@tanstack/ai-grok 0.6.7 → 0.7.0

package/dist/esm/realtime/adapter.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"adapter.js","sources":["../../../src/realtime/adapter.ts"],"sourcesContent":["import { resolveDebugOption } from '@tanstack/ai/adapter-internals'\nimport type {\n  AnyClientTool,\n  AudioVisualization,\n  RealtimeEvent,\n  RealtimeEventHandler,\n  RealtimeMessage,\n  RealtimeMode,\n  RealtimeSessionConfig,\n  RealtimeStatus,\n  RealtimeToken,\n} from '@tanstack/ai'\nimport type { InternalLogger } from '@tanstack/ai/adapter-internals'\nimport type { RealtimeAdapter, RealtimeConnection } from './realtime-contract'\nimport type { GrokRealtimeOptions } from './types'\n\nconst GROK_REALTIME_URL = 'https://api.x.ai/v1/realtime'\n\n/**\n * Runtime-checked field readers for untyped server events. Replace the\n * drive-by `event.X as string` / `event.X as Record<string, unknown>` casts\n * with readers that return `undefined` when the shape doesn't match, so a\n * malformed frame can't throw a TypeError inside `handleServerEvent`.\n */\nfunction readString(\n  obj: Record<string, unknown>,\n  key: string,\n): string | undefined {\n  const value = obj[key]\n  return typeof value === 'string' ? value : undefined\n}\n\nfunction readObject(\n  obj: Record<string, unknown>,\n  key: string,\n): Record<string, unknown> | undefined {\n  const value = obj[key]\n  return value && typeof value === 'object' && !Array.isArray(value)\n    ? (value as Record<string, unknown>)\n    : undefined\n}\n\nfunction readObjectArray(\n  obj: Record<string, unknown>,\n  key: string,\n): Array<Record<string, unknown>> | undefined {\n  const value = obj[key]\n  if (!Array.isArray(value)) return undefined\n  return value.filter(\n    (item): item is Record<string, unknown> =>\n      item !== null && typeof item === 'object' && !Array.isArray(item),\n  )\n}\n\ntype RealtimeServerError = Error & {\n  code?: string\n  type?: string\n  param?: string\n}\n\n/**\n * Creates a Grok realtime adapter for client-side use.\n *\n * Uses WebRTC for browser connections (default). Mirrors the OpenAI realtime\n * adapter because xAI's Voice Agent API is OpenAI-realtime-compatible — the\n * only differences are the endpoint URL and default model.\n *\n * @example\n * ```typescript\n * import { RealtimeClient } from '@tanstack/ai-client'\n * import { grokRealtime } from '@tanstack/ai-grok'\n *\n * const client = new RealtimeClient({\n *   getToken: () => fetch('/api/realtime-token').then(r => r.json()),\n *   adapter: grokRealtime(),\n * })\n * ```\n */\nexport function grokRealtime(\n  options: GrokRealtimeOptions = {},\n): RealtimeAdapter {\n  const connectionMode = options.connectionMode ?? 'webrtc'\n  const logger = resolveDebugOption(options.debug)\n\n  return {\n    provider: 'grok',\n\n    async connect(\n      token: RealtimeToken,\n      _clientTools?: ReadonlyArray<AnyClientTool>,\n    ): Promise<RealtimeConnection> {\n      const model = token.config.model ?? 'grok-voice-fast-1.0'\n      logger.request(`activity=realtime provider=grok model=${model}`, {\n        provider: 'grok',\n        model,\n      })\n\n      if (connectionMode === 'webrtc') {\n        return createWebRTCConnection(token, logger)\n      }\n      const error = new Error('WebSocket connection mode not yet implemented')\n      logger.errors('grok.realtime fatal', {\n        error,\n        source: 'grok.realtime',\n      })\n      throw error\n    },\n  }\n}\n\n/**\n * Creates a WebRTC connection to xAI's realtime API.\n */\nasync function createWebRTCConnection(\n  token: RealtimeToken,\n  logger: InternalLogger,\n): Promise<RealtimeConnection> {\n  const model = token.config.model ?? 'grok-voice-fast-1.0'\n  const eventHandlers = new Map<RealtimeEvent, Set<RealtimeEventHandler<any>>>()\n\n  const pc = new RTCPeerConnection()\n\n  let audioContext: AudioContext | null = null\n  let inputAnalyser: AnalyserNode | null = null\n  let outputAnalyser: AnalyserNode | null = null\n  let inputSource: MediaStreamAudioSourceNode | null = null\n  let outputSource: MediaStreamAudioSourceNode | null = null\n  let localStream: MediaStream | null = null\n\n  let audioElement: HTMLAudioElement | null = null\n\n  let dataChannel: RTCDataChannel | null = null\n\n  let currentMode: RealtimeMode = 'idle'\n  let currentMessageId: string | null = null\n\n  // Flipped by `teardownConnection`. Guards `sendEvent` so post-disconnect\n  // calls (e.g. a React `useEffect` cleanup flushing queued events) are\n  // logged and skipped instead of silently piling up in `pendingEvents`.\n  let isTornDown = false\n\n  // Outbound events queued while the data channel isn't yet open. Declared\n  // here (rather than next to `sendEvent`) so `teardownConnection` — which\n  // lives higher up and can run from the SDP-path catch before `sendEvent`\n  // is defined — can drain it without hitting the TDZ.\n  const pendingEvents: Array<Record<string, unknown>> = []\n\n  // Tracks whether we've sent the first session.update. On the first update\n  // we attach a default input_audio_transcription so the server will emit\n  // user transcripts unless the caller opts out via\n  // `providerOptions.inputAudioTranscription = null | false`.\n  let hasSentInitialSessionUpdate = false\n\n  // Size hints for the fallback buffers returned when an analyser isn't yet\n  // populated. We return a *fresh* `Uint8Array` on each call so a caller\n  // that draws into it (e.g. a canvas visualiser zeroing the buffer) can't\n  // mutate a shared module-level instance for every other consumer.\n  const FALLBACK_FREQUENCY_BIN_COUNT = 1024\n  const FALLBACK_TIME_DOMAIN_SIZE = 2048\n  const FALLBACK_TIME_DOMAIN_FILL = 128\n\n  function emit<TEvent extends RealtimeEvent>(\n    event: TEvent,\n    payload: Parameters<RealtimeEventHandler<TEvent>>[0],\n  ) {\n    const handlers = eventHandlers.get(event)\n    if (handlers) {\n      for (const handler of handlers) {\n        handler(payload)\n      }\n    }\n  }\n\n  dataChannel = pc.createDataChannel('oai-events')\n\n  let dataChannelOpened = false\n  let rejectDataChannelReady: ((reason: unknown) => void) | null = null\n  let dataChannelReadyTimeout: ReturnType<typeof setTimeout> | null = null\n\n  const dataChannelReady = new Promise<void>((resolve, reject) => {\n    rejectDataChannelReady = (reason) => {\n      if (dataChannelReadyTimeout !== null) {\n        clearTimeout(dataChannelReadyTimeout)\n        dataChannelReadyTimeout = null\n      }\n      // One-shot: null out so later state transitions don't reject twice.\n      rejectDataChannelReady = null\n      reject(reason)\n    }\n\n    dataChannelReadyTimeout = setTimeout(() => {\n      if (!dataChannelOpened) {\n        rejectDataChannelReady?.(\n          new Error(\n            'Data channel did not open within 15000ms — aborting connection',\n          ),\n        )\n      }\n    }, 15000)\n\n    dataChannel!.onopen = () => {\n      dataChannelOpened = true\n      if (dataChannelReadyTimeout !== null) {\n        clearTimeout(dataChannelReadyTimeout)\n        dataChannelReadyTimeout = null\n      }\n      // Once resolved, rejecting is a no-op — null out so teardown paths\n      // don't attempt a redundant reject on an already-settled promise.\n      rejectDataChannelReady = null\n      flushPendingEvents()\n      emit('status_change', { status: 'connected' as RealtimeStatus })\n      resolve()\n    }\n  })\n\n  dataChannel.onmessage = (event) => {\n    try {\n      const message = JSON.parse(event.data)\n      const messageRecord: Record<string, unknown> =\n        message !== null && typeof message === 'object' ? message : {}\n      logger.provider(\n        `provider=grok direction=in type=${readString(messageRecord, 'type') ?? '<unknown>'}`,\n        { frame: messageRecord },\n      )\n      handleServerEvent(messageRecord)\n    } catch (parseErr) {\n      logger.errors('grok.realtime fatal', {\n        error: parseErr,\n        source: 'grok.realtime',\n      })\n      emit('error', {\n        error:\n          parseErr instanceof Error ? parseErr : new Error(String(parseErr)),\n      })\n    }\n  }\n\n  dataChannel.onerror = (error) => {\n    // Closing the peer connection cascades into `onerror`/`onclose` on the\n    // data channel. Once teardown has started, re-surfacing those as\n    // `emit('error')` is noise that confuses consumers (they just called\n    // `disconnect()` — they don't want an error toast for it).