | null = null\n  private failure: CallError | null = null\n  private lastLevelAt = 0\n  private speaking = false\n\n  constructor(private readonly options: DialgemCallOptions) {\n    super()\n    this.platform = { ...browser(), ...(options.platform ?? {}) } as Platform\n  }\n\n  get state(): CallState {\n    return this._state\n  }\n\n  /** Set once the call has started. */\n  get callId(): string | null {\n    return this._callId\n  }\n\n  get muted(): boolean {\n    return this._muted\n  }\n\n  /**\n   * Opens the microphone and the call. Call it directly from a click or tap: the browser decides whether the page\n   * may play sound at the moment this runs, and a call started from a timer or after other awaits can be silent.\n   * Resolves when the connection is open; the call has started when `started` fires. A failure is reported as an\n   * `error` event and also rejects.\n   */\n  async start(): Promise<void> {\n    if (this._state !== \"idle\") throw new Error(\"A call object can only be started once. Create a new DialgemCall for each call.\")\n    this.setState(\"connecting\")\n    const p = this.platform\n    if (!p.WebSocket || !p.AudioContext || !p.AudioWorkletNode || !p.getUserMedia || !p.createObjectURL || !p.Blob) {\n      throw this.fail({ code: \"unsupported\", message: \"This browser cannot run voice calls (it needs WebSocket, Web Audio with AudioWorklet, and a microphone API).\" })\n    }\n\n    // Synchronously, before any await: this is what a browser counts as \"started by the visitor\".\n    try {\n      this.ctx = new p.AudioContext()\n      void this.ctx.resume()\n    } catch {\n      throw this.fail({ code: \"unsupported\", message: \"This browser could not start audio.\" })\n    }\n    this.player = new PcmPlayer(this.ctx)\n\n    const mic = this.options.microphone ?? {}\n    try {\n      this.stream = await p.getUserMedia({\n        audio: {\n          channelCount: 1,\n          echoCancellation: mic.echoCancellation ?? true,\n          noiseSuppression: mic.noiseSuppression ?? true,\n          autoGainControl: mic.autoGainControl ?? true,\n          ...(mic.deviceId ? { deviceId: { exact: mic.deviceId } } : {}),\n        },\n      })\n    } catch (error) {\n      const name = (error as { name?: string } | null)?.name ?? \"\"\n      throw this.fail(\n        name === \"NotAllowedError\" || name === \"SecurityError\"\n          ? { code: \"mic_denied\", message: \"Microphone access was denied. Allow the microphone for this page and try again.\" }\n          : { code: \"mic_unavailable\", message: \"The microphone could not be opened. Check that one is connected and not in use by another app.\" },\n      )\n    }\n    if (this.dead()) return this.cleanup()\n\n    try {\n      this.moduleUrl = p.createObjectURL(new p.Blob([WORKLET_SOURCE], { type: \"application/javascript\" }))\n      await this.ctx.audioWorklet.addModule(this.moduleUrl)\n    } catch {\n      throw this.fail({\n        code: \"unsupported\",\n        message: \"The audio processor could not be loaded. If this page sets a Content-Security-Policy, allow blob: in script-src.\",\n      })\n    }\n    if (this.dead()) return this.cleanup()\n\n    // The token is fetched last: it has 60 seconds, and everything above can take a visitor longer than that.\n    let token: string\n    try {\n      const t = this.options.accessToken\n      token = typeof t === \"function\" ? await t() : t\n      if (!token) throw new Error(\"empty token\")\n    } catch {\n      throw this.fail({ code: \"token_unavailable\", message: \"Could not get an access token for the call.\" })\n    }\n    if (this.dead()) return this.cleanup()\n\n    let url: string\n    try {\n      url = buildCallUrl(this.options.baseURL ?? DEFAULT_BASE_URL, token)\n    } catch (error) {\n      throw this.fail({ code: \"connection_failed\", message: error instanceof UrlError ? error.message : \"Invalid call address.\" })\n    }\n\n    await new Promise<void>((resolve) => {\n      const ws = new p.WebSocket(url)\n      ws.binaryType = \"arraybuffer\"\n      this.ws = ws\n      ws.onopen = () => {\n        ws.send(INIT_FRAME)\n        resolve()\n      }\n      ws.onmessage = (event) => this.onMessage(event.data)\n      ws.onerror = () => resolve()\n      ws.onclose = (event) => {\n        resolve()\n        this.onClose(event.code)\n      }\n    })\n    if (this.failure) throw this.fail(this.failure)\n  }\n\n  /** Ends the call. Safe to call at any time, any number of times. */\n  stop(): void {\n    if (this._state === \"idle\") {\n      this.setState(\"ended\")\n      this.emit(\"ended\", { reason: \"hangup\" })\n      return\n    }\n    if (this._state !