agents 0.21.0 → 0.23.0

package/dist/voice/client.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"client.js","names":[],"sources":["../../src/voice/client.ts"],"sourcesContent":["import { PartySocket } from \"partysocket\";\nimport { ClientDiagnostics } from \"./diagnostics\";\nimport { toVoiceError } from \"./errors\";\nimport { VOICE_PROTOCOL_VERSION } from \"./types\";\n\nfunction camelCaseToKebabCase(str: string): string {\n  if (str === str.toUpperCase() && str !== str.toLowerCase()) {\n    return str.toLowerCase().replace(/_/g, \"-\");\n  }\n  let kebabified = str.replace(\n    /[A-Z]/g,\n    (letter) => `-${letter.toLowerCase()}`\n  );\n  kebabified = kebabified.startsWith(\"-\") ? kebabified.slice(1) : kebabified;\n  return kebabified.replace(/_/g, \"-\").replace(/-$/, \"\");\n}\nimport type {\n  VoiceStatus,\n  VoiceRole,\n  VoiceAudioFormat,\n  VoiceAudioInput,\n  VoiceTransport,\n  VoiceTransportCloseInfo,\n  VoiceError,\n  VoiceCompletionOutcome,\n  TranscriptMessage,\n  VoicePipelineMetrics,\n  VoiceTurnMetrics,\n  VoiceDiagnosticEvent\n} from \"./types\";\n\n// Re-export shared types for consumers importing from this module\nexport { VOICE_PROTOCOL_VERSION } from \"./types\";\nexport type {\n  VoiceStatus,\n  VoiceRole,\n  VoiceAudioFormat,\n  VoiceAudioInput,\n  VoiceTransport,\n  VoiceTransportCloseInfo,\n  VoiceError,\n  VoiceErrorCode,\n  VoiceErrorStage,\n  VoiceCompletionOutcome,\n  VoiceCompletionOutcomeCode,\n  VoiceModelFinishReason,\n  VoiceDiagnosticsOptions,\n  VoiceDiagnosticEvent,\n  VoicePipelineMetrics,\n  VoiceTurnMetrics,\n  VoiceTurnOutcome,\n  VoiceTurnSource,\n  TranscriptMessage\n} from \"./types\";\n\nexport interface VoiceClientOptions {\n  /** Agent name (matches the server-side Durable Object class). */\n  agent: string;\n  /** Instance name for the agent. @default \"default\" */\n  name?: string;\n\n  // Connection options (optional — defaults work for same-origin)\n  /** Host to connect to. @default window.location.host */\n  host?: string;\n\n  /** Query parameters appended to the WebSocket URL. */\n  query?: Record<string, string | null | undefined>;\n\n  /**\n   * Custom transport for sending/receiving data.\n   * Defaults to a WebSocket transport via PartySocket.\n   * Provide a custom implementation for WebRTC, SFU, or other transports.\n   */\n  transport?: VoiceTransport;\n\n  /**\n   * Custom audio input source. When provided, VoiceClient does NOT\n   * use its built-in AudioWorklet mic capture. The audio input is\n   * responsible for capturing and routing audio to the server.\n   * It must report audio levels via `onAudioLevel` for silence and\n   * interrupt detection to work.\n   */\n  audioInput?: VoiceAudioInput;\n\n  /**\n   * Preferred audio format for server responses. Sent in `start_call`\n   * as a hint — the server may ignore it if it cannot produce that format.\n   * The actual format is declared in the server's `audio_config` message.\n   */\n  preferredFormat?: VoiceAudioFormat;\n\n  // Tuning knobs with sensible defaults\n  /** RMS threshold below which audio is considered silence. @default 0.04 */\n  silenceThreshold?: number;\n  /** How long silence must last before sending end_of_speech (ms). @default 500 */\n  silenceDurationMs?: number;\n  /** RMS threshold for detecting user speech during agent playback. @default 0.05 */\n  interruptThreshold?: number;\n  /** Consecutive high-RMS chunks needed to trigger an interrupt. @default 2 */\n  interruptChunks?: number;\n  /** Maximum transcript messages to keep in memory. @default 200 */\n  maxTranscriptMessages?: number;\n\n  /**\n   * Preferred audio output device for assistant playback.\n   * Pass a MediaDeviceInfo.deviceId from an audiooutput device.\n   * Unsupported browsers continue playing through the default output.\n   * @default \"default\"\n   */\n  outputDeviceId?: string;\n}\n\ntype AudioElementWithSinkId = HTMLAudioElement & {\n  setSinkId?: (sinkId: string) => Promise<void>;\n};\n\nconst UNSUPPORTED_OUTPUT_DEVICE_ERROR =\n  \"Audio output device selection is not supported in this browser.\";\nconst OUTPUT_DEVICE_SWITCH_ERROR = \"Could not switch audio output device.\";\n\n/**\n * Low-level browser transport information for diagnosing connection failures.\n * Error causes stay as the raw `unknown` value because browsers do not\n * guarantee useful fields on WebSocket error events. The SDK does not\n * serialize or log this local event; consumers must narrow it before use.\n */\nexport type VoiceConnectionDiagnostic =\n  | { type: \"error\"; cause: unknown }\n  | ({ type: \"close\" } & VoiceTransportCloseInfo);\n\n/** Maps each event name to the data type passed to its listeners. */\nexport interface VoiceClientEventMap {\n  statuschange: VoiceStatus;\n  transcriptchange: TranscriptMessage[];\n  interimtranscript: string | null;\n  metricschange: VoicePipelineMetrics | null;\n  turnmetrics: VoiceTurnMetrics;\n  audiolevelchange: number;\n  connectionchange: boolean;\n  connectiondiagnostic: VoiceConnectionDiagnostic;\n  voiceerror: VoiceError;\n  completionoutcome: VoiceCompletionOutcome;\n  error: string | null;\n  outputdeviceerror: string | null;\n  mutechange: boolean;\n  custommessage: unknown;\n}\n\nexport type VoiceClientEvent = keyof VoiceClientEventMap;\n\n// --- Audio helpers (not exported) ---\n\nconst WORKLET_PROCESSOR = `\nclass AudioCaptureProcessor extends AudioWorkletProcessor {\n  constructor() {\n    super();\n    this.buffer = [];\n    this.sampleRate = sampleRate;\n    this.targetRate = 16000;\n    this.ratio = this.sampleRate / this.targetRate;\n  }\n\n  process(inputs) {\n    const input = inputs[0];\n    if (!input || !input[0]) return true;\n\n    const channelData = input[0];\n\n    // Linear interpolation resampling (e.g. 48kHz → 16kHz).\n    // Nearest-neighbor (picking every Nth sample) introduces aliasing\n    // artifacts, especially on sibilants (s, f, th). Linear interpolation\n    // blends adjacent samples, acting as a basic low-pass filter.\n    for (let i = 0; i < channelData.length; i += this.ratio) {\n      const idx = Math.floor(i);\n      const frac = i - idx;\n      if (idx + 1 < channelData.length) {\n        this.buffer.push(channelData[idx] * (1 - frac) + channelData[idx + 1] * frac);\n      } else if (idx < channelData.length) {\n        this.buffer.push(channelData[idx]);\n      }\n    }\n\n    if (this.buffer.length >= 1600) {\n      const chunk = new Float32Array(this.buffer);\n      this.port.postMessage({ type: 'audio', samples: chunk }, [chunk.buffer]);\n      this.buffer = [];\n    }\n\n    return true;\n  }\n}\n\nregisterProcessor('audio-capture-processor', AudioCaptureProcessor);\n`;\n\nfunction floatTo16BitPCM(samples: Float32Array): ArrayBuffer {\n  const buffer = new ArrayBuffer(samples.length * 2);\n  const view = new DataView(buffer);\n  for (let i = 0; i < samples.length; i++) {\n    const s = Math.max(-1, Math.min(1, samples[i]));\n    view.setInt16(i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n  }\n  return buffer;\n}\n\nfunction computeRMS(samples: Float32Array): number {\n  let sum = 0;\n  for (let i = 0; i < samples.length; i++) {\n    sum += samples[i] * samples[i];\n  }\n  return Math.sqrt(sum / samples.length);\n}\n\n// --- Default WebSocket transport ---\n\n/**\n * Default VoiceTransport backed by PartySocket (reconnecting WebSocket).\n * Created automatically when no custom transport is provided.\n */\nexport class WebSocketVoiceTransport implements VoiceTransport {\n  #socket: PartySocket | null = null;\n  #options: {\n    agent: string;\n    name?: string;\n    host?: string;\n    query?: Record<string, string | null | undefined>;\n  };\n\n  onopen: (() => void) | null = null;\n  onclose: ((info?: VoiceTransportCloseInfo) => void) | null = null;\n  onerror: ((error?: unknown) => void) | null = null;\n  onmessage: ((data: string | ArrayBuffer | Blob) => void) | null = null;\n\n  constructor(options: {\n    agent: string;\n    name?: string;\n    host?: string;\n    query?: Record<string, string | null | undefined>;\n  }) {\n    this.