\n    if (isTornDown) return\n    logger.errors('grok.realtime fatal', {\n      error,\n      source: 'grok.realtime',\n    })\n    // RTCErrorEvent exposes a typed `.error`; fall back to the event type\n    // name, then to a string representation, so the emitted error message\n    // doesn't end up as \"[object Event]\".\n    // `onerror` always fires with an Event (often an RTCErrorEvent), so we\n    // can read it via the untyped helpers without first proving object-ness.\n    const errorRecord = error as unknown as Record<string, unknown>\n    const rtcError = readObject(errorRecord, 'error')\n    const msg =\n      (rtcError && readString(rtcError, 'message')) ?? (error.type || 'unknown')\n    const dcErr = new Error(`Data channel error: ${msg}`)\n    if (!dataChannelOpened) {\n      rejectDataChannelReady?.(dcErr)\n    }\n    emit('error', { error: dcErr })\n  }\n\n  dataChannel.onclose = () => {\n    // Same rationale as `onerror` above: `pc.close()` during teardown\n    // cascades to the data channel's `onclose`. If we've already started\n    // teardown, there's nothing to do here.\n    if (isTornDown) return\n    if (!dataChannelOpened) {\n      rejectDataChannelReady?.(new Error('Data channel closed before opening'))\n    }\n  }\n\n  pc.ontrack = (event) => {\n    if (event.track.kind === 'audio' && event.streams[0]) {\n      setupOutputAudioAnalysis(event.streams[0])\n    }\n  }\n\n  // `status_change` has a single source of truth: `onconnectionstatechange`\n  // (the higher-level aggregate state). `oniceconnectionstatechange` is\n  // responsible only for rejecting `dataChannelReady` on ICE failures so we\n  // surface them without waiting for the 15s timeout.\n  pc.onconnectionstatechange = () => {\n    const state = pc.connectionState\n    logger.provider(`provider=grok pc.connectionState=${state}`, {\n      state,\n    })\n    if (state === 'failed' || state === 'disconnected' || state === 'closed') {\n      // Suppress the `status_change` emission when teardown is in progress:\n      // the user-facing `disconnect()` already emits `status_change: 'idle'`\n      // and then calls `teardownConnection()` → `pc.close()`, which fires\n      // `onconnectionstatechange` with state === 'closed'. Without this\n      // guard listeners would see two `idle` events per disconnect.\n      if (!isTornDown) {\n        emit('status_change', {\n          status:\n            state === 'failed'\n              ? ('error' as RealtimeStatus)\n              : ('idle' as RealtimeStatus),\n        })\n      }\n      if (!dataChannelOpened) {\n        // Reject on any terminal-ish pre-open state so callers don't hang\n        // for the full 15s timeout. The reject is one-shot — subsequent\n        // state changes become no-ops via the null-out in\n        // `rejectDataChannelReady`.\n        const message =\n          state === 'failed'\n            ? `PeerConnection failed before data channel opened`\n            : `PeerConnection entered state '${state}' before data channel opened`\n        rejectDataChannelReady?.(new Error(message))\n      }\n      // Auto-teardown on `failed`: without this the mic track, pc, and\n      // AudioContext stay allocated after a fatal connection failure, so the\n      // browser's mic indicator stays on and the user sees a broken\n      // \"connected mic\" state. `closed` already means pc was torn down\n      // (usually by teardownConnection itself) so nothing extra to do.\n      // `disconnected` is transient per the WebRTC spec and may recover, so\n      // we leave resources in place. `teardownConnection` is idempotent so\n      // a subsequent consumer `disconnect()` remains safe.\n      if (state === 'failed' && !isTornDown) {\n        void teardownConnection()\n      }\n    }\n  }\n\n  pc.oniceconnectionstatechange = () => {\n    const state = pc.iceConnectionState\n    logger.provider(`provider=grok pc.iceConnectionState=${state}`, {\n      state,\n    })\n    if (\n      !dataChannelOpened &&\n      (state === 'failed' || state === 'closed' || state === 'disconnected')\n    ) {\n      const message =\n        state === 'failed'\n          ? `ICE connection failed before data channel opened`\n          : `ICE connection entered state '${state}' before data channel opened`\n      rejectDataChannelReady?.(new Error(message))\n    }\n  }\n\n  /**\n   * Tear down every resource we may have allocated so the mic/pc/audio\n   * nodes/audio element don't leak on a failed connect. Safe to call from\n   * any point after `new RTCPeerConnection()`; each branch null-guards and\n   * swallows errors because cascading closes (e.g. `pc.close()` closing the\n   * data channel implicitly) are expected.\n   *\n   * Shared between the SDP-path catch, the post-SDP catch, and (implicitly\n   * via idempotency) the `disconnect()` entry point.\n   */\n  async function teardownConnection() {\n    // Flip the teardown flag BEFORE any awaits so handlers that fire during\n    // `await audioContext.close()` (or any other async step below) can guard\n    // on it — otherwise a late `pc.onconnectionstatechange` or `pc.ontrack`\n    // can allocate new resources or re-emit `status_change: idle` after the\n    // user-facing `disconnect()` already emitted one.\n    isTornDown = true\n\n    // Drop any queued events the caller sent before the data channel opened\n    // up front. Without this they'd accumulate across reconnect attempts\n    // (each connect allocates a fresh closure, but a caller holding the old\n    // `connection` reference could otherwise keep appending forever). Done\n    // at the top — before the awaits below — so `sendEvent` calls racing\n    // with teardown don't push into a list we're about to drain.\n    pendingEvents.length = 0\n\n    // Clear the data-channel-open timeout / reject the readiness promise\n    // if it's still pending. `rejectDataChannelReady` is one-shot and nulls\n    // itself on first call, so calling it from `disconnect()` after a\n    // successful open is a no-op.\n    rejectDataChannelReady?.(\n      new Error('Connection torn down before data channel opened'),\n    )\n\n    if (localStream) {\n      for (const track of localStream.getTracks()) {\n        track.stop()\n      }\n      localStream = null\n    }\n\n    // Output audio (populated by `pc.ontrack` → setupOutputAudioAnalysis,\n    // which may have fired during SDP negotiation before we threw).\n    if (audioElement) {\n      try {\n        audioElement.pause()\n      } catch {\n        // ignore — element may already be unloaded\n      }\n      audioElement.srcObject = null\n      audioElement = null\n    }\n    if (outputSource) {\n      try {\n        outputSource.disconnect()\n      } catch {\n        // ignore\n      }\n      outputSource = null\n    }\n    if (outputAnalyser) {\n      try {\n        outputAnalyser.disconnect()\n      } catch {\n        // ignore\n      }\n      outputAnalyser = null\n    }\n\n    // Input audio (populated by setupInputAudioAnalysis after SDP).\n    if (inputSource) {\n      try {\n        inputSource.disconnect()\n      } catch {\n        // ignore\n      }\n      inputSource = null\n    }\n    if (inputAnalyser) {\n      try {\n        inputAnalyser.disconnect()\n      } catch {\n        // ignore\n      }\n      inputAnalyser = null\n    }\n\n    if (dataChannel) {\n      try {\n        dataChannel.close()\n      } catch {\n        // ignore — channel may already be closed by pc.close()\n      }\n      dataChannel = null\n    }\n\n    try {\n      pc.close()\n    } catch {\n      // ignore — pc may already be closed\n    }\n\n    if (audioContext) {\n      try {\n        await audioContext.close()\n      } catch {\n        // ignore — context may already be closed\n      }\n      audioContext = null\n    }\n  }\n\n  // xAI requires an audio track in the SDP offer, same as OpenAI realtime.\n  //\n  // This try/catch also covers `getUserMedia` failure (e.g. the user denies\n  // microphone permission). `pc` + `dataChannel` are already allocated above\n  // and the 15s `dataChannelReady` timeout is already armed, so we MUST\n  // teardown on failure here — otherwise they leak until the tab closes.\n  // `teardownConnection` is idempotent and null-safe (runs fine even if the\n  // mic was never acquired).\n  try {\n    try {\n      localStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          echoCancellation: true,\n          noiseSuppression: true,\n          sampleRate: 24000,\n        },\n      })\n    } catch (error) {\n      logger.errors('grok.realtime fatal', {\n        error,\n        source: 'grok.realtime.getUserMedia',\n      })\n      // Re-throw with the descriptive message callers rely on. Teardown runs\n      // in the outer catch below.