add-dictation

---
name: add-dictation
description: >-
  Use when the user runs /add-dictation or wants speech turned into text with
  Grok speech-to-text: a mic button that dictates into the composer, live
  captions, or transcribing recorded audio (files, uploads, URLs) with word
  timestamps, diarization, subtitles, meeting notes. STT, transcribe,
  transcription. For a voice agent that talks back use /add-voice.
---

# Add Dictation

Add Grok Speech to Text to an existing app: a mic button that dictates into the composer, live captions, or transcripts of recorded audio. Run on `/add-dictation`, typed **Dictate**, or clear “transcribe” intent. Cursor has no mic; wire the **app**, not the IDE.

## Docs

- https://docs.x.ai/developers/model-capabilities/audio/speech-to-text
- Model selection (read first): https://docs.x.ai/developers/model-capabilities/audio/speech-to-text#model-selection
- Pricing (cite docs only): https://docs.x.ai/developers/pricing

## Model

Before any STT call, read the model-selection section and pass the **latest listed** model as `model`. Do not invent ids or reuse a stale one from this skill. REST: form field before `file`. Streaming: query param. Official examples currently send `grok-voice-transcribe-2.0`. Listed today: `grok-voice-transcribe-2.0` (best), `grok-voice-transcribe-1.0` (original; API default when `model` is omitted).

## Pick the path

| Need | Path |
| --- | --- |
| Tap, speak, tap, text appears. Uploaded files. URLs. | **Batch** `POST https://api.x.ai/v1/stt` (default) |
| Text appears while speaking: captions, long dictation, push-to-talk | **Streaming** `wss://api.x.ai/v1/stt` through a backend relay |

Batch is the default for a composer mic button: one request, no socket, the key never leaves the server. Go streaming only when the UX needs interim text.

## Auth

- Bearer `XAI_API_KEY`, server side only. The STT docs document no ephemeral-token flow, and browsers cannot set WebSocket headers, so browser streaming goes through your backend relay. Do not invent a token flow.
- Never put the key in a client bundle. Do not paste keys in chat.

## Steps

1. **Map the app**
   - Composer or input component, where the text should land (insert at cursor vs replace), server framework, package manager.
   - The **microphone icon belongs to dictation**. If `/add-voice` is installed, its waveform primary button stays as is; add the mic as a secondary ghost button beside it.
   - Existing mic capture? If `/add-voice` ran, its PCM capture can feed streaming STT; pass its rate as `sample_rate`. 16 kHz is the model’s native rate; other supported rates (8000, 16000, 22050, 24000, 44100, 48000) are resampled server side.

2. **Batch path (default)**
   - Client: `MediaRecorder` → `Blob` → `POST` to your own route. The endpoint auto-detects containers (WAV, MP3, OGG, Opus, FLAC, AAC, MP4, M4A, MKV, WebM), so send whatever `MediaRecorder` produces.
   - Server: forward as `multipart/form-data`. Option fields first, **`file` last**; fields after `file` may be ignored. `file` or `url`, max 500 MB. `model` from docs (latest listed).

```ts
// server (any runtime with fetch + FormData)
export async function transcribe(blob: Blob, filename: string) {
  const form = new FormData();
  form.append("model", "grok-voice-transcribe-2.0"); // latest from docs; re-read model-selection
  form.append("format", "true");     // written-form numbers/currency; requires language
  form.append("language", "en");
  // form.append("keyterm", "Acme"); // repeat per term, ≤100 terms × 50 chars
  form.append("file", blob, filename); // last
  const res = await fetch("https://api.x.ai/v1/stt", {
    method: "POST",
    headers: { Authorization: `Bearer ${process.env.XAI_API_KEY}` },
    body: form,
  });
  if (!res.ok) throw new Error(`STT ${res.status}`); // 400 bad input, 413 >500 MB, 429 back off, 502 url fetch failed, 503 retry
  return (await res.json()) as {
    text: string; language: string; duration: number;
    words?: { text: string; start: number; end: number; speaker?: number }[];
    channels?: { index: number; text: string; words: unknown[] }[];
  };
}
```

```ts
// client
const mime = MediaRecorder.isTypeSupported("audio/webm;codecs=opus") ? "audio/webm;codecs=opus" : "audio/mp4";
const rec = new MediaRecorder(stream, { mimeType: mime });
const parts: BlobPart[] = [];
rec.ondataavailable = (e) => parts.push(e.data);
rec.onstop = async () => {
  const fd = new FormData();
  fd.append("file", new Blob(parts, { type: mime }), "dictation");
  const { text } = await (await fetch("/api/dictation", { method: "POST", body: fd })).json();
  insertAtCursor(text);
};
rec.start(); // second tap: rec.stop()
```