#options = options;\n  }\n\n  get connected(): boolean {\n    return this.#socket?.readyState === WebSocket.OPEN;\n  }\n\n  sendJSON(data: Record<string, unknown>): void {\n    if (this.#socket?.readyState === WebSocket.OPEN) {\n      this.#socket.send(JSON.stringify(data));\n    }\n  }\n\n  sendBinary(data: ArrayBuffer): void {\n    if (this.#socket?.readyState === WebSocket.OPEN) {\n      this.#socket.send(data);\n    }\n  }\n\n  connect(): void {\n    if (this.#socket) return;\n\n    const agentNamespace = camelCaseToKebabCase(this.#options.agent);\n\n    const socket = new PartySocket({\n      party: agentNamespace,\n      room: this.#options.name ?? \"default\",\n      host: this.#options.host ?? window.location.host,\n      prefix: \"agents\",\n      query: this.#options.query\n    });\n\n    socket.onopen = () => this.onopen?.();\n    socket.onclose = (event?: CloseEvent) =>\n      this.onclose?.(\n        event\n          ? {\n              code: event.code,\n              reason: event.reason,\n              wasClean: event.wasClean\n            }\n          : undefined\n      );\n    socket.onerror = (event) => this.onerror?.(event);\n    socket.onmessage = (event: MessageEvent) => {\n      this.onmessage?.(event.data);\n    };\n\n    this.#socket = socket;\n  }\n\n  disconnect(): void {\n    this.#socket?.close();\n    this.#socket = null;\n  }\n}\n\n// --- VoiceClient ---\n\n// oxlint-disable-next-line @typescript-eslint/no-explicit-any -- generic listener storage\ntype AnyListener = (data: any) => void;\n\nexport class VoiceClient {\n  // Internal state\n  #status: VoiceStatus = \"idle\";\n  #transcript: TranscriptMessage[] = [];\n  #metrics: VoicePipelineMetrics | null = null;\n  #turnMetrics: VoiceTurnMetrics | null = null;\n  #audioLevel = 0;\n  #isMuted = false;\n  #connected = false;\n  #error: string | null = null;\n  #outputDeviceError: string | null = null;\n  #lastCustomMessage: unknown = null;\n  #audioFormat: VoiceAudioFormat | null = null;\n  /** Sample rate for raw pcm16 payloads; set from server `audio_config`. */\n  #sampleRate = 16000;\n  #interimTranscript: string | null = null;\n  #serverProtocolVersion: number | null = null;\n  #inCall = false;\n  #callGeneration = 0;\n  #serverCallAcknowledged = false;\n  #diagnostics = new ClientDiagnostics();\n  #audioReceivedForCurrentTurn = false;\n  #diagnosticTurnId: string | null = null;\n\n  // Options (with defaults applied)\n  #silenceThreshold: number;\n  #silenceDurationMs: number;\n  #interruptThreshold: number;\n  #interruptChunks: number;\n  #maxTranscriptMessages: number;\n\n  // Transport\n  #transport: VoiceTransport | null = null;\n  #options: VoiceClientOptions;\n\n  // Audio refs\n  #audioContext: AudioContext | null = null;\n  #workletRegistered = false;\n  #workletNode: AudioWorkletNode | null = null;\n  #stream: MediaStream | null = null;\n  #silenceTimer: ReturnType<typeof setTimeout> | null = null;\n  #isSpeaking = false;\n  #playbackQueue: ArrayBuffer[] = [];\n  #isPlaying = false;\n  #isScheduling = false;\n  #scheduledSources = new Set<AudioBufferSourceNode>();\n  #playbackCursor = 0;\n  // End time (on the AudioContext clock) of the most recently scheduled chunk.\n  // Used to detect when the playback bridge has been idle across the gap\n  // between turns so it can be rebuilt (see #playAudio).\n  #lastPlaybackEnd: number | null = null;\n  #playbackElement: HTMLAudioElement | null = null;\n  #playbackDestination: MediaStreamAudioDestinationNode | null = null;\n  #playbackDestinationPromise: Promise<AudioNode> | null = null;\n  #useDefaultPlaybackDestination = false;\n  #outputDeviceId: string;\n  #outputDeviceSwitchGeneration = 0;\n  #playbackOutputGeneration = 0;\n  #playbackGeneration = 0;\n  #interruptChunkCount = 0;\n\n  // Event listeners\n  #listeners = new Map<VoiceClientEvent, Set<AnyListener>>();\n\n  constructor(options: VoiceClientOptions) {\n    this.#options = options;\n    this.#silenceThreshold = options.silenceThreshold ?? 0.04;\n    this.#silenceDurationMs = options.silenceDurationMs ?? 500;\n    this.#interruptThreshold = options.interruptThreshold ?? 0.05;\n    this.#interruptChunks = options.interruptChunks ?? 2;\n    this.#maxTranscriptMessages = options.maxTranscriptMessages ?? 200;\n    this.#outputDeviceId = options.outputDeviceId ?? \"default\";\n  }\n\n  // --- Public getters ---\n\n  get status(): VoiceStatus {\n    return this.#status;\n  }\n\n  get transcript(): TranscriptMessage[] {\n    return this.#transcript;\n  }\n\n  get metrics(): VoicePipelineMetrics | null {\n    return this.#metrics;\n  }\n\n  /** Last stable terminal summary received for any speech or text turn. */\n  get turnMetrics(): VoiceTurnMetrics | null {\n    return this.#turnMetrics;\n  }\n\n  get audioLevel(): number {\n    return this.#audioLevel;\n  }\n\n  get isMuted(): boolean {\n    return this.#isMuted;\n  }\n\n  get connected(): boolean {\n    return this.#connected;\n  }\n\n  get error(): string | null {\n    return this.#error;\n  }\n\n  get outputDeviceError(): string | null {\n    return this.#outputDeviceError;\n  }\n\n  /**\n   * The current interim (partial) transcript from streaming STT.\n   * Updates in real time as the user speaks. Cleared when the final\n   * transcript is produced or the call reaches a terminal reset.\n   * null when no interim text is available.\n   */\n  get interimTranscript(): string | null {\n    return this.#interimTranscript;\n  }\n\n  /**\n   * The protocol version reported by the server.\n   * null until the server sends its welcome message.\n   */\n  get serverProtocolVersion(): number | null {\n    return this.#serverProtocolVersion;\n  }\n\n  // --- Event system ---\n\n  addEventListener<K extends VoiceClientEvent>(\n    event: K,\n    listener: (data: VoiceClientEventMap[K]) => void\n  ): void {\n    let set = this.#listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.#listeners.set(event, set);\n    }\n    set.add(listener as AnyListener);\n  }\n\n  removeEventListener<K extends VoiceClientEvent>(\n    event: K,\n    listener: (data: VoiceClientEventMap[K]) => void\n  ): void {\n    this.#listeners.get(event)?.delete(listener as AnyListener);\n  }\n\n  #emit<K extends VoiceClientEvent>(\n    event: K,\n    data: VoiceClientEventMap[K]\n  ): void {\n    const set = this.#listeners.get(event);\n    if (set) {\n      for (const listener of set) {\n        listener(data);\n      }\n    }\n  }\n\n  #trimTranscript(): void {\n    if (this.#transcript.length > this.#maxTranscriptMessages) {\n      this.#transcript = this.#transcript.slice(-this.#maxTranscriptMessages);\n    }\n  }\n\n  #setOutputDeviceError(error: string | null): void {\n    if (this.#outputDeviceError === error) return;\n    this.#outputDeviceError = error;\n    this.#emit(\"outputdeviceerror\", error);\n  }\n\n  #clearInterimTranscript(): void {\n    this.#interimTranscript = null;\n    this.#emit(\"interimtranscript\", null);\n  }\n\n  // --- Connection ---\n\n  connect(): void {\n    if (this.#transport) return;\n\n    const transport =\n      this.#options.transport ??\n      new WebSocketVoiceTransport({\n        agent: this.#options.agent,\n        name: this.#options.name,\n        host: this.#options.host,\n        query: this.#options.query\n      });\n\n    transport.onopen = () => {\n      this.#connected = true;\n      this.#error = null;\n      // Announce our protocol version to the server\n      transport.sendJSON({\n        type: \"hello\",\n        protocol_version: VOICE_PROTOCOL_VERSION\n      });\n      this.#emit(\"connectionchange\", true);\n      this.#emit(\"error\", null);\n\n      // Reconnect recovery: if we were in a call when the connection\n      // dropped, re-establish it on the new connection. The mic is\n      // still running (not stopped on disconnect), so audio resumes\n      // flowing as soon as the server processes start_call.\n      if (this.#inCall) {\n        this.#serverCallAcknowledged = false;\n        transport.sendJSON({ type: \"start_call\" });\n      }\n    };\n\n    transport.onclose = (info) => {\n      this.