\n      throw new Error(\n        `Microphone access required for realtime voice: ${error instanceof Error ? error.message : error}`,\n      )\n    }\n\n    for (const track of localStream.getAudioTracks()) {\n      pc.addTrack(track, localStream)\n    }\n\n    const offer = await pc.createOffer()\n    await pc.setLocalDescription(offer)\n\n    const sdpResponse = await fetch(`${GROK_REALTIME_URL}?model=${model}`, {\n      method: 'POST',\n      headers: {\n        Authorization: `Bearer ${token.token}`,\n        'Content-Type': 'application/sdp',\n      },\n      body: offer.sdp,\n    })\n\n    if (!sdpResponse.ok) {\n      const errorText = await sdpResponse.text()\n      const error = new Error(\n        `Failed to establish WebRTC connection: ${sdpResponse.status} - ${errorText}`,\n      )\n      logger.errors('grok.realtime fatal', {\n        error,\n        source: 'grok.realtime.sdp',\n        status: sdpResponse.status,\n      })\n      throw error\n    }\n\n    const answerSdp = await sdpResponse.text()\n    await pc.setRemoteDescription({ type: 'answer', sdp: answerSdp })\n  } catch (err) {\n    await teardownConnection()\n    throw err\n  }\n\n  // Second cleanup scope: after SDP succeeds we still have to set up input\n  // audio analysis and wait for the data channel to open. Both can fail\n  // (AudioContext allocation, 15s timeout, ICE failure, pc.close from the\n  // other end, etc.) and those failures must NOT leave the mic/pc/audio\n  // nodes running.\n  try {\n    setupInputAudioAnalysis(localStream)\n    await dataChannelReady\n  } catch (err) {\n    await teardownConnection()\n    throw err\n  }\n\n  function handleServerEvent(event: Record<string, unknown>) {\n    const type = readString(event, 'type')\n\n    switch (type) {\n      case 'session.created':\n      case 'session.updated':\n        break\n\n      case 'input_audio_buffer.speech_started':\n        currentMode = 'listening'\n        emit('mode_change', { mode: 'listening' })\n        break\n\n      case 'input_audio_buffer.speech_stopped':\n        currentMode = 'thinking'\n        emit('mode_change', { mode: 'thinking' })\n        break\n\n      case 'input_audio_buffer.committed':\n        break\n\n      case 'conversation.item.input_audio_transcription.completed': {\n        const transcript = readString(event, 'transcript')\n        if (transcript === undefined) break\n        emit('transcript', { role: 'user', transcript, isFinal: true })\n        break\n      }\n\n      case 'response.created':\n        // Reset message id so a tool-only response (which never emits\n        // response.output_item.added for a message) can't reuse the previous\n        // turn's id when `response.done` later inspects this flag.\n        currentMessageId = null\n        currentMode = 'thinking'\n        emit('mode_change', { mode: 'thinking' })\n        break\n\n      case 'response.output_item.added': {\n        const item = readObject(event, 'item')\n        if (item && readString(item, 'type') === 'message') {\n          const id = readString(item, 'id')\n          if (id !== undefined) currentMessageId = id\n        }\n        break\n      }\n\n      // xAI realtime per docs uses `response.output_audio_transcript.*`;\n      // accept the legacy OpenAI-realtime `response.audio_transcript.*` as\n      // an alias so this adapter stays compatible across protocol versions.\n      case 'response.output_audio_transcript.delta':\n      case 'response.audio_transcript.delta': {\n        const delta = readString(event, 'delta')\n        if (delta === undefined) break\n        emit('transcript', {\n          role: 'assistant',\n          transcript: delta,\n          isFinal: false,\n        })\n        break\n      }\n\n      case 'response.output_audio_transcript.done':\n      case 'response.audio_transcript.done': {\n        const transcript = readString(event, 'transcript')\n        if (transcript === undefined) break\n        emit('transcript', { role: 'assistant', transcript, isFinal: true })\n        break\n      }\n\n      // xAI realtime per docs uses `response.text.*`; accept the legacy\n      // OpenAI-realtime `response.output_text.*` as an alias.\n      case 'response.text.delta':\n      case 'response.output_text.delta': {\n        const delta = readString(event, 'delta')\n        if (delta === undefined) break\n        emit('transcript', {\n          role: 'assistant',\n          transcript: delta,\n          isFinal: false,\n        })\n        break\n      }\n\n      case 'response.text.done':\n      case 'response.output_text.done': {\n        const text = readString(event, 'text')\n        if (text === undefined) break\n        emit('transcript', {\n          role: 'assistant',\n          transcript: text,\n          isFinal: true,\n        })\n        break\n      }\n\n      // xAI realtime per docs uses `response.output_audio.*`; accept the\n      // legacy OpenAI-realtime `response.audio.*` as an alias.\n      case 'response.output_audio.delta':\n      case 'response.audio.delta':\n        if (currentMode !== 'speaking') {\n          currentMode = 'speaking'\n          emit('mode_change', { mode: 'speaking' })\n        }\n        break\n\n      case 'response.output_audio.done':\n      case 'response.audio.done':\n        break\n\n      case 'response.function_call_arguments.done': {\n        // Only `call_id` is valid for `sendToolResult` correlation. Falling\n        // back to `item_id` would produce a tool-call id the server doesn't\n        // recognise when the result is posted back, silently dropping the\n        // tool execution. If `call_id` is missing we surface an error event\n        // so the UI can react instead of pretending the tool call succeeded.\n        const callId = readString(event, 'call_id')\n        const name = readString(event, 'name') ?? ''\n        const args = readString(event, 'arguments') ?? ''\n        if (!callId) {\n          logger.errors(\n            'grok.realtime tool_call missing call_id — dropping tool_call',\n            {\n              source: 'grok.realtime',\n              event_type: 'response.function_call_arguments.done',\n              item_id: event.item_id,\n            },\n          )\n          emit('error', {\n            error: new Error(\n              'Realtime tool call missing call_id; tool will not execute',\n            ),\n          })\n          break\n        }\n        try {\n          const input = JSON.parse(args)\n          emit('tool_call', { toolCallId: callId, toolName: name, input })\n        } catch {\n          emit('tool_call', { toolCallId: callId, toolName: name, input: args })\n        }\n        break\n      }\n\n      case 'response.done': {\n        const response = readObject(event, 'response') ?? {}\n        const output = readObjectArray(response, 'output')\n\n        // Only transition back to `listening` if the user hasn't already\n        // stopped capture — otherwise we'd override their explicit `idle`\n        // state and re-arm the mic visualisation.\n        if (currentMode !== 'idle') {\n          currentMode = 'listening'\n          emit('mode_change', { mode: 'listening' })\n        }\n\n        if (currentMessageId) {\n          const message: RealtimeMessage = {\n            id: currentMessageId,\n            role: 'assistant',\n            timestamp: Date.now(),\n            parts: [],\n          }\n\n          for (const item of output ?? []) {\n            if (readString(item, 'type') !== 'message') continue\n            const content = readObjectArray(item, 'content')\n            if (!content) continue\n            for (const part of content) {\n              const partType = readString(part, 'type')\n              if (partType === 'audio') {\n                const transcript = readString(part, 'transcript')\n                if (transcript) {\n                  message.parts.push({ type: 'audio', transcript })\n                }\n              } else if (partType === 'text') {\n                const content = readString(part, 'text')\n                if (content) {\n                  message.parts.push({ type: 'text', content })\n                }\n              }\n            }\n          }\n\n          emit('message_complete', { message })\n          currentMessageId = null\n        }\n        break\n      }\n\n      case 'conversation.item.truncated':\n        // Assistant playback was interrupted — flip mode back to `listening`\n        // unless the user already called `stopAudioCapture()` (idle). Without\n        // this the visualisation would stay stuck on `speaking` even though\n        // no audio is playing.