== \"active\" && this._state !== \"connecting\") return\n    try {\n      if (this.ws?.readyState === 1) this.ws.send(\"hangup\")\n    } catch {\n      // closing anyway\n    }\n    this.finish(\"hangup\")\n  }\n\n  /** While muted the agent hears silence. The call stays open. */\n  setMuted(muted: boolean): void {\n    this._muted = muted\n    if (muted) this.emit(\"level\", { level: 0 })\n  }\n\n  // ── server to client ──────────────────────────────────────────────────────────\n\n  private onMessage(data: unknown): void {\n    if (typeof data !== \"string\") {\n      if (data instanceof ArrayBuffer) this.onAudio(data)\n      return\n    }\n    const frame = parseServerFrame(data)\n    switch (frame.type) {\n      case \"session_started\":\n        this.onStarted(frame.sessionId)\n        break\n      case \"transcript\":\n        this.emit(\"transcript\", { role: frame.role, text: frame.text, final: true })\n        break\n      case \"transcript_partial\":\n        this.emit(\"transcript\", { role: \"user\", text: frame.text, final: false })\n        break\n      case \"agent_interrupted\":\n        this.player?.interrupt()\n        this.setSpeaking(false)\n        this.emit(\"interrupted\", {})\n        break\n      case \"call_ended\":\n        this.onCallEnded()\n        break\n      case \"error\":\n        // The close that follows carries the code; this carries the reason.\n        this.pendingRefusal = frame.reason\n        break\n      case \"ignored\":\n        break\n    }\n  }\n\n  private onStarted(sessionId: string): void {\n    if (this._state !== \"connecting\" || !this.ctx || !this.stream) return\n    this._callId = sessionId || null\n    this.setState(\"active\")\n\n    const ctx = this.ctx\n    this.source = ctx.createMediaStreamSource(this.stream)\n    this.node = new this.platform.AudioWorkletNode(ctx, WORKLET_NAME, {\n      numberOfInputs: 1,\n      numberOfOutputs: 1,\n      outputChannelCount: [1],\n      processorOptions: { targetRate: SAMPLE_RATE, chunkSamples: CAPTURE_CHUNK_SAMPLES },\n    })\n    // Muted to the speakers: the node only needs to be pulled by the graph.\n    this.sink = ctx.createGain()\n    this.sink.gain.value = 0\n    this.source.connect(this.node)\n    this.node.connect(this.sink)\n    this.sink.connect(ctx.destination)\n    this.node.port.onmessage = (event: MessageEvent<{ pcm: ArrayBuffer; peak: number }>) => this.onCaptured(event.data)\n\n    this.emit(\"started\", { callId: this._callId ?? \"\" })\n  }\n\n  private onCaptured(frame: { pcm: ArrayBuffer; peak: number }): void {\n    const ws = this.ws\n    if (!ws || ws.readyState !== 1 || this.ending) return\n    // Muted: silence of the same length, so the server's speech detection keeps its rhythm instead of seeing a gap.\n    ws.send(this._muted ? new ArrayBuffer(frame.pcm.byteLength) : frame.pcm)\n    const now = Date.now()\n    if (!this._muted && now - this.lastLevelAt >= LEVEL_INTERVAL_MS) {\n      this.lastLevelAt = now\n      this.emit(\"level\", { level: Math.min(1, frame.peak) })\n    }\n  }\n\n  private onAudio(data: ArrayBuffer): void {\n    if (this._state !== \"active\" || !this.player) return\n    this.player.feed(data)\n    this.setSpeaking(true)\n    // Speaking until the queued audio has played out, not for a fixed time after the last frame.\n    if (this.speakingTimer) clearTimeout(this.speakingTimer)\n    this.speakingTimer = setTimeout(() => this.setSpeaking(false), this.player.remainingMs() + 50)\n  }\n\n  private setSpeaking(speaking: boolean): void {\n    if (speaking === this.speaking) return\n    this.speaking = speaking\n    if (!speaking && this.speakingTimer) {\n      clearTimeout(this.speakingTimer)\n      this.speakingTimer = null\n    }\n    this.emit(\"agent_speaking\", { speaking })\n  }\n\n  private onCallEnded(): void {\n    if (this.ending || this.dead()) return\n    // Everything has been sent, but a goodbye can still be mid-sentence in the speaker: let it finish.\n    this.ending = true\n    const wait = Math.min(MAX_DRAIN_MS, (this.player?.remainingMs() ?? 0) + 50)\n    this.drainTimer = setTimeout(() => this.finish(\"agent_ended\"), wait)\n  }\n\n  private onClose(code: number): void {\n    if (this.dead()) return\n    if (this._state === \"active\") {\n      // A graceful end already has its own timer; a close during it must not cut the goodbye short.