#diagnostics.emit(\"connection.closed\", {\n        ...(info?.code === undefined ? {} : { code: info.code }),\n        clean: info?.wasClean ?? false\n      });\n      this.#connected = false;\n      this.#clearInterimTranscript();\n      this.#emit(\"connectionchange\", false);\n      this.#emit(\"connectiondiagnostic\", { type: \"close\", ...info });\n    };\n\n    transport.onerror = (cause) => {\n      this.#diagnostics.emit(\"connection.error\", {\n        error: toVoiceError(cause, \"Connection failed\")\n      });\n      this.#emit(\"connectiondiagnostic\", { type: \"error\", cause });\n      this.#error = \"Connection lost. Reconnecting...\";\n      this.#emit(\"error\", this.#error);\n    };\n\n    transport.onmessage = (data: string | ArrayBuffer | Blob) => {\n      if (typeof data === \"string\") {\n        this.#handleJSONMessage(data);\n      } else if (data instanceof Blob) {\n        data.arrayBuffer().then((buffer) => {\n          this.#recordAudioReceived(buffer.byteLength);\n          this.#playbackQueue.push(buffer);\n          this.#processPlaybackQueue();\n        });\n      } else if (data instanceof ArrayBuffer) {\n        this.#recordAudioReceived(data.byteLength);\n        this.#playbackQueue.push(data);\n        this.#processPlaybackQueue();\n      }\n    };\n\n    this.#transport = transport;\n    transport.connect();\n  }\n\n  disconnect(): void {\n    this.endCall();\n    this.#diagnostics.emit(\"connection.disconnecting\");\n    this.#transport?.disconnect();\n    this.#transport = null;\n    this.#connected = false;\n    this.#emit(\"connectionchange\", false);\n  }\n\n  // --- Public actions ---\n\n  async startCall(): Promise<void> {\n    if (!this.#transport?.connected) {\n      this.#error = \"Cannot start call: not connected. Call connect() first.\";\n      this.#emit(\"error\", this.#error);\n      return;\n    }\n    if (this.#inCall) return;\n\n    this.#clearInterimTranscript();\n    const callGeneration = ++this.#callGeneration;\n    this.#inCall = true;\n    this.#serverCallAcknowledged = false;\n    this.#error = null;\n    this.#metrics = null;\n    this.#emit(\"error\", null);\n    this.#emit(\"metricschange\", null);\n    const startMsg: Record<string, unknown> = { type: \"start_call\" };\n    if (this.#options.preferredFormat) {\n      startMsg.preferred_format = this.#options.preferredFormat;\n    }\n    this.#diagnostics.emit(\"call.starting\");\n    this.#transport.sendJSON(startMsg);\n    const ctx = await this.#getAudioContext();\n    if (this.#abortStaleCallStartup(callGeneration)) return;\n    await this.#getPlaybackDestination(ctx);\n    if (this.#abortStaleCallStartup(callGeneration)) return;\n    if (this.#options.audioInput) {\n      this.#diagnostics.emit(\"microphone.starting\", { source: \"custom\" });\n      this.#options.audioInput.onAudioLevel = (rms) =>\n        this.#processAudioLevel(rms);\n      this.#options.audioInput.onAudioData = (pcm) => {\n        if (this.#transport?.connected && !this.#isMuted) {\n          this.#transport.sendBinary(pcm);\n        }\n      };\n      await this.#options.audioInput.start();\n      this.#diagnostics.emit(\"microphone.ready\", { source: \"custom\" });\n    } else {\n      await this.#startMic();\n    }\n    if (!this.#abortStaleCallStartup(callGeneration)) {\n      this.#diagnostics.emit(\"call.local_ready\");\n    }\n  }\n\n  endCall(): void {\n    const wasInCall = this.#inCall;\n    this.#callGeneration++;\n    this.#inCall = false;\n    this.#serverCallAcknowledged = false;\n    if (this.#transport?.connected) {\n      this.#transport.sendJSON({ type: \"end_call\" });\n    }\n    this.#stopLocalCall();\n    this.#clearInterimTranscript();\n    this.#status = \"idle\";\n    this.#emit(\"statuschange\", \"idle\");\n    if (wasInCall) {\n      this.#diagnostics.emit(\"call.ended\", { reason: \"requested\" });\n    }\n  }\n\n  #isCurrentCallStartup(callGeneration: number): boolean {\n    return this.#inCall && this.#callGeneration === callGeneration;\n  }\n\n  #abortStaleCallStartup(callGeneration: number): boolean {\n    if (this.#isCurrentCallStartup(callGeneration)) return false;\n    if (!this.#inCall) this.#stopLocalCall();\n    return true;\n  }\n\n  #stopLocalCall(): void {\n    if (this.#options.audioInput) {\n      this.#options.audioInput.stop();\n      this.#options.audioInput.onAudioLevel = null;\n      this.#options.audioInput.onAudioData = null;\n    } else {\n      this.#stopMic();\n    }\n    this.#stopPlayback();\n    this.#closeAudioContext();\n    this.#resetDetection();\n  }\n\n  toggleMute(): void {\n    this.#isMuted = !this.#isMuted;\n\n    // Reset audio level so the UI shows silence while muted.\n    if (this.#isMuted) {\n      this.#audioLevel = 0;\n      this.#emit(\"audiolevelchange\", 0);\n    }\n\n    // If muting while speaking, flush the current utterance so the server\n    // processes accumulated audio instead of waiting forever (deadlock:\n    // muted → no audio frames → silence timer never starts → no end_of_speech).\n    if (this.#isMuted && this.#isSpeaking) {\n      this.#isSpeaking = false;\n      if (this.#silenceTimer) {\n        clearTimeout(this.#silenceTimer);\n        this.#silenceTimer = null;\n      }\n      if (this.#transport?.connected) {\n        this.#transport.sendJSON({ type: \"end_of_speech\" });\n      }\n    }\n\n    this.#emit(\"mutechange\", this.#isMuted);\n  }\n\n  /**\n   * Send a text message to the agent. The agent processes it through\n   * `onTurn()` (bypassing STT) and responds with text transcript and\n   * TTS audio (if in a call) or text-only (if not).\n   */\n  sendText(text: string): void {\n    if (this.#transport?.connected) {\n      this.#transport.sendJSON({ type: \"text_message\", text });\n    }\n  }\n\n  /**\n   * Send arbitrary JSON to the agent. Use this for app-level messages\n   * that are not part of the voice protocol (e.g. `{ type: \"kick_speaker\" }`).\n   * The server receives these in the consumer's `onMessage()` handler.\n   */\n  sendJSON(data: Record<string, unknown>): void {\n    if (this.#transport?.connected) {\n      this.#transport.sendJSON(data);\n    }\n  }\n\n  /**\n   * Set the preferred audio output device for assistant playback.\n   * Unsupported browsers continue playing through the default output.\n   */\n  async setOutputDevice(outputDeviceId?: string): Promise<void> {\n    this.#outputDeviceId = outputDeviceId ?? \"default\";\n    const generation = ++this.#outputDeviceSwitchGeneration;\n    if (this.#playbackElement) {\n      await this.#applyOutputDevice(this.#playbackElement, generation);\n    }\n  }\n\n  /**\n   * The last custom (non-voice-protocol) message received from the server.\n   * Listen for the `\"custommessage\"` event to be notified when this changes.\n   */\n  get lastCustomMessage(): unknown {\n    return this.#lastCustomMessage;\n  }\n\n  /**\n   * The audio format the server declared for binary payloads.\n   * Set when the server sends `audio_config` at call start.\n   */\n  get audioFormat(): VoiceAudioFormat | null {\n    return this.#audioFormat;\n  }\n\n  /**\n   * The sample rate (Hz) the server declared for raw pcm16 payloads.\n   * Set when the server sends `audio_config` at call start. Defaults to 16000.\n   */\n  get sampleRate(): number {\n    return this.#sampleRate;\n  }\n\n  // --- Voice protocol handler ---\n\n  #recordAudioReceived(bytes: number): void {\n    if (this.#audioReceivedForCurrentTurn) return;\n    this.#audioReceivedForCurrentTurn = true;\n    this.#diagnostics.emit(\"audio.received\", {\n      bytes,\n      ...(this.#diagnosticTurnId === null\n        ? {}\n        : { turn_id: this.#diagnosticTurnId })\n    });\n  }\n\n  #handleJSONMessage(data: string): void {\n    let msg: Record<string, unknown>;\n    try {\n      msg = JSON.parse(data);\n    } catch {\n      // Not JSON — ignore (e.g. state sync binary frames)\n      return;\n    }\n\n    switch (msg.type) {\n      case \"welcome\": {\n        const diagnosticOptions =\n          typeof msg.diagnostics === \"object\" && msg.diagnostics !