\n        if (currentMode !== 'idle') {\n          currentMode = 'listening'\n          emit('mode_change', { mode: 'listening' })\n        }\n        emit('interrupted', { messageId: currentMessageId ?? undefined })\n        break\n\n      case 'error': {\n        // The realtime server's `error` envelope isn't guaranteed to carry\n        // an `error` object at all (network-layer corruption, protocol\n        // drift, etc.). Validate shape before dereferencing so a malformed\n        // payload can't throw a TypeError inside this handler and stop the\n        // switch from running for the rest of the session.\n        const errorObj = readObject(event, 'error') ?? {}\n        const message =\n          readString(errorObj, 'message') ?? 'Unknown realtime server error'\n        const err: RealtimeServerError = new Error(message)\n        // Preserve `code` / `type` / `param` on the Error as extra props so\n        // consumers can branch on them without re-parsing the raw event.\n        const code = readString(errorObj, 'code')\n        if (code !== undefined) err.code = code\n        const errType = readString(errorObj, 'type')\n        if (errType !== undefined) err.type = errType\n        const param = readString(errorObj, 'param')\n        if (param !== undefined) err.param = param\n        logger.errors('grok.realtime server error', {\n          ...errorObj,\n          source: 'grok.realtime server',\n        })\n        emit('error', { error: err })\n        break\n      }\n\n      default:\n        // The xAI realtime protocol is a moving target; log unhandled event\n        // types at provider level so they're visible during debugging without\n        // emitting a user-visible error.\n        logger.provider('grok.realtime unhandled server event', {\n          type: event.type,\n        })\n        break\n    }\n  }\n\n  function setupOutputAudioAnalysis(stream: MediaStream) {\n    // Bail out if teardown has already started. `pc.ontrack` can fire\n    // asynchronously after `teardownConnection()` has flipped `isTornDown`\n    // (e.g. a remote track arriving mid-close); without this guard we'd\n    // allocate a fresh AudioContext / audio element that nothing would ever\n    // clean up.\n    if (isTornDown) return\n\n    // Tear down any prior output audio before allocating new resources.\n    // `pc.ontrack` can fire multiple times over the lifetime of a session\n    // (e.g. after renegotiation), and without this we'd leak audio elements\n    // and analyser nodes.\n    if (audioElement) {\n      try {\n        audioElement.pause()\n      } catch {\n        // ignore — element may already be unloaded\n      }\n      audioElement.srcObject = null\n      audioElement = null\n    }\n    if (outputSource) {\n      try {\n        outputSource.disconnect()\n      } catch {\n        // ignore — may already be disconnected\n      }\n      outputSource = null\n    }\n    if (outputAnalyser) {\n      try {\n        outputAnalyser.disconnect()\n      } catch {\n        // ignore\n      }\n      outputAnalyser = null\n    }\n\n    audioElement = new Audio()\n    audioElement.srcObject = stream\n    audioElement.autoplay = true\n    audioElement.play().catch((e) => {\n      // Autoplay is commonly blocked until the user interacts with the page\n      // (browser gesture requirement). Surfacing this as a fatal `error`\n      // event makes the UI render a red/error state even though the\n      // connection is healthy — the page just needs a click. Log at a\n      // dedicated source tag so it's debuggable, but don't emit `error`.\n      logger.errors('grok.realtime audio autoplay blocked', {\n        error: e,\n        source: 'grok.realtime.audio_permission_required',\n      })\n    })\n\n    if (!audioContext) {\n      audioContext = new AudioContext()\n    }\n\n    if (audioContext.state === 'suspended') {\n      audioContext.resume().catch((err) => {\n        // Same rationale as the autoplay catch: `resume()` failure usually\n        // means the user hasn't interacted yet. Logging only — no error\n        // emit — so the UI doesn't go into a fatal state for a recoverable\n        // condition.\n        logger.errors('grok.realtime audioContext.resume failed', {\n          error: err,\n          source: 'grok.realtime',\n        })\n      })\n    }\n\n    outputAnalyser = audioContext.createAnalyser()\n    outputAnalyser.fftSize = 2048\n    outputAnalyser.smoothingTimeConstant = 0.3\n\n    outputSource = audioContext.createMediaStreamSource(stream)\n    outputSource.connect(outputAnalyser)\n  }\n\n  function setupInputAudioAnalysis(stream: MediaStream) {\n    // Defensive symmetry with `setupOutputAudioAnalysis`. Today this is\n    // only called inline after SDP negotiation, but keeping the guard\n    // means any future caller path (e.g. renegotiation) won't leak a fresh\n    // AudioContext after teardown.\n    if (isTornDown) return\n\n    if (!audioContext) {\n      audioContext = new AudioContext()\n    }\n\n    if (audioContext.state === 'suspended') {\n      audioContext.resume().catch((err) => {\n        // Same rationale as in setupOutputAudioAnalysis: a suspended\n        // AudioContext usually resumes after a user gesture. Log only —\n        // surfacing this as a fatal error makes the UI look broken for a\n        // recoverable condition.\n        logger.errors('grok.realtime audioContext.resume failed', {\n          error: err,\n          source: 'grok.realtime',\n        })\n      })\n    }\n\n    inputAnalyser = audioContext.createAnalyser()\n    inputAnalyser.fftSize = 2048\n    inputAnalyser.smoothingTimeConstant = 0.3\n\n    inputSource = audioContext.createMediaStreamSource(stream)\n    inputSource.connect(inputAnalyser)\n  }\n\n  function sendEvent(event: Record<string, unknown>) {\n    if (isTornDown) {\n      // The caller is holding onto a `connection` object after `disconnect()`\n      // (or a failed connect). Silently queueing would leak memory and the\n      // events would never flush. Log + drop so the misuse is visible in\n      // debug mode without escalating to a throw — throwing from a React\n      // useEffect cleanup path can break teardown ordering in the UI.\n      logger.errors('grok.realtime sendEvent after disconnect', {\n        eventType: readString(event, 'type') ?? '<unknown>',\n        source: 'grok.realtime',\n      })\n      return\n    }\n    if (dataChannel?.readyState === 'open') {\n      logger.provider(\n        `provider=grok direction=out type=${readString(event, 'type') ?? '<unknown>'}`,\n        { frame: event },\n      )\n      // Mirror the try/catch in `flushPendingEvents` — `dataChannel.send`\n      // can synchronously throw if the channel flipped to `closing` between\n      // our readyState check and this call, or if `JSON.stringify` chokes\n      // on a caller-supplied payload. Log + emit error instead of letting\n      // the exception propagate up through public `sendText` / `sendImage`\n      // / `updateSession` call sites.\n      try {\n        dataChannel.send(JSON.stringify(event))\n      } catch (error) {\n        logger.errors('grok.realtime sendEvent failed', {\n          error,\n          eventType: readString(event, 'type') ?? '<unknown>',\n          source: 'grok.realtime',\n        })\n        emit('error', {\n          error: error instanceof Error ? error : new Error(String(error)),\n        })\n      }\n    } else {\n      pendingEvents.push(event)\n    }\n  }\n\n  function flushPendingEvents() {\n    try {\n      for (const event of pendingEvents) {\n        logger.provider(\n          `provider=grok direction=out type=${readString(event, 'type') ?? '<unknown>'}`,\n          { frame: event },\n        )\n        dataChannel!.send(JSON.stringify(event))\n      }\n      pendingEvents.length = 0\n    } catch (error) {\n      // A send failure here (e.g. dataChannel went from 'open' back to\n      // 'closing' mid-flush, or JSON.stringify on a caller-provided event\n      // threw) would otherwise be silently swallowed. By the time we're\n      // called, `onopen` has already resolved `dataChannelReady`, so the\n      // consumer-facing signal is `emit('error')` — try rejectDataChannelReady\n      // as a defensive belt-and-braces in case this ever runs pre-resolve.\n      logger.errors('grok.realtime flushPendingEvents failed', {\n        error,\n        source: 'grok.realtime',\n      })\n      const err = error instanceof Error ? error : new Error(String(error))\n      rejectDataChannelReady?.