\n      if (!this.ending) this.finish(\"connection_lost\")\n      return\n    }\n    if (this.pendingRefusal) {\n      this.fail({ ...describeRefusal(this.pendingRefusal), reason: this.pendingRefusal })\n      return\n    }\n    this.fail(describeClose(code))\n  }\n\n  // ── ending ────────────────────────────────────────────────────────────────────\n\n  private dead(): boolean {\n    return this._state === \"ended\" || this._state === \"error\"\n  }\n\n  /** Ends the call as failed and reports it once. Returns the error, for `start()` to throw. */\n  private fail(error: CallError): Error {\n    if (!this.dead()) {\n      this.failure = error\n      this.cleanup()\n      this.setState(\"error\")\n      this.emit(\"error\", error)\n    }\n    const shown = this.failure ?? error\n    return Object.assign(new Error(shown.message), { code: shown.code, reason: shown.reason })\n  }\n\n  private finish(reason: EndReason): void {\n    if (this.dead()) return\n    this.cleanup()\n    this.setState(\"ended\")\n    this.emit(\"ended\", { reason })\n  }\n\n  private setState(state: CallState): void {\n    this._state = state\n    this.emit(\"state\", state)\n  }\n\n  private cleanup(): void {\n    this.ending = true\n    for (const t of [this.speakingTimer, this.drainTimer]) if (t) clearTimeout(t)\n    this.speakingTimer = this.drainTimer = null\n    if (this.speaking) {\n      this.speaking = false\n      this.emit(\"agent_speaking\", { speaking: false })\n    }\n    this.player?.interrupt()\n    if (this.node) {\n      this.node.port.onmessage = null\n      try {\n        this.node.disconnect()\n      } catch {\n        // already disconnected\n      }\n    }\n    try {\n      this.source?.disconnect()\n      this.sink?.disconnect()\n    } catch {\n      // already disconnected\n    }\n    this.stream?.getTracks().forEach((t) => t.stop())\n    if (this.ws) {\n      this.ws.onmessage = this.ws.onerror = this.ws.onopen = null\n      try {\n        this.ws.close()\n      } catch {\n        // already closed\n      }\n    }\n    if (this.moduleUrl) this.platform.revokeObjectURL?.(this.moduleUrl)\n    void this.ctx?.close().catch(() => undefined)\n    this.node = this.source = this.sink = this.stream = this.ctx = this.player = null\n    this.moduleUrl = null\n  }\n}\n","/** Kept equal to package.json's version by a test. */\nexport const VERSION = \"1.0.0\"\n"]}

package/dist/index.d.cts

@@ -0,0 +1,19 @@
+export { C as CallError, a as CallEvents, b as CallState, D as DialgemCall, c as DialgemCallOptions, E as EndReason, d as ErrorCode, M as MicrophoneOptions, T as TranscriptEvent } from './call-Cv2GrNfJ.cjs';
+
+/**
+ * What the call WebSocket speaks. Everything here is pure, so it is tested without a browser.
+ *
+ * Client to server: one JSON text frame to open the call, then binary frames of 16 kHz mono signed
+ * 16-bit little-endian PCM, and the text frame `hangup` to end it.
+ * Server to client: JSON text frames (events) and binary frames of 16 kHz mono signed 16-bit PCM
+ * (the agent's voice), each sentence beginning with a 44-byte RIFF header.
+ */
+/** The only address the SDK talks to unless a self-hosted deployment says otherwise. */
+declare const DEFAULT_BASE_URL = "wss://api.dialgem.com";
+/** Both directions of the call's audio. */
+declare const SAMPLE_RATE = 16000;
+
+/** Kept equal to package.json's version by a test. */
+declare const VERSION = "1.0.0";
+
+export { DEFAULT_BASE_URL, SAMPLE_RATE, VERSION };

package/dist/index.d.ts

@@ -0,0 +1,19 @@
+export { C as CallError, a as CallEvents, b as CallState, D as DialgemCall, c as DialgemCallOptions, E as EndReason, d as ErrorCode, M as MicrophoneOptions, T as TranscriptEvent } from './call-Cv2GrNfJ.js';
+
+/**
+ * What the call WebSocket speaks. Everything here is pure, so it is tested without a browser.
+ *
+ * Client to server: one JSON text frame to open the call, then binary frames of 16 kHz mono signed
+ * 16-bit little-endian PCM, and the text frame `hangup` to end it.
+ * Server to client: JSON text frames (events) and binary frames of 16 kHz mono signed 16-bit PCM
+ * (the agent's voice), each sentence beginning with a 44-byte RIFF header.
+ */
+/** The only address the SDK talks to unless a self-hosted deployment says otherwise. */
+declare const DEFAULT_BASE_URL = "wss://api.dialgem.com";
+/** Both directions of the call's audio. */
+declare const SAMPLE_RATE = 16000;
+
+/** Kept equal to package.json's version by a test. */
+declare const VERSION = "1.0.0";
+
+export { DEFAULT_BASE_URL, SAMPLE_RATE, VERSION };