== null\n            ? (msg.diagnostics as Record<string, unknown>)\n            : null;\n        const diagnosticsEnabled = diagnosticOptions?.browser_console === true;\n        this.#diagnostics.setEnabled(diagnosticsEnabled);\n        if (diagnosticsEnabled) {\n          this.#diagnostics.emit(\"connection.ready\", {\n            protocol_version: msg.protocol_version as number\n          });\n        }\n        this.#serverProtocolVersion = msg.protocol_version as number;\n        if (msg.protocol_version !== VOICE_PROTOCOL_VERSION) {\n          console.warn(\n            `[VoiceClient] Protocol version mismatch: client=${VOICE_PROTOCOL_VERSION}, server=${msg.protocol_version}`\n          );\n        }\n        break;\n      }\n      case \"diagnostic\":\n        if (\n          typeof msg.event === \"string\" &&\n          typeof msg.timestamp === \"number\" &&\n          Number.isFinite(msg.timestamp)\n        ) {\n          const data =\n            typeof msg.data === \"object\" && msg.data !== null\n              ? (msg.data as VoiceDiagnosticEvent[\"data\"])\n              : undefined;\n          if (\n            msg.event === \"audio.first_sent\" &&\n            typeof data?.turn_id === \"string\"\n          ) {\n            this.#diagnosticTurnId = data.turn_id;\n            this.#audioReceivedForCurrentTurn = false;\n          }\n          this.#diagnostics.receive({\n            event: msg.event,\n            timestamp: msg.timestamp,\n            ...(data === undefined ? {} : { data })\n          });\n        }\n        break;\n      case \"audio_config\":\n        this.#serverCallAcknowledged = true;\n        this.#audioFormat = msg.format as VoiceAudioFormat;\n        this.#sampleRate =\n          typeof msg.sampleRate === \"number\" && msg.sampleRate > 0\n            ? msg.sampleRate\n            : 16000;\n        break;\n      case \"status\":\n        this.#status = msg.status as VoiceStatus;\n        if (msg.status === \"thinking\" || msg.status === \"speaking\") {\n          this.#audioReceivedForCurrentTurn = false;\n          if (msg.status === \"thinking\") this.#diagnosticTurnId = null;\n        }\n        if (msg.status === \"idle\" && this.#inCall) {\n          const shouldEndLocalCall =\n            this.#serverCallAcknowledged || this.#error !== null;\n          if (!shouldEndLocalCall) {\n            this.#emit(\"statuschange\", this.#status);\n            break;\n          }\n          this.#callGeneration++;\n          this.#inCall = false;\n          this.#serverCallAcknowledged = false;\n          this.#stopLocalCall();\n          this.#clearInterimTranscript();\n        }\n        if (msg.status === \"listening\") {\n          this.#serverCallAcknowledged = true;\n          this.#error = null;\n          this.#emit(\"error\", null);\n        } else if (msg.status === \"thinking\" || msg.status === \"speaking\") {\n          this.#serverCallAcknowledged = true;\n        }\n        this.#emit(\"statuschange\", this.#status);\n        break;\n      case \"transcript_interim\":\n        this.#interimTranscript = msg.text as string;\n        this.#emit(\"interimtranscript\", this.#interimTranscript);\n        break;\n      case \"playback_interrupt\":\n        this.#stopPlayback(\"server_interrupt\");\n        break;\n      case \"transcript\":\n        // Final transcript arrived — clear interim\n        this.#interimTranscript = null;\n        this.#emit(\"interimtranscript\", null);\n\n        // New user utterance while agent is playing -- stop playback.\n        // With continuous STT, the model detects the user's speech\n        // server-side, so this arrives before the client's interrupt\n        // detection fires.\n        if (msg.role === \"user\" && this.#isPlaying) {\n          this.#stopPlayback(\"user_utterance\");\n        }\n\n        this.#transcript = [\n          ...this.#transcript,\n          {\n            role: msg.role as VoiceRole,\n            text: msg.text as string,\n            timestamp: Date.now()\n          }\n        ];\n        this.#trimTranscript();\n        this.#emit(\"transcriptchange\", this.#transcript);\n        break;\n      case \"transcript_start\":\n        this.#transcript = [\n          ...this.#transcript,\n          { role: \"assistant\", text: \"\", timestamp: Date.now() }\n        ];\n        this.#trimTranscript();\n        this.#emit(\"transcriptchange\", this.#transcript);\n        break;\n      case \"transcript_delta\": {\n        if (this.#transcript.length === 0) break;\n        const updated = [...this.#transcript];\n        const last = updated[updated.length - 1];\n        if (last.role === \"assistant\") {\n          updated[updated.length - 1] = {\n            ...last,\n            text: last.text + (msg.text as string)\n          };\n          this.#transcript = updated;\n          this.#emit(\"transcriptchange\", this.#transcript);\n        }\n        break;\n      }\n      case \"transcript_end\": {\n        if (this.#transcript.length === 0) break;\n        const updated = [...this.#transcript];\n        const last = updated[updated.length - 1];\n        if (last.role === \"assistant\") {\n          updated[updated.length - 1] = {\n            ...last,\n            text: msg.text as string\n          };\n          this.#transcript = updated;\n          this.#emit(\"transcriptchange\", this.#transcript);\n        }\n        break;\n      }\n      case \"metrics\":\n        this.#metrics = {\n          llm_ms: msg.llm_ms as number,\n          tts_ms: msg.tts_ms as number,\n          first_audio_ms: msg.first_audio_ms as number,\n          total_ms: msg.total_ms as number\n        };\n        this.#emit(\"metricschange\", this.#metrics);\n        break;\n      case \"turn_metrics\": {\n        const { type: _type, ...payload } = msg;\n        const metrics = payload as unknown as VoiceTurnMetrics;\n        this.#turnMetrics = metrics;\n        this.#emit(\"turnmetrics\", metrics);\n        break;\n      }\n      case \"completion_outcome\": {\n        const outcome: VoiceCompletionOutcome = {\n          code: msg.code as VoiceCompletionOutcome[\"code\"],\n          stage: \"llm\",\n          ...(msg.finishReason !== undefined\n            ? {\n                finishReason:\n                  msg.finishReason as VoiceCompletionOutcome[\"finishReason\"]\n              }\n            : {}),\n          partialOutput: msg.partialOutput as boolean\n        };\n        this.#emit(\"completionoutcome\", outcome);\n        break;\n      }\n      case \"error\": {\n        const voiceError: VoiceError = {\n          message: msg.message as string,\n          ...(msg.code !== undefined\n            ? { code: msg.code as VoiceError[\"code\"] }\n            : {}),\n          ...(msg.stage !== undefined\n            ? { stage: msg.stage as VoiceError[\"stage\"] }\n            : {}),\n          ...(msg.retryable !== undefined\n            ? { retryable: msg.retryable as boolean }\n            : {})\n        };\n        this.#error = voiceError.message;\n        this.#emit(\"voiceerror\", voiceError);\n        this.#emit(\"error\", this.#error);\n        break;\n      }\n      default:\n        // App-level custom message — surface via event\n        this.#lastCustomMessage = msg;\n        this.#emit(\"custommessage\", msg);\n        break;\n    }\n  }\n\n  // --- Audio context management ---\n\n  /** Get or create the shared AudioContext. */\n  async #getAudioContext(): Promise<AudioContext> {\n    if (!this.#audioContext) {\n      this.#audioContext = new AudioContext({ sampleRate: 48000 });\n    }\n    if (this.#audioContext.state === \"suspended\") {\n      await this.#audioContext.resume();\n    }\n    return this.#audioContext;\n  }\n\n  /** Close the AudioContext and release resources. */\n  #closeAudioContext(): void {\n    if (this.#audioContext) {\n      this.#closePlaybackOutput();\n      this.#audioContext.close().catch(() => {});\n      this.#audioContext = null;\n      this.#workletRegistered = false;\n    }\n  }\n\n  async #getPlaybackDestination(ctx: AudioContext): Promise<AudioNode> {\n    if (this.