(err)\n      emit('error', { error: err })\n    }\n  }\n\n  const connection: RealtimeConnection = {\n    async disconnect() {\n      // Reuse the same teardown path as the failed-connect branches so\n      // every cleanup site stays in sync (input analyser, output analyser,\n      // output source, audio element, etc.).\n      await teardownConnection()\n      emit('status_change', { status: 'idle' as RealtimeStatus })\n    },\n\n    async startAudioCapture() {\n      if (localStream) {\n        for (const track of localStream.getAudioTracks()) {\n          track.enabled = true\n        }\n      }\n      currentMode = 'listening'\n      emit('mode_change', { mode: 'listening' })\n    },\n\n    stopAudioCapture() {\n      if (localStream) {\n        for (const track of localStream.getAudioTracks()) {\n          track.enabled = false\n        }\n      }\n      currentMode = 'idle'\n      emit('mode_change', { mode: 'idle' })\n    },\n\n    sendText(text: string) {\n      sendEvent({\n        type: 'conversation.item.create',\n        item: {\n          type: 'message',\n          role: 'user',\n          content: [{ type: 'input_text', text }],\n        },\n      })\n      sendEvent({ type: 'response.create' })\n    },\n\n    sendImage(imageData: string, mimeType: string) {\n      // Accept:\n      //  - http(s):// URLs → forward as-is\n      //  - data: URIs (e.g. from FileReader.readAsDataURL) → forward as-is\n      //    so we don't double-wrap into `data:image/png;base64,data:image/png;base64,…`\n      //  - bare base64 → wrap in `data:${mimeType};base64,…`\n      const isAlreadyUrlOrDataUri =\n        imageData.startsWith('http://') ||\n        imageData.startsWith('https://') ||\n        imageData.startsWith('data:')\n      const imageContent = {\n        type: 'input_image',\n        // The OpenAI-realtime content part (which this adapter mirrors) nests\n        // the URL under an `image_url: { url: ... }` object, not a bare\n        // string.\n        image_url: {\n          url: isAlreadyUrlOrDataUri\n            ? imageData\n            : `data:${mimeType};base64,${imageData}`,\n        },\n      }\n\n      sendEvent({\n        type: 'conversation.item.create',\n        item: {\n          type: 'message',\n          role: 'user',\n          content: [imageContent],\n        },\n      })\n      sendEvent({ type: 'response.create' })\n    },\n\n    sendToolResult(callId: string, result: string) {\n      sendEvent({\n        type: 'conversation.item.create',\n        item: {\n          type: 'function_call_output',\n          call_id: callId,\n          output: result,\n        },\n      })\n      sendEvent({ type: 'response.create' })\n    },\n\n    updateSession(config: Partial<RealtimeSessionConfig>) {\n      const sessionUpdate: Record<string, unknown> = {}\n\n      if (config.instructions) {\n        sessionUpdate.instructions = config.instructions\n      }\n\n      if (config.voice) {\n        sessionUpdate.voice = config.voice\n      }\n\n      if (config.vadMode) {\n        if (config.vadMode === 'semantic') {\n          sessionUpdate.turn_detection = {\n            type: 'semantic_vad',\n            eagerness: config.semanticEagerness ?? 'medium',\n          }\n        } else if (config.vadMode === 'server') {\n          sessionUpdate.turn_detection = {\n            type: 'server_vad',\n            threshold: config.vadConfig?.threshold ?? 0.5,\n            prefix_padding_ms: config.vadConfig?.prefixPaddingMs ?? 300,\n            silence_duration_ms: config.vadConfig?.silenceDurationMs ?? 500,\n          }\n        } else {\n          sessionUpdate.turn_detection = null\n        }\n      }\n\n      if (config.tools !== undefined) {\n        sessionUpdate.tools = config.tools.map((t) => ({\n          type: 'function',\n          name: t.name,\n          description: t.description,\n          parameters: t.inputSchema ?? { type: 'object', properties: {} },\n        }))\n        sessionUpdate.tool_choice = 'auto'\n      }\n\n      if (config.outputModalities) {\n        sessionUpdate.modalities = config.outputModalities\n      }\n\n      if (config.temperature !== undefined) {\n        sessionUpdate.temperature = config.temperature\n      }\n\n      if (config.maxOutputTokens !== undefined) {\n        sessionUpdate.max_response_output_tokens = config.maxOutputTokens\n      }\n\n      // Let callers forward an explicit `input_audio_transcription` value\n      // through `providerOptions` — including `null` / `false` to disable\n      // the feature. Only apply our `grok-stt` default on the first\n      // session.update and only if the caller hasn't set it themselves.\n      const providerOptions: Record<string, unknown> =\n        config.providerOptions ?? {}\n      const callerTranscription =\n        'inputAudioTranscription' in providerOptions\n          ? providerOptions.inputAudioTranscription\n          : 'input_audio_transcription' in providerOptions\n            ? providerOptions.input_audio_transcription\n            : undefined\n      if (callerTranscription !== undefined) {\n        sessionUpdate.input_audio_transcription =\n          callerTranscription === false ? null : callerTranscription\n      } else if (!hasSentInitialSessionUpdate) {\n        sessionUpdate.input_audio_transcription = { model: 'grok-stt' }\n      }\n\n      if (Object.keys(sessionUpdate).length > 0) {\n        sendEvent({\n          type: 'session.update',\n          session: sessionUpdate,\n        })\n        hasSentInitialSessionUpdate = true\n      }\n    },\n\n    interrupt() {\n      sendEvent({ type: 'response.cancel' })\n      currentMode = 'listening'\n      emit('mode_change', { mode: 'listening' })\n      emit('interrupted', { messageId: currentMessageId ?? undefined })\n    },\n\n    on<TEvent extends RealtimeEvent>(\n      event: TEvent,\n      handler: RealtimeEventHandler<TEvent>,\n    ): () => void {\n      if (!eventHandlers.has(event)) {\n        eventHandlers.set(event, new Set())\n      }\n      eventHandlers.get(event)!.add(handler)\n\n      return () => {\n        eventHandlers.get(event)?.delete(handler)\n      }\n    },\n\n    getAudioVisualization(): AudioVisualization {\n      function calculateLevel(analyser: AnalyserNode): number {\n        const data = new Uint8Array(analyser.fftSize)\n        analyser.getByteTimeDomainData(data)\n\n        let maxDeviation = 0\n        for (const sample of data) {\n          const deviation = Math.abs(sample - 128)\n          if (deviation > maxDeviation) {\n            maxDeviation = deviation\n          }\n        }\n\n        const normalized = maxDeviation / 128\n        return Math.min(1, normalized * 1.5)\n      }\n\n      return {\n        get inputLevel() {\n          if (!inputAnalyser) return 0\n          return calculateLevel(inputAnalyser)\n        },\n\n        get outputLevel() {\n          if (!outputAnalyser) return 0\n          return calculateLevel(outputAnalyser)\n        },\n\n        getInputFrequencyData() {\n          if (!inputAnalyser)\n            return new Uint8Array(FALLBACK_FREQUENCY_BIN_COUNT)\n          const data = new Uint8Array(inputAnalyser.frequencyBinCount)\n          inputAnalyser.getByteFrequencyData(data)\n          return data\n        },\n\n        getOutputFrequencyData() {\n          if (!outputAnalyser)\n            return new Uint8Array(FALLBACK_FREQUENCY_BIN_COUNT)\n          const data = new Uint8Array(outputAnalyser.frequencyBinCount)\n          outputAnalyser.getByteFrequencyData(data)\n          return data\n        },\n\n        getInputTimeDomainData() {\n          if (!inputAnalyser)\n            return new Uint8Array(FALLBACK_TIME_DOMAIN_SIZE).fill(\n              FALLBACK_TIME_DOMAIN_FILL,\n            )\n          const data = new Uint8Array(inputAnalyser.fftSize)\n          inputAnalyser.getByteTimeDomainData(data)\n          return data\n        },\n\n        getOutputTimeDomainData() {\n          if (!outputAnalyser)\n            return new Uint8Array(FALLBACK_TIME_DOMAIN_SIZE).fill(\n              FALLBACK_TIME_DOMAIN_FILL,\n            )\n          const data = new Uint8Array(outputAnalyser.fftSize)\n          outputAnalyser.getByteTimeDomainData(data)\n          return data\n        },\n\n        get inputSampleRate() {\n          return 24000\n        },\n\n        get outputSampleRate() {\n          return 24000\n        },\n      }\n    },\n  }\n\n  // `dataChannelReady` was already awaited inside the post-SDP try/catch\n  // above so we can short-circuit on failures with full teardown.\n  return 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package/dist/esm/realtime/index.d.ts