3. **Streaming path**
   - Relay: server holds the key, upgrades the browser socket, forwards binary frames and client control messages up, JSON events down. Build the query string server side. `model` from docs (latest listed).

```ts
import { WebSocketServer, WebSocket } from "ws";

new WebSocketServer({ port: 8788 }).on("connection", (client) => {
  const q = new URLSearchParams({ model: "grok-voice-transcribe-2.0", sample_rate: "16000", encoding: "pcm", interim_results: "true", language: "en" });
  const up = new WebSocket(`wss://api.x.ai/v1/stt?${q}`, { headers: { Authorization: `Bearer ${process.env.XAI_API_KEY}` } });
  up.on("message", (d) => client.send(d.toString()));                       // transcript.* and error events
  client.on("message", (d, isBinary) => up.readyState === WebSocket.OPEN && up.send(d, { binary: isBinary })); // audio + finalize/audio.done
  const end = () => { client.close(); up.close(); };
  up.on("close", end); up.on("error", end); client.on("close", end);
});
```

   - Browser capture: PCM16 little-endian, mono, 16 kHz, **100 ms frames = 3,200 bytes**, raw binary, no base64. Wait for `transcript.created` before sending. `MediaRecorder` output is a container, not raw frames; do not stream it.

```ts
const ws = new WebSocket(relayUrl); ws.binaryType = "arraybuffer";
const ctx = new AudioContext({ sampleRate: 16000 }); // if ctx.sampleRate !== 16000, downsample in the worklet
await ctx.audioWorklet.addModule("/pcm16-worklet.js"); // Float32 → Int16LE, posts one 3,200-byte frame per 100 ms
const node = new AudioWorkletNode(ctx, "pcm16");
ctx.createMediaStreamSource(stream).connect(node);
let ready = false;
node.port.onmessage = (e) => ready && ws.readyState === WebSocket.OPEN && ws.send(e.data);

let committed = "", locked = "", live = "";
ws.addEventListener("message", (e) => {
  const ev = JSON.parse(e.data);
  if (ev.type === "transcript.created") ready = true;
  else if (ev.type === "transcript.partial") {
    if (ev.speech_final) { committed += ev.text + " "; locked = ""; live = ""; } // complete stitched utterance
    else if (ev.is_final) { locked += ev.text + " "; live = ""; }                // chunk final: text will not change
    else live = ev.text;                                                          // interim: may change
    render(committed + locked + live);
  } else if (ev.type === "transcript.done") ws.close();                          // after audio.done
  else if (ev.type === "error") showError(ev.message);                           // most errors close the socket
});
// stop: ws.send(JSON.stringify({ type: "audio.done" }))
// push-to-talk release: ws.send(JSON.stringify({ type: "Finalize" })) then keep streaming (docs show both `finalize` and `Finalize`; the examples use `Finalize`)
```

4. **Options** (query params for streaming, form fields for batch)

| Want | Set |
| --- | --- |
| Latest STT | Read docs #model-selection, pass that `model`. Snapshot: `grok-voice-transcribe-2.0`. Omit → `grok-voice-transcribe-1.0`. Batch: form field. Streaming: query param. |
| Text while speaking | `interim_results=true` |
| “one hundred dollars” → `$100` | streaming: `language=en`; batch: `format=true` + `language=en` |
| Product names, jargon | `keyterm=` repeated |
| Not cut off mid-sentence while dictating numbers | `smart_turn=0.7&smart_turn_timeout=3000` |
| Faster or slower end of utterance | `endpointing=` ms, default 400 |
| Who said what (meetings) | `diarize=true` → `words[].speaker` |
| Agent and customer on separate channels | `multichannel=true&channels=2` (PCM only, not Opus) |
| Keep “um”, “uh” | `filler_words=true` (removed by default) |
| Low bandwidth or mobile | `encoding=opus`, exactly one raw Opus packet per frame, omit `sample_rate` |
| Raw audio to batch | `audio_format=pcm|mulaw|alaw` + `sample_rate` |
| Quiet or telephony audio | lower `vad_threshold` (streaming default 0.08, batch 0.5) |

5. **Python twin (only if the server is Python)**

```python
import os, requests
r = requests.post(
    "https://api.x.ai/v1/stt",
    headers={"Authorization": f"Bearer {os.environ['XAI_API_KEY']}"},
    data=[("model", "grok-voice-transcribe-2.0"), ("format", "true"), ("language", "en")],
    files={"file": ("dictation.webm", blob, "audio/webm")},  # requests sends data fields before files
)
r.raise_for_status(); text = r.json()["text"]
# streaming: websockets.connect(url, additional_headers={"Authorization": f"Bearer {key}"}); await ws.send(pcm_bytes)
```

6. **Smoke**
   - Batch: latest `model` from docs (currently `grok-voice-transcribe-2.0`): `curl -X POST https://api.x.ai/v1/stt -H "Authorization: Bearer $XAI_API_KEY" -F model=grok-voice-transcribe-2.0 -F language=en -F file=@short.wav` → 200 with `text`. Same call with `-F format=true` and no `language` → 400.
   - Streaming: dictate two sentences with a pause between them. Expect interim text, then a final; no duplicated or vanished words at the utterance boundary (if words vanish, the stitched `speech_final` text did not include the chunk finals: append instead of replacing `locked`). `audio.done` → `transcript.done`, socket closes.
   - Search the client bundle for `XAI_API_KEY`; it must not be there.
   - Debug from logs with `/debug-voice`; swap its hook points to `transcript.*` events.

## Out of scope

- Speech that talks back (`/add-voice`), speaking text (`/add-read-aloud`)
- Inventing an STT token flow, endpoints, model ids, or event names not in the docs