#playbackDestinationPromise) {\n      return this.#playbackDestinationPromise;\n    }\n    if (this.#playbackDestination) return this.#playbackDestination;\n    if (this.#useDefaultPlaybackDestination) return ctx.destination;\n\n    const outputGeneration = this.#playbackOutputGeneration;\n    const promise = this.#initializePlaybackDestination(ctx, outputGeneration);\n    this.#playbackDestinationPromise = promise;\n    try {\n      return await promise;\n    } finally {\n      if (this.#playbackDestinationPromise === promise) {\n        this.#playbackDestinationPromise = null;\n      }\n    }\n  }\n\n  async #initializePlaybackDestination(\n    ctx: AudioContext,\n    outputGeneration: number\n  ): Promise<AudioNode> {\n    try {\n      const destination = ctx.createMediaStreamDestination();\n      const audio = new Audio();\n      audio.autoplay = true;\n      audio.srcObject = destination.stream;\n      this.#playbackElement = audio;\n      this.#playbackDestination = destination;\n      await this.#applyOutputDevice(audio, this.#outputDeviceSwitchGeneration);\n      if (!this.#isCurrentPlaybackOutput(audio, outputGeneration)) {\n        this.#releasePlaybackElement(audio);\n        return ctx.destination;\n      }\n      await audio.play();\n      if (!this.#isCurrentPlaybackOutput(audio, outputGeneration)) {\n        this.#releasePlaybackElement(audio);\n        return ctx.destination;\n      }\n      return destination;\n    } catch (err) {\n      console.warn(\n        \"[VoiceClient] HTMLAudioElement playback output unavailable; using default AudioContext destination.\",\n        err\n      );\n      this.#closePlaybackOutput();\n      this.#useDefaultPlaybackDestination = true;\n      return ctx.destination;\n    }\n  }\n\n  #isCurrentPlaybackOutput(\n    audio: HTMLAudioElement,\n    outputGeneration: number\n  ): boolean {\n    return (\n      this.#playbackElement === audio &&\n      this.#playbackOutputGeneration === outputGeneration\n    );\n  }\n\n  async #applyOutputDevice(\n    audio: HTMLAudioElement,\n    generation: number\n  ): Promise<void> {\n    const sinkId = this.#outputDeviceId;\n    const setSinkId = (audio as AudioElementWithSinkId).setSinkId;\n    if (!setSinkId) {\n      if (sinkId === \"default\") {\n        this.#setOutputDeviceError(null);\n        return;\n      }\n      this.#setOutputDeviceError(UNSUPPORTED_OUTPUT_DEVICE_ERROR);\n      return;\n    }\n\n    try {\n      await setSinkId.call(audio, sinkId);\n      if (\n        generation !== this.#outputDeviceSwitchGeneration ||\n        sinkId !== this.#outputDeviceId\n      ) {\n        if (this.#playbackElement === audio) {\n          await this.#applyOutputDevice(\n            audio,\n            this.#outputDeviceSwitchGeneration\n          );\n        }\n        return;\n      }\n      if (\n        this.#outputDeviceError === UNSUPPORTED_OUTPUT_DEVICE_ERROR ||\n        this.#outputDeviceError === OUTPUT_DEVICE_SWITCH_ERROR\n      ) {\n        this.#setOutputDeviceError(null);\n      }\n    } catch {\n      if (\n        generation !== this.#outputDeviceSwitchGeneration ||\n        sinkId !== this.#outputDeviceId\n      ) {\n        if (this.#playbackElement === audio) {\n          await this.#applyOutputDevice(\n            audio,\n            this.#outputDeviceSwitchGeneration\n          );\n        }\n        return;\n      }\n      this.#setOutputDeviceError(OUTPUT_DEVICE_SWITCH_ERROR);\n    }\n  }\n\n  #closePlaybackOutput(): void {\n    this.#playbackOutputGeneration++;\n    if (this.#playbackElement) {\n      this.#releasePlaybackElement(this.#playbackElement);\n      this.#playbackElement = null;\n    }\n    this.#playbackDestination = null;\n    this.#playbackDestinationPromise = null;\n    this.#useDefaultPlaybackDestination = false;\n  }\n\n  #releasePlaybackElement(audio: HTMLAudioElement): void {\n    audio.pause();\n    audio.srcObject = null;\n  }\n\n  // --- Audio playback ---\n\n  async #playAudio(audioData: ArrayBuffer, generation: number): Promise<void> {\n    try {\n      const ctx = await this.#getAudioContext();\n\n      let audioBuffer: AudioBuffer;\n      if (this.#audioFormat === \"pcm16\") {\n        // Raw 16-bit LE mono PCM — sample rate comes from server audio_config\n        const int16 = new Int16Array(audioData);\n        audioBuffer = ctx.createBuffer(1, int16.length, this.#sampleRate);\n        const channel = audioBuffer.getChannelData(0);\n        for (let i = 0; i < int16.length; i++) {\n          channel[i] = int16[i] / 32768;\n        }\n      } else {\n        // mp3, wav, opus — let the browser decode\n        audioBuffer = await ctx.decodeAudioData(audioData.slice(0));\n      }\n      if (generation !== this.#playbackGeneration) return;\n\n      // Assistant playback is routed through a MediaStreamAudioDestinationNode\n      // -> HTMLAudioElement bridge (so a selected output device can be honored\n      // via HTMLMediaElement.setSinkId). That bridge is a live playout path:\n      // when it is reused for a new turn after sitting idle through the gap\n      // between turns, the element resumes the fresh burst at the wrong rate and\n      // the new turn plays back noticeably slow (it then re-converges to normal\n      // over the turn). A freshly created element does not show this, so once\n      // the bridge has fully drained and been idle past a short threshold, tear\n      // it down; #getPlaybackDestination rebuilds a fresh element for this turn.\n      // The scheduledSources.size === 0 guard means this never fires mid-turn:\n      // chunks within a turn are scheduled ahead on the cursor, keeping at least\n      // one source live until the turn finishes.\n      const BRIDGE_IDLE_REBUILD_SECONDS = 0.3;\n      if (\n        this.#playbackElement &&\n        this.#scheduledSources.size === 0 &&\n        this.#lastPlaybackEnd !== null &&\n        ctx.currentTime - this.#lastPlaybackEnd > BRIDGE_IDLE_REBUILD_SECONDS\n      ) {\n        this.#closePlaybackOutput();\n      }\n\n      const destination = await this.#getPlaybackDestination(ctx);\n      if (generation !== this.#playbackGeneration) return;\n\n      const source = ctx.createBufferSource();\n      source.buffer = audioBuffer;\n      source.connect(destination);\n      this.#scheduledSources.add(source);\n      source.onended = () => {\n        this.#scheduledSources.delete(source);\n        if (\n          generation === this.#playbackGeneration &&\n          !this.#isScheduling &&\n          this.#scheduledSources.size === 0 &&\n          this.#playbackQueue.length === 0\n        ) {\n          this.#isPlaying = false;\n          this.#diagnostics.emit(\"playback.completed\");\n        }\n      };\n\n      // Schedule on the audio clock, butted against the previous chunk.\n      // Starting at currentTime only after the previous chunk's `ended`\n      // event leaves a few ms of silence at every chunk seam, audible as\n      // a periodic click during agent speech.\n      const startAt = Math.max(ctx.currentTime, this.#playbackCursor);\n      this.#playbackCursor = startAt + audioBuffer.duration;\n      this.#lastPlaybackEnd = this.#playbackCursor;\n      source.start(startAt);\n    } catch (err) {\n      this.#diagnostics.emit(\"playback.error\", {\n        error: toVoiceError(err, \"Playback failed\")\n      });\n      console.error(\"[VoiceClient] Audio playback error:\", err);\n    }\n  }\n\n  async #processPlaybackQueue(): Promise<void> {\n    if (this.#isScheduling || this.#playbackQueue.length === 0) return;\n    this.#isScheduling = true;\n    this.#isPlaying = true;\n    this.#diagnostics.emit(\"playback.started\", {\n      chunks: this.#playbackQueue.length\n    });\n    const generation = this.#playbackGeneration;\n\n    while (\n      generation === this.#playbackGeneration &&\n      this.#playbackQueue.length > 0\n    ) {\n      const audioData = this.#playbackQueue.shift()!;\n      await this.#playAudio(audioData, generation);\n    }\n\n    if (generation === this.#playbackGeneration) {\n      this.#isScheduling = false;\n      if (this.