@@ -0,0 +1,4 @@
+export { grokRealtimeToken } from './token.js';
+export { grokRealtime } from './adapter.js';
+export type { GrokRealtimeVoice, GrokRealtimeTokenOptions, GrokRealtimeOptions, GrokTurnDetection, GrokSemanticVADConfig, GrokServerVADConfig, } from './types.js';
+export type { GrokRealtimeModel } from '../model-meta.js';

package/dist/esm/realtime/realtime-contract.d.ts

@@ -0,0 +1,30 @@
+import { AnyClientTool, AudioVisualization, RealtimeEvent, RealtimeEventHandler, RealtimeSessionConfig, RealtimeToken } from '@tanstack/ai';
+/**
+ * Structural contract for the `RealtimeAdapter` / `RealtimeConnection` types
+ * from `@tanstack/ai-client`.
+ *
+ * We duplicate the shapes here (and verify structural compatibility in a
+ * dev-only type check — see `tests/realtime-contract.drift.test-d.ts`) so that
+ * `@tanstack/ai-grok` does not impose `@tanstack/ai-client` as a `peerDependency`.
+ * Consumers only need `@tanstack/ai-client` at the point where they actually
+ * construct a `RealtimeClient`, not when they import this adapter.
+ *
+ * If `@tanstack/ai-client` ever changes these interfaces, the drift check
+ * will fail and we must update this file in lockstep.
+ */
+export interface RealtimeAdapter {
+    provider: string;
+    connect: (token: RealtimeToken, clientTools?: ReadonlyArray<AnyClientTool>) => Promise<RealtimeConnection>;
+}
+export interface RealtimeConnection {
+    disconnect: () => Promise<void>;
+    startAudioCapture: () => Promise<void>;
+    stopAudioCapture: () => void;
+    sendText: (text: string) => void;
+    sendImage: (imageData: string, mimeType: string) => void;
+    sendToolResult: (callId: string, result: string) => void;
+    updateSession: (config: Partial<RealtimeSessionConfig>) => void;
+    interrupt: () => void;
+    on: <TEvent extends RealtimeEvent>(event: TEvent, handler: RealtimeEventHandler<TEvent>) => () => void;
+    getAudioVisualization: () => AudioVisualization;
+}

package/dist/esm/realtime/token.d.ts

@@ -0,0 +1,22 @@
+import { RealtimeTokenAdapter } from '@tanstack/ai';
+import { GrokRealtimeTokenOptions } from './types.js';
+/**
+ * Creates a Grok realtime token adapter.
+ *
+ * Generates ephemeral client secrets for browser-side WebRTC connections to
+ * the xAI Voice Agent API.
+ *
+ * @param options - Configuration options for the realtime session.
+ * @returns A RealtimeTokenAdapter for use with `realtimeToken()`.
+ *
+ * @example
+ * ```typescript
+ * import { realtimeToken } from '@tanstack/ai'
+ * import { grokRealtimeToken } from '@tanstack/ai-grok'
+ *
+ * const token = await realtimeToken({
+ *   adapter: grokRealtimeToken({ model: 'grok-voice-fast-1.0' }),
+ * })
+ * ```
+ */
+export declare function grokRealtimeToken(options?: GrokRealtimeTokenOptions): RealtimeTokenAdapter;

package/dist/esm/realtime/token.js

@@ -0,0 +1,73 @@
+import { resolveDebugOption } from "@tanstack/ai/adapter-internals";
+import { getGrokApiKeyFromEnv } from "../utils/client.js";
+const GROK_REALTIME_CLIENT_SECRETS_URL = "https://api.x.ai/v1/realtime/client_secrets";
+const DEFAULT_TOKEN_FETCH_TIMEOUT_MS = 15e3;
+function grokRealtimeToken(options = {}) {
+  const apiKey = getGrokApiKeyFromEnv();
+  const logger = resolveDebugOption(options.debug);
+  return {
+    provider: "grok",
+    async generateToken() {
+      const model = options.model ?? "grok-voice-fast-1.0";
+      logger.request(`activity=realtimeToken provider=grok model=${model}`, {
+        provider: "grok",
+        model
+      });
+      const requestBody = {
+        session: { model }
+      };
+      const controller = new AbortController();
+      const timeout = setTimeout(
+        () => controller.abort(new Error("Grok realtime token request timed out")),
+        DEFAULT_TOKEN_FETCH_TIMEOUT_MS
+      );
+      try {
+        const response = await fetch(GROK_REALTIME_CLIENT_SECRETS_URL, {
+          method: "POST",
+          headers: {
+            Authorization: `Bearer ${apiKey}`,
+            "Content-Type": "application/json"
+          },
+          body: JSON.stringify(requestBody),
+          signal: controller.signal
+        });
+        if (!response.ok) {
+          const errorText = await response.text();
+          throw new Error(
+            `Grok realtime session creation failed: ${response.status} ${errorText}`
+          );
+        }
+        const sessionData = await response.json();
+        const clientSecret = sessionData?.client_secret;
+        if (!clientSecret || typeof clientSecret.value !== "string" || typeof clientSecret.expires_at !== "number" || !Number.isFinite(clientSecret.expires_at)) {
+          throw new Error(
+            "Grok realtime session response missing or malformed `client_secret`"
+          );
+        }
+        const sessionModel = sessionData.model ?? model;
+        const raw = clientSecret.expires_at;
+        const expiresAt = raw > 1e12 ? raw : raw * 1e3;
+        return {
+          provider: "grok",
+          token: clientSecret.value,
+          expiresAt,
+          config: {
+            model: sessionModel
+          }
+        };
+      } catch (error) {
+        logger.errors("grok.realtimeToken fatal", {
+          error,
+          source: "grok.realtimeToken"
+        });
+        throw error;
+      } finally {
+        clearTimeout(timeout);
+      }
+    }
+  };
+}
+export {
+  grokRealtimeToken
+};
+//# sourceMappingURL=token.js.map