#scheduledSources.size === 0) {\n        this.#isPlaying = false;\n      }\n    }\n  }\n\n  #stopPlayback(reason = \"stopped\"): void {\n    const wasActive =\n      this.#isPlaying ||\n      this.#isScheduling ||\n      this.#playbackQueue.length > 0 ||\n      this.#scheduledSources.size > 0;\n    this.#playbackGeneration++;\n    const sources = [...this.#scheduledSources];\n    this.#scheduledSources.clear();\n    for (const source of sources) {\n      try {\n        source.stop();\n      } catch {\n        // The source may already have ended or been stopped by another signal.\n      }\n    }\n    this.#playbackQueue = [];\n    this.#isPlaying = false;\n    this.#isScheduling = false;\n    this.#playbackCursor = 0;\n    // An interrupt (barge-in or a user transcript) stops playback abruptly, so\n    // the bridge goes idle from now, not from the cut chunk's scheduled end.\n    // Record the current audio-clock time as the idle start so the next turn's\n    // rebuild check in #playAudio still fires after interrupt -> long pause ->\n    // new turn. Nulling this here would disable that check and let the slow-\n    // playback symptom reappear on the post-interrupt turn.\n    this.#lastPlaybackEnd = this.#audioContext\n      ? this.#audioContext.currentTime\n      : null;\n    if (wasActive) {\n      this.#diagnostics.emit(\"playback.stopped\", { reason });\n    }\n  }\n\n  // --- Mic capture ---\n\n  async #startMic(): Promise<void> {\n    this.#diagnostics.emit(\"microphone.starting\", { source: \"browser\" });\n    try {\n      const stream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          sampleRate: { ideal: 48000 },\n          channelCount: 1,\n          echoCancellation: true,\n          noiseSuppression: true,\n          autoGainControl: true\n        }\n      });\n      this.#stream = stream;\n\n      const ctx = await this.#getAudioContext();\n\n      // Only register the worklet processor once per AudioContext.\n      // Calling addModule twice with the same processor name throws.\n      if (!this.#workletRegistered) {\n        const blob = new Blob([WORKLET_PROCESSOR], {\n          type: \"application/javascript\"\n        });\n        const workletUrl = URL.createObjectURL(blob);\n        await ctx.audioWorklet.addModule(workletUrl);\n        URL.revokeObjectURL(workletUrl);\n        this.#workletRegistered = true;\n      }\n\n      const source = ctx.createMediaStreamSource(stream);\n      const workletNode = new AudioWorkletNode(ctx, \"audio-capture-processor\");\n      this.#workletNode = workletNode;\n\n      workletNode.port.onmessage = (event: MessageEvent) => {\n        if (event.data.type === \"audio\" && !this.#isMuted) {\n          const samples = event.data.samples as Float32Array;\n          const rms = computeRMS(samples);\n\n          // Send PCM to agent\n          const pcm = floatTo16BitPCM(samples);\n          if (this.#transport?.connected) {\n            this.#transport.sendBinary(pcm);\n          }\n\n          this.#processAudioLevel(rms);\n        }\n      };\n\n      source.connect(workletNode);\n      workletNode.connect(ctx.destination);\n      this.#diagnostics.emit(\"microphone.ready\", { source: \"browser\" });\n    } catch (err) {\n      this.#diagnostics.emit(\"microphone.error\", {\n        error: toVoiceError(err, \"Microphone failed\")\n      });\n      console.error(\"[VoiceClient] Mic error:\", err);\n      this.#error =\n        \"Microphone access denied. Please allow microphone access and try again.\";\n      this.#emit(\"error\", this.#error);\n    }\n  }\n\n  #stopMic(): void {\n    const wasActive = this.#workletNode !== null || this.#stream !== null;\n    this.#workletNode?.disconnect();\n    this.#workletNode = null;\n    this.#stream?.getTracks().forEach((track) => track.stop());\n    this.#stream = null;\n    if (wasActive) this.#diagnostics.emit(\"microphone.stopped\");\n    this.#resetDetection();\n  }\n\n  // --- Audio level processing (shared between built-in mic and custom audioInput) ---\n\n  #processAudioLevel(rms: number): void {\n    // When muted, ignore incoming audio levels. This prevents false\n    // speech detection when a custom audioInput keeps reporting levels.\n    // The built-in mic already gates on !#isMuted before calling here,\n    // but audioInput implementations don't know about mute state.\n    if (this.#isMuted) return;\n\n    this.#audioLevel = rms;\n    this.#emit(\"audiolevelchange\", rms);\n\n    // Interruption detection: user speaking during agent playback\n    if (this.#isPlaying && rms > this.#interruptThreshold) {\n      this.#interruptChunkCount++;\n      if (this.#interruptChunkCount >= this.#interruptChunks) {\n        this.#diagnostics.emit(\"speech.interrupt_detected\");\n        this.#stopPlayback(\"speech_interrupt\");\n        this.#interruptChunkCount = 0;\n        if (this.#transport?.connected) {\n          this.#transport.sendJSON({ type: \"interrupt\" });\n        }\n      }\n    } else {\n      this.#interruptChunkCount = 0;\n    }\n\n    // Silence detection\n    if (rms > this.#silenceThreshold) {\n      if (!this.#isSpeaking) {\n        this.#isSpeaking = true;\n        this.#diagnostics.emit(\"speech.started\");\n        // Notify server that speech started (for streaming STT)\n        if (this.#transport?.connected) {\n          this.#transport.sendJSON({ type: \"start_of_speech\" });\n        }\n      }\n      if (this.#silenceTimer) {\n        clearTimeout(this.#silenceTimer);\n        this.#silenceTimer = null;\n      }\n    } else if (this.#isSpeaking) {\n      if (!this.#silenceTimer) {\n        this.#silenceTimer = setTimeout(() => {\n          this.#isSpeaking = false;\n          this.#silenceTimer = null;\n          this.#diagnostics.emit(\"speech.ended\");\n          if (this.#transport?.connected) {\n            this.#transport.sendJSON({ type: \"end_of_speech\" });\n          }\n        }, this.#silenceDurationMs);\n      }\n    }\n  }\n\n  #resetDetection(): void {\n    if (this.#silenceTimer) {\n      clearTimeout(this.#silenceTimer);\n      this.#silenceTimer = null;\n    }\n    this.#isSpeaking = false;\n    this.#interruptChunkCount = 0;\n    this.#audioLevel = 0;\n    this.#emit(\"audiolevelchange\", 0);\n  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package/dist/voice/errors.d.ts

@@ -0,0 +1,43 @@
+//#region src/voice/errors.d.ts
+interface VoiceProviderErrorOptions {
+  code?: string | number;
+  status?: number;
+  closeCode?: number;
+  closeReason?: string;
+  wasClean?: boolean;
+}
+/**
+ * Error raised at a voice provider boundary. Only known, content-free provider
+ * metadata belongs here. Provider response bodies and raw events must stay out.
+ */
+declare class VoiceProviderError extends Error {
+  readonly code?: string | number;
+  readonly status?: number;
+  readonly closeCode?: number;
+  readonly closeReason?: string;
+  readonly wasClean?: boolean;
+  constructor(message: string, options?: VoiceProviderErrorOptions);
+}
+interface VoiceErrorLogOptions {
+  component: string;
+  stage: string;
+  message: string;
+  error: Error;
+  connectionId?: string;
+}
+/** Convert a value caught from an external boundary without inspecting it. */
+declare function toVoiceError(error: unknown, fallback: string): Error;
+/** Return a bounded direct message suitable for the existing string wire API. */
+declare function voiceErrorMessage(error: Error, fallback: string): string;
+/** Emit one structured object so Workers Logs can extract diagnostic fields. */
+declare function logVoiceError(options: VoiceErrorLogOptions): void;
+//#endregion
+export {
+  VoiceErrorLogOptions,
+  VoiceProviderError,
+  VoiceProviderErrorOptions,
+  logVoiceError,
+  toVoiceError,
+  voiceErrorMessage
+};
+//# sourceMappingURL=errors.d.ts.map

package/dist/voice/errors.js

@@ -0,0 +1,41 @@
+//#region src/voice/errors.ts
+const MAX_CLIENT_MESSAGE_LENGTH = 300;
+/**
+* Error raised at a voice provider boundary. Only known, content-free provider
+* metadata belongs here. Provider response bodies and raw events must stay out.
+*/
+var VoiceProviderError = class extends Error {
+	constructor(message, options = {}) {
+		super(message);
+		this.name = "VoiceProviderError";
+		if (options.code !== void 0) this.code = options.code;
+		if (options.status !== void 0) this.status = options.status;
+		if (options.closeCode !== void 0) this.closeCode = options.closeCode;
+		if (options.closeReason !== void 0) this.closeReason = options.closeReason;
+		if (options.wasClean !== void 0) this.wasClean = options.wasClean;
+	}
+};
+/** Convert a value caught from an external boundary without inspecting it. */
+function toVoiceError(error, fallback) {
+	return error instanceof Error ? error : new Error(fallback);
+}
+/** Return a bounded direct message suitable for the existing string wire API. */
+function voiceErrorMessage(error, fallback) {
+	const message = error.message.trim();
+	if (!message || message === "[object Object]") return fallback;
+	return message.slice(0, MAX_CLIENT_MESSAGE_LENGTH);
+}
+/** Emit one structured object so Workers Logs can extract diagnostic fields. */
+function logVoiceError(options) {
+	console.error({
+		component: options.component,
+		stage: options.stage,
+		message: options.message,
+		...options.connectionId ? { connectionId: options.connectionId } : {},
+		error: options.error
+	});
+}
+//#endregion
+export { VoiceProviderError, logVoiceError, toVoiceError, voiceErrorMessage };
+
+//# sourceMappingURL=errors.js.map

package/dist/voice/errors.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"errors.js","names":[],"sources":["../../src/voice/errors.ts"],"sourcesContent":["const MAX_CLIENT_MESSAGE_LENGTH = 300;\n\nexport interface VoiceProviderErrorOptions {\n  code?: string | number;\n  status?: number;\n  closeCode?: number;\n  closeReason?: string;\n  wasClean?: boolean;\n}\n\n/**\n * Error raised at a voice provider boundary. Only known, content-free provider\n * metadata belongs here. Provider response bodies and raw events must stay out.\n */\nexport class VoiceProviderError extends Error {\n  readonly code?: string | number;\n  readonly status?: number;\n  readonly closeCode?: number;\n  readonly closeReason?: string;\n  readonly wasClean?: boolean;\n\n  constructor(message: string, options: VoiceProviderErrorOptions = {}) {\n    super(message);\n    this.name = \"VoiceProviderError\";\n    if (options.code !== undefined) this.code = options.code;\n    if (options.status !== undefined) this.status = options.status;\n    if (options.closeCode !== undefined) this.closeCode = options.closeCode;\n    if (options.closeReason !== undefined) {\n      this.closeReason = options.closeReason;\n    }\n    if (options.wasClean !== undefined) this.wasClean = options.wasClean;\n  }\n}\n\nexport interface VoiceErrorLogOptions {\n  component: string;\n  stage: string;\n  message: string;\n  error: Error;\n  connectionId?: string;\n}\n\n/** Convert a value caught from an external boundary without inspecting it. */\nexport function toVoiceError(error: unknown, fallback: string): Error {\n  return error instanceof Error ? error : new Error(fallback);\n}\n\n/** Return a bounded direct message suitable for the existing string wire API. */\nexport function voiceErrorMessage(error: Error, fallback: string): string {\n  const message = error.message.trim();\n  if (!message || message === \"[object Object]\") return fallback;\n  return message.slice(0, MAX_CLIENT_MESSAGE_LENGTH);\n}\n\n/** Emit one structured object so Workers Logs can extract diagnostic fields. */\nexport function logVoiceError(options: VoiceErrorLogOptions): void {\n  console.error({\n    component: options.component,\n    stage: options.stage,\n    message: options.message,\n    ...(options.connectionId ? { connectionId: options.connectionId } : {}),\n    error: options.error\n  });\n}\n"],"mappings":";AAAA,MAAM,4BAA4B;;;;;AAclC,IAAa,qBAAb,cAAwC,MAAM;CAO5C,YAAY,SAAiB,UAAqC,CAAC,GAAG;EACpE,MAAM,OAAO;EACb,KAAK,OAAO;EACZ,IAAI,QAAQ,SAAS,KAAA,GAAW,KAAK,OAAO,QAAQ;EACpD,IAAI,QAAQ,WAAW,KAAA,GAAW,KAAK,SAAS,QAAQ;EACxD,IAAI,QAAQ,cAAc,KAAA,GAAW,KAAK,YAAY,QAAQ;EAC9D,IAAI,QAAQ,gBAAgB,KAAA,GAC1B,KAAK,cAAc,QAAQ;EAE7B,IAAI,QAAQ,aAAa,KAAA,GAAW,KAAK,WAAW,QAAQ;CAC9D;AACF;;AAWA,SAAgB,aAAa,OAAgB,UAAyB;CACpE,OAAO,iBAAiB,QAAQ,QAAQ,IAAI,MAAM,QAAQ;AAC5D;;AAGA,SAAgB,kBAAkB,OAAc,UAA0B;CACxE,MAAM,UAAU,MAAM,QAAQ,KAAK;CACnC,IAAI,CAAC,WAAW,YAAY,mBAAmB,OAAO;CACtD,OAAO,QAAQ,MAAM,GAAG,yBAAyB;AACnD;;AAGA,SAAgB,cAAc,SAAqC;CACjE,QAAQ,MAAM;EACZ,WAAW,QAAQ;EACnB,OAAO,QAAQ;EACf,SAAS,QAAQ;EACjB,GAAI,QAAQ,eAAe,EAAE,cAAc,QAAQ,aAAa,IAAI,CAAC;EACrE,OAAO,QAAQ;CACjB,CAAC;AACH"}

package/dist/voice/index.d.ts

@@ -0,0 +1,271 @@
+import { b as Connection } from "../capability-runner-Be_-PLR1.js";
+import { c as Agent } from "../agent-routing-DE5zmCQ8.js";
+import {
+  C as VoiceTransport,
+  D as VoiceTurnSource,
+  E as VoiceTurnOutcome,
+  S as VoiceStatus,
+  T as VoiceTurnMetrics,
+  _ as VoiceErrorStage,
+  a as TranscriberSessionOptions,
+  b as VoiceRole,
+  c as VoiceAudioFormat,
+  d as VoiceCompletionOutcome,
+  f as VoiceCompletionOutcomeCode,
+  g as VoiceErrorCode,
+  h as VoiceError,
+  i as TranscriberSession,
+  l as VoiceAudioInput,
+  m as VoiceDiagnosticsOptions,
+  n as TTSProvider,
+  o as TranscriptMessage,
+  p as VoiceDiagnosticEvent,
+  r as Transcriber,
+  s as VOICE_PROTOCOL_VERSION,
+  t as StreamingTTSProvider,
+  u as VoiceClientMessage,
+  v as VoiceModelFinishReason,
+  w as VoiceTransportCloseInfo,
+  x as VoiceServerMessage,
+  y as VoicePipelineMetrics
+} from "../types-_Faxb570.js";
+import {
+  n as TextSource,
+  r as iterateText,
+  t as SentenceChunker
+} from "../sentence-chunker-BAidJ4DA.js";
+import {
+  SFUConfig,
+  addSFUTracks,
+  createSFUSession,
+  createSFUWebSocketAdapter,
+  decodeVarint,
+  downsample48kStereoTo16kMono,
+  encodePayloadToProtobuf,
+  encodeVarint,
+  extractPayloadFromProtobuf,
+  renegotiateSFUSession,
+  sfuFetch,
+  upsample16kMonoTo48kStereo
+} from "./sfu.js";
+import {
+  WorkersAIFluxSTT,
+  WorkersAIFluxSTTOptions,
+  WorkersAINova3STT,
+  WorkersAINova3STTOptions,
+  WorkersAITTS,
+  WorkersAITTSOptions
+} from "./workers-ai.js";
+
+//#region src/voice/voice-input.d.ts
+type Constructor$1<T = object> = new (...args: any[]) => T;
+type AgentLike$1 = Constructor$1<Pick<Agent<Cloudflare.Env>, "keepAlive">>;
+/** Configuration options for the voice input mixin. */
+interface VoiceInputOptions {
+  /** Optional diagnostic output. Diagnostic event names and metadata are not stable API. */
+  diagnostics?: VoiceDiagnosticsOptions;
+}
+/** Public surface of the voice input mixin, used as an explicit return type to satisfy TS6 declaration emit. */
+interface VoiceInputMixinMembers {
+  transcriber?: Transcriber;
+  onTranscript(text: string, connection: Connection): void | Promise<void>;
+  createTranscriber(connection: Connection): Transcriber | null;
+  beforeCallStart(connection: Connection): boolean | Promise<boolean>;
+  onCallStart(connection: Connection): void | Promise<void>;
+  onCallEnd(connection: Connection): void | Promise<void>;
+  onInterrupt(connection: Connection): void | Promise<void>;
+  afterTranscribe(
+    transcript: string,
+    connection: Connection
+  ): string | null | Promise<string | null>;
+}
+type VoiceInputMixinReturn<TBase extends AgentLike$1> = TBase &
+  (new (...args: any[]) => VoiceInputMixinMembers);
+/**
+ * Voice-to-text input mixin. Adds STT-only voice input to an Agent class.