package/dist/esm/realtime/token.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"token.js","sources":["../../../src/realtime/token.ts"],"sourcesContent":["import { resolveDebugOption } from '@tanstack/ai/adapter-internals'\nimport { getGrokApiKeyFromEnv } from '../utils'\nimport type { RealtimeToken, RealtimeTokenAdapter } from '@tanstack/ai'\nimport type { GrokRealtimeModel } from '../model-meta'\nimport type {\n  GrokRealtimeSessionResponse,\n  GrokRealtimeTokenOptions,\n} from './types'\n\nconst GROK_REALTIME_CLIENT_SECRETS_URL =\n  'https://api.x.ai/v1/realtime/client_secrets'\n\nconst DEFAULT_TOKEN_FETCH_TIMEOUT_MS = 15_000\n\n/**\n * Creates a Grok realtime token adapter.\n *\n * Generates ephemeral client secrets for browser-side WebRTC connections to\n * the xAI Voice Agent API.\n *\n * @param options - Configuration options for the realtime session.\n * @returns A RealtimeTokenAdapter for use with `realtimeToken()`.\n *\n * @example\n * ```typescript\n * import { realtimeToken } from '@tanstack/ai'\n * import { grokRealtimeToken } from '@tanstack/ai-grok'\n *\n * const token = await realtimeToken({\n *   adapter: grokRealtimeToken({ model: 'grok-voice-fast-1.0' }),\n * })\n * ```\n */\nexport function grokRealtimeToken(\n  options: GrokRealtimeTokenOptions = {},\n): RealtimeTokenAdapter {\n  const apiKey = getGrokApiKeyFromEnv()\n  const logger = resolveDebugOption(options.debug)\n\n  return {\n    provider: 'grok',\n\n    async generateToken(): Promise<RealtimeToken> {\n      const model: GrokRealtimeModel = options.model ?? 'grok-voice-fast-1.0'\n\n      logger.request(`activity=realtimeToken provider=grok model=${model}`, {\n        provider: 'grok',\n        model,\n      })\n\n      // xAI docs (docs.x.ai/developers/rest-api-reference/inference/voice)\n      // specify the body as `{ session: { model } }`. `expires_after` is\n      // available to shorten the default 600s TTL but we don't expose it\n      // yet — the caller can still call `generateToken()` more often if\n      // they want a shorter-lived session.\n      const requestBody: Record<string, unknown> = {\n        session: { model },\n      }\n\n      // Abort the fetch if xAI never responds. Without this the whole\n      // realtime connect flow hangs forever on a dead endpoint.\n      const controller = new AbortController()\n      const timeout = setTimeout(\n        () =>\n          controller.abort(new Error('Grok realtime token request timed out')),\n        DEFAULT_TOKEN_FETCH_TIMEOUT_MS,\n      )\n\n      try {\n        const response = await fetch(GROK_REALTIME_CLIENT_SECRETS_URL, {\n          method: 'POST',\n          headers: {\n            Authorization: `Bearer ${apiKey}`,\n            'Content-Type': 'application/json',\n          },\n          body: JSON.stringify(requestBody),\n          signal: controller.signal,\n        })\n\n        if (!response.ok) {\n          const errorText = await response.text()\n          throw new Error(\n            `Grok realtime session creation failed: ${response.status} ${errorText}`,\n          )\n        }\n\n        const sessionData = (await response.json()) as\n          | Partial<GrokRealtimeSessionResponse>\n          | undefined\n\n        // Validate shape before dereferencing — xAI could return an error\n        // envelope with 200 status, or a partial response under protocol drift.\n        const clientSecret = sessionData?.client_secret\n        if (\n          !clientSecret ||\n          typeof clientSecret.value !== 'string' ||\n          typeof clientSecret.expires_at !== 'number' ||\n          !Number.isFinite(clientSecret.expires_at)\n        ) {\n          throw new Error(\n            'Grok realtime session response missing or malformed `client_secret`',\n          )\n        }\n        const sessionModel = sessionData.model ?? model\n\n        // xAI docs describe `expires_at` as a unix timestamp in seconds, but\n        // in practice different deployments have returned milliseconds. Treat\n        // any value that already looks like ms (>1e12 ≈ Sep 2001 in ms) as ms.\n        const raw = clientSecret.expires_at\n        const expiresAt = raw > 1e12 ? raw : raw * 1000\n\n        return {\n          provider: 'grok',\n          token: clientSecret.value,\n          expiresAt,\n          config: {\n            model: sessionModel,\n          },\n        }\n      } catch (error) {\n        logger.errors('grok.realtimeToken fatal', {\n          error,\n          source: 'grok.realtimeToken',\n        })\n        throw error\n      } finally {\n        clearTimeout(timeout)\n      }\n    },\n  }\n}\n"],"names":[],"mappings":";;AASA,MAAM,mCACJ;AAEF,MAAM,iCAAiC;AAqBhC,SAAS,kBACd,UAAoC,IACd;AACtB,QAAM,SAAS,qBAAA;AACf,QAAM,SAAS,mBAAmB,QAAQ,KAAK;AAE/C,SAAO;AAAA,IACL,UAAU;AAAA,IAEV,MAAM,gBAAwC;AAC5C,YAAM,QAA2B,QAAQ,SAAS;AAElD,aAAO,QAAQ,8CAA8C,KAAK,IAAI;AAAA,QACpE,UAAU;AAAA,QACV;AAAA,MAAA,CACD;AAOD,YAAM,cAAuC;AAAA,QAC3C,SAAS,EAAE,MAAA;AAAA,MAAM;AAKnB,YAAM,aAAa,IAAI,gBAAA;AACvB,YAAM,UAAU;AAAA,QACd,MACE,WAAW,MAAM,IAAI,MAAM,uCAAuC,CAAC;AAAA,QACrE;AAAA,MAAA;AAGF,UAAI;AACF,cAAM,WAAW,MAAM,MAAM,kCAAkC;AAAA,UAC7D,QAAQ;AAAA,UACR,SAAS;AAAA,YACP,eAAe,UAAU,MAAM;AAAA,YAC/B,gBAAgB;AAAA,UAAA;AAAA,UAElB,MAAM,KAAK,UAAU,WAAW;AAAA,UAChC,QAAQ,WAAW;AAAA,QAAA,CACpB;AAED,YAAI,CAAC,SAAS,IAAI;AAChB,gBAAM,YAAY,MAAM,SAAS,KAAA;AACjC,gBAAM,IAAI;AAAA,YACR,0CAA0C,SAAS,MAAM,IAAI,SAAS;AAAA,UAAA;AAAA,QAE1E;AAEA,cAAM,cAAe,MAAM,SAAS,KAAA;AAMpC,cAAM,eAAe,aAAa;AAClC,YACE,CAAC,gBACD,OAAO,aAAa,UAAU,YAC9B,OAAO,aAAa,eAAe,YACnC,CAAC,OAAO,SAAS,aAAa,UAAU,GACxC;AACA,gBAAM,IAAI;AAAA,YACR;AAAA,UAAA;AAAA,QAEJ;AACA,cAAM,eAAe,YAAY,SAAS;AAK1C,cAAM,MAAM,aAAa;AACzB,cAAM,YAAY,MAAM,OAAO,MAAM,MAAM;AAE3C,eAAO;AAAA,UACL,UAAU;AAAA,UACV,OAAO,aAAa;AAAA,UACpB;AAAA,UACA,QAAQ;AAAA,YACN,OAAO;AAAA,UAAA;AAAA,QACT;AAAA,MAEJ,SAAS,OAAO;AACd,eAAO,OAAO,4BAA4B;AAAA,UACxC;AAAA,UACA,QAAQ;AAAA,QAAA,CACT;AACD,cAAM;AAAA,MACR,UAAA;AACE,qBAAa,OAAO;AAAA,MACtB;AAAA,IACF;AAAA,EAAA;AAEJ;"}

package/dist/esm/realtime/types.d.ts

@@ -0,0 +1,95 @@
+import { DebugOption, VADConfig } from '@tanstack/ai';
+import { GrokRealtimeModel } from '../model-meta.js';
+/**
+ * Grok realtime voice options (Voice Agent API).
+ * https://docs.x.ai/developers/model-capabilities/audio/voice-agent
+ */
+export type GrokRealtimeVoice = 'eve' | 'ara' | 'rex' | 'sal' | 'leo';
+/**
+ * Grok semantic VAD configuration.
+ */
+export interface GrokSemanticVADConfig {
+    type: 'semantic_vad';
+    /** Eagerness level for turn detection */
+    eagerness?: 'low' | 'medium' | 'high';
+}
+/**
+ * Grok server VAD configuration.
+ */
+export interface GrokServerVADConfig extends VADConfig {
+    type: 'server_vad';
+}
+/**
+ * Grok turn detection configuration.
+ */
+export type GrokTurnDetection = GrokSemanticVADConfig | GrokServerVADConfig | null;
+/**
+ * Options for the Grok realtime token adapter.
+ */
+export interface GrokRealtimeTokenOptions {
+    /** Model to use (default: 'grok-voice-fast-1.0'). */
+    model?: GrokRealtimeModel;
+    /**
+     * Enable debug logging for token creation.
+     *
+     * - `true`: log all categories via the default `ConsoleLogger`
+     * - `false`: silence everything including errors
+     * - `DebugConfig`: per-category toggles plus an optional custom `logger`
+     * - omitted: only the `errors` category is active (default behaviour)
+     */
+    debug?: DebugOption;
+}
+/**
+ * Options for the Grok realtime client adapter.
+ */
+export interface GrokRealtimeOptions {
+    /** Connection mode (default: 'webrtc' in browser). */
+    connectionMode?: 'webrtc' | 'websocket';
+    /**
+     * Enable debug logging for this adapter.
+     *
+     * - `true`: log all categories via the default `ConsoleLogger`
+     * - `false`: silence everything including errors
+     * - `DebugConfig`: per-category toggles plus an optional custom `logger`
+     * - omitted: only the `errors` category is active (default behaviour)
+     */
+    debug?: DebugOption;
+}
+/**
+ * Grok realtime session response from the `/v1/realtime/client_secrets`
+ * endpoint. Shape matches OpenAI's `/v1/realtime/sessions` response since
+ * xAI advertises its voice agent API as OpenAI-realtime-compatible.
+ */
+export interface GrokRealtimeSessionResponse {
+    id: string;
+    object: string;
+    model: string;
+    modalities: Array<string>;
+    instructions: string;
+    voice: string;
+    input_audio_format: string;
+    output_audio_format: string;
+    input_audio_transcription: {
+        model: string;
+    } | null;
+    turn_detection: {
+        type: string;
+        threshold?: number;
+        prefix_padding_ms?: number;
+        silence_duration_ms?: number;
+        eagerness?: string;
+    } | null;
+    tools: Array<{
+        type: string;
+        name: string;
+        description: string;
+        parameters: Record<string, unknown>;
+    }>;
+    tool_choice: string;
+    temperature: number;
+    max_response_output_tokens: number | string;
+    client_secret: {
+        value: string;
+        expires_at: number;
+    };
+}

package/dist/esm/utils/audio.d.ts

@@ -0,0 +1,23 @@
+/**
+ * Coerce the various audio input shapes accepted by `TranscriptionOptions.audio`
+ * into a `File` suitable for `multipart/form-data` uploads.
+ *
+ * For base64 string inputs we require an explicit MIME type — either via a
+ * `data:<mime>;base64,<payload>` URI prefix, or via the caller-provided
+ * `audioFormat` parameter. Bare base64 without either is rejected, because
+ * silently defaulting to `audio/mpeg` misreports non-mp3 audio to the server.
+ *
+ * The same rule applies to raw `ArrayBuffer` inputs: the caller must supply
+ * an `audioFormat` so we know what MIME type and extension to use.
+ */
+export declare function toAudioFile(audio: string | File | Blob | ArrayBuffer, audioFormat?: string): File;
+/**
+ * Cross-runtime ArrayBuffer → base64 conversion.
+ *
+ * Uses Node's `Buffer` when available (fastest path on server) and falls
+ * back to `btoa` + chunked `String.fromCharCode` everywhere else (browser,
+ * Cloudflare Workers, Bun, Deno). Chunking is required because a very
+ * large audio buffer spread into `String.fromCharCode(...bytes)` in one
+ * call can hit `Maximum call stack size exceeded`.
+ */
+export declare function arrayBufferToBase64(buffer: ArrayBuffer): string;