+ *
+ * Subclasses must set a `transcriber` property (or override `createTranscriber`).
+ * No TTS provider is needed. Override `onTranscript` to handle each
+ * transcribed utterance.
+ *
+ * @param Base - The Agent class to extend (e.g. `Agent`).
+ * @param voiceInputOptions - Optional pipeline configuration.
+ *
+ * @example
+ * ```typescript
+ * import { Agent } from "../index";
+ * import { withVoiceInput, WorkersAINova3STT } from "agents/voice";
+ *
+ * const InputAgent = withVoiceInput(Agent);
+ *
+ * class MyAgent extends InputAgent<Env> {
+ *   transcriber = new WorkersAINova3STT(this.env.AI);
+ *
+ *   onTranscript(text, connection) {
+ *     console.log("User said:", text);
+ *   }
+ * }
+ * ```
+ */
+declare function withVoiceInput<TBase extends AgentLike$1>(
+  Base: TBase,
+  voiceInputOptions?: VoiceInputOptions
+): VoiceInputMixinReturn<TBase>;
+//#endregion
+//#region src/voice/index.d.ts
+/** Context passed to the `onTurn()` hook. */
+interface VoiceTurnContext {
+  connection: Connection;
+  /** Completed conversation history before the current transcript. */
+  messages: Array<{
+    role: VoiceRole;
+    content: string;
+  }>;
+  signal: AbortSignal;
+}
+/** Configuration options for the voice mixin. Passed to `withVoice()`. */
+interface VoiceAgentOptions {
+  /** Max conversation history messages loaded for context. @default 20 */
+  historyLimit?: number;
+  /** Audio format used for binary audio payloads sent to the client. @default "mp3" */
+  audioFormat?: VoiceAudioFormat;
+  /**
+   * Sample rate (Hz) of raw PCM audio payloads sent to the client.
+   * Declared in the `audio_config` message so the client can play `pcm16`
+   * at the provider's native rate (e.g. 24000 for Gemini TTS).
+   * Encoded formats (mp3/wav/opus) carry their own rate and ignore this.
+   * @default 16000
+   */
+  sampleRate?: number;
+  /** Max conversation messages to keep in SQLite. Oldest are pruned. @default 1000 */
+  maxMessageCount?: number;
+  /** Optional diagnostic output. Diagnostic event names and metadata are not stable API. */
+  diagnostics?: VoiceDiagnosticsOptions;
+}
+type Constructor<T = object> = new (...args: any[]) => T;
+type AgentLike = Constructor<
+  Pick<Agent<Cloudflare.Env>, "sql" | "getConnections" | "keepAlive">
+>;
+/** Public surface of the voice mixin, used as an explicit return type to satisfy TS6 declaration emit. */
+interface VoiceAgentMixinMembers {
+  transcriber?: Transcriber;
+  tts?: (TTSProvider & Partial<StreamingTTSProvider>) | undefined;
+  onTurn(transcript: string, context: VoiceTurnContext): Promise<TextSource>;
+  createTranscriber(connection: Connection): Transcriber | null;
+  beforeCallStart(connection: Connection): boolean | Promise<boolean>;
+  onCallStart(connection: Connection): void | Promise<void>;
+  onCallEnd(connection: Connection): void | Promise<void>;
+  onInterrupt(connection: Connection): void | Promise<void>;
+  afterTranscribe(
+    transcript: string,
+    connection: Connection
+  ): string | null | Promise<string | null>;
+  beforeSynthesize(
+    text: string,
+    connection: Connection
+  ): string | null | Promise<string | null>;
+  afterSynthesize(
+    audio: ArrayBuffer | null,
+    text: string,
+    connection: Connection
+  ): ArrayBuffer | null | Promise<ArrayBuffer | null>;
+  saveMessage(role: "user" | "assistant", text: string): void;
+  getConversationHistory(limit?: number): Array<{
+    role: VoiceRole;
+    content: string;
+  }>;
+  forceEndCall(connection: Connection): void;
+  speak(connection: Connection, text: string): Promise<void>;
+  speakAll(text: string): Promise<void>;
+}
+type VoiceAgentMixinReturn<TBase extends AgentLike> = TBase &
+  (new (...args: any[]) => VoiceAgentMixinMembers);
+/**
+ * Voice pipeline mixin. Adds the full voice pipeline to an Agent class.
+ *
+ * Subclasses must set a `transcriber` property (or override `createTranscriber`)
+ * and a `tts` provider property. The transcriber session is per-call — created
+ * at start_call and closed at end_call. The model handles turn detection.
+ *
+ * @param Base - The Agent class to extend (e.g. `Agent`).
+ * @param voiceOptions - Optional pipeline configuration.
+ *
+ * @example
+ * ```typescript
+ * import { Agent } from "../index";
+ * import { withVoice, WorkersAIFluxSTT, WorkersAITTS } from "agents/voice";
+ *
+ * const VoiceAgent = withVoice(Agent);
+ *
+ * class MyAgent extends VoiceAgent<Env> {
+ *   transcriber = new WorkersAIFluxSTT(this.env.AI);
+ *   tts = new WorkersAITTS(this.env.AI);
+ *
+ *   async onTurn(transcript, context) {
+ *     return "Hello! I heard you say: " + transcript;
+ *   }
+ * }
+ * ```
+ */
+declare function withVoice<TBase extends AgentLike>(
+  Base: TBase,
+  voiceOptions?: VoiceAgentOptions
+): VoiceAgentMixinReturn<TBase>;
+//#endregion
+export {
+  type SFUConfig,
+  SentenceChunker,
+  type StreamingTTSProvider,
+  type TTSProvider,
+  type TextSource,
+  type Transcriber,
+  type TranscriberSession,
+  type TranscriberSessionOptions,
+  type TranscriptMessage,
+  VOICE_PROTOCOL_VERSION,
+  VoiceAgentMixinMembers,
+  VoiceAgentOptions,
+  type VoiceAudioFormat,
+  type VoiceAudioInput,
+  type VoiceClientMessage,
+  type VoiceCompletionOutcome,
+  type VoiceCompletionOutcomeCode,
+  type VoiceDiagnosticEvent,
+  type VoiceDiagnosticsOptions,
+  type VoiceError,
+  type VoiceErrorCode,
+  type VoiceErrorStage,
+  type VoiceInputOptions,
+  type VoiceModelFinishReason,
+  type VoicePipelineMetrics,
+  type VoiceRole,
+  type VoiceServerMessage,
+  type VoiceStatus,
+  type VoiceTransport,
+  type VoiceTransportCloseInfo,
+  VoiceTurnContext,
+  type VoiceTurnMetrics,
+  type VoiceTurnOutcome,
+  type VoiceTurnSource,
+  WorkersAIFluxSTT,
+  type WorkersAIFluxSTTOptions,
+  WorkersAINova3STT,
+  type WorkersAINova3STTOptions,
+  WorkersAITTS,
+  type WorkersAITTSOptions,
+  addSFUTracks,
+  createSFUSession,
+  createSFUWebSocketAdapter,
+  decodeVarint,
+  downsample48kStereoTo16kMono,
+  encodePayloadToProtobuf,
+  encodeVarint,
+  extractPayloadFromProtobuf,
+  iterateText,
+  renegotiateSFUSession,
+  sfuFetch,
+  upsample16kMonoTo48kStereo,
+  withVoice,
+  withVoiceInput
+};
+//# sourceMappingURL=index.d.ts.map