package/dist/esm/utils/audio.js

@@ -0,0 +1,171 @@
+function toAudioFile(audio, audioFormat) {
+  if (typeof File !== "undefined" && audio instanceof File) {
+    if (audioFormat) {
+      const mimeType = toMimeType(audioFormat);
+      return new File([audio], `audio.${extensionFor(mimeType)}`, {
+        type: mimeType
+      });
+    }
+    if (audio.type) {
+      return audio;
+    }
+    throw new Error(
+      'toAudioFile cannot infer type for File input with empty .type — pass an explicit audioFormat (e.g. "mp3", "wav", "audio/mpeg")'
+    );
+  }
+  if (typeof Blob !== "undefined" && audio instanceof Blob) {
+    const mimeType = audioFormat ? toMimeType(audioFormat) : audio.type || void 0;
+    if (!mimeType) {
+      throw new Error(
+        'toAudioFile cannot infer type for Blob input with empty .type — pass an explicit audioFormat (e.g. "mp3", "wav", "audio/mpeg")'
+      );
+    }
+    return new File([audio], `audio.${extensionFor(mimeType)}`, {
+      type: mimeType
+    });
+  }
+  if (audio instanceof ArrayBuffer) {
+    if (!audioFormat) {
+      throw new Error(
+        'toAudioFile cannot infer type for ArrayBuffer input — pass an explicit audioFormat (e.g. "mp3", "wav", "audio/mpeg")'
+      );
+    }
+    const mimeType = toMimeType(audioFormat);
+    return new File([audio], `audio.${extensionFor(mimeType)}`, {
+      type: mimeType
+    });
+  }
+  if (typeof audio === "string") {
+    if (audio.startsWith("data:")) {
+      const [header, base64Data] = audio.split(",");
+      const headerMatch = header?.match(/data:([^;]+)/);
+      const uriMimeType = headerMatch?.[1];
+      if (!uriMimeType) {
+        throw new Error(
+          "Malformed data: URI in toAudioFile: cannot parse MIME type — expected data:<mime>[;charset=…][;base64],<payload>"
+        );
+      }
+      if (base64Data === void 0 || base64Data.trim() === "") {
+        throw new Error(
+          "Malformed data: URI in toAudioFile: missing base64 payload after comma"
+        );
+      }
+      const mimeType2 = audioFormat ? toMimeType(audioFormat) : uriMimeType;
+      const buffer2 = base64ToArrayBuffer(base64Data);
+      return new File([buffer2], `audio.${extensionFor(mimeType2)}`, {
+        type: mimeType2
+      });
+    }
+    if (!audioFormat) {
+      throw new Error(
+        "toAudioFile requires a data: URI (e.g. data:audio/wav;base64,...) or an explicit audioFormat argument — bare base64 strings have no MIME type to infer"
+      );
+    }
+    const buffer = base64ToArrayBuffer(audio);
+    const mimeType = toMimeType(audioFormat);
+    return new File([buffer], `audio.${extensionFor(mimeType)}`, {
+      type: mimeType
+    });
+  }
+  throw new Error("Invalid audio input type");
+}
+function toMimeType(audioFormat) {
+  if (audioFormat.includes("/")) return audioFormat;
+  const ext = audioFormat.toLowerCase();
+  switch (ext) {
+    case "mp3":
+      return "audio/mpeg";
+    case "wav":
+      return "audio/wav";
+    case "ogg":
+      return "audio/ogg";
+    case "opus":
+      return "audio/opus";
+    case "flac":
+      return "audio/flac";
+    case "aac":
+      return "audio/aac";
+    case "mp4":
+      return "audio/mp4";
+    case "m4a":
+      return "audio/mp4";
+    case "webm":
+      return "audio/webm";
+    case "pcm":
+      return "audio/L16";
+    case "mulaw":
+      return "audio/basic";
+    case "alaw":
+      return "audio/x-alaw-basic";
+    default:
+      return `audio/${ext}`;
+  }
+}
+function extensionFor(mimeType) {
+  switch (mimeType) {
+    case "audio/mpeg":
+      return "mp3";
+    case "audio/wav":
+    case "audio/x-wav":
+      return "wav";
+    case "audio/ogg":
+      return "ogg";
+    case "audio/opus":
+      return "opus";
+    case "audio/flac":
+      return "flac";
+    case "audio/aac":
+      return "aac";
+    case "audio/mp4":
+      return "m4a";
+    case "audio/webm":
+      return "webm";
+    case "audio/L16":
+      return "pcm";
+    case "audio/basic":
+      return "mulaw";
+    case "audio/x-alaw-basic":
+      return "alaw";
+    default: {
+      const slash = mimeType.indexOf("/");
+      if (slash === -1) return "bin";
+      return mimeType.slice(slash + 1) || "bin";
+    }
+  }
+}
+function arrayBufferToBase64(buffer) {
+  if (typeof Buffer !== "undefined") {
+    return Buffer.from(buffer).toString("base64");
+  }
+  const bytes = new Uint8Array(buffer);
+  const chunkSize = 32768;
+  let binary = "";
+  for (let i = 0; i < bytes.length; i += chunkSize) {
+    const end = Math.min(i + chunkSize, bytes.length);
+    binary += String.fromCharCode.apply(
+      null,
+      bytes.subarray(i, end)
+    );
+  }
+  return btoa(binary);
+}
+function base64ToArrayBuffer(base64) {
+  let binary;
+  try {
+    binary = atob(base64);
+  } catch (err) {
+    const msg = err instanceof Error ? err.message : String(err);
+    throw new Error(`Invalid base64 input to toAudioFile: ${msg}`);
+  }
+  const buffer = new ArrayBuffer(binary.length);
+  const bytes = new Uint8Array(buffer);
+  for (let i = 0; i < binary.length; i++) {
+    bytes[i] = binary.charCodeAt(i);
+  }
+  return buffer;
+}
+export {
+  arrayBufferToBase64,
+  toAudioFile
+};
+//# sourceMappingURL=audio.js.map

package/dist/esm/utils/audio.js.map

@@ -0,0 +1 @@
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Bare base64 without either is rejected, because\n * silently defaulting to `audio/mpeg` misreports non-mp3 audio to the server.\n *\n * The same rule applies to raw `ArrayBuffer` inputs: the caller must supply\n * an `audioFormat` so we know what MIME type and extension to use.\n */\nexport function toAudioFile(\n  audio: string | File | Blob | ArrayBuffer,\n  audioFormat?: string,\n): File {\n  if (typeof File !== 'undefined' && audio instanceof File) {\n    // Prefer the caller-supplied `audioFormat` over a potentially empty or\n    // incorrect `File.type` — callers pass `audioFormat` precisely because\n    // they have more context than the browser does about the payload. 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package/dist/esm/utils/index.d.ts

@@ -1,2 +1,3 @@
 export { createGrokClient, getGrokApiKeyFromEnv, generateId, type GrokClientConfig, } from './client.js';
 export { makeGrokStructuredOutputCompatible, transformNullsToUndefined, } from './schema-converter.js';
+export { toAudioFile, arrayBufferToBase64 } from './audio.js';

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@tanstack/ai-grok",
-  "version": "0.6.7",
+  "version": "0.7.0",
   "description": "Grok (xAI) adapter for TanStack AI",
   "author": "",
   "license": "MIT",
@@ -38,11 +38,14 @@
   },
   "devDependencies": {
     "@vitest/coverage-v8": "4.0.14",
-    "vite": "^7.2.7"
+    "vite": "^7.2.7",
+    "zod": "^4.2.0",
+    "@tanstack/ai": "0.14.0",
+    "@tanstack/ai-client": "0.8.0"
   },
   "peerDependencies": {
     "zod": "^4.0.0",
-    "@tanstack/ai": "^0.12.0"
+    "@tanstack/ai": "^0.14.0"
   },
   "scripts": {
     "build": "vite build",