add-read-aloud git:20260908.2b8ae2eA

Immutable. This exact content is served forever at /api/v1/blob/5b11114611f0950e.

---
name: add-read-aloud
description: >-
  Use when the user runs /add-read-aloud or wants the app to speak text with
  Grok text-to-speech: read-aloud button on assistant replies, auto-speak, TTS,
  voice output, narration, IVR prompts, speech tags, voice_id. For a two-way
  voice agent use /add-voice. For speech-to-text use /add-dictation.
---

# Add Read Aloud

Add Grok Text to Speech to an existing app: a speaker button on assistant replies, auto-speak, or narration of any text. Run on `/add-read-aloud`, typed **Read aloud**, or clear “speak this” / “TTS” intent. Cursor has no speaker; wire the **app**, not the IDE.

## Docs

- https://docs.x.ai/developers/model-capabilities/audio/text-to-speech
- API reference: https://docs.x.ai/developers/rest-api-reference/inference/voice
- Custom voices: https://docs.x.ai/developers/model-capabilities/audio/custom-voices
- Pricing (cite docs only): https://docs.x.ai/developers/pricing

## Pick the path

| Need | Path |
| --- | --- |
| Tap speaker, hear the finished reply. Narrate a page. Generate a file. | **Batch** `POST https://api.x.ai/v1/tts` (default) |
| Audio starts while the LLM is still streaming; barge-in; texts over 15,000 chars | **Streaming** `wss://api.x.ai/v1/tts` through a backend relay |

Batch is the default for a read-aloud button: one request, one MP3, cacheable, the key never leaves the server. Go streaming only when the UX needs audio before the text is complete. `POST /v1/tts` has no documented streaming flag; do not invent one.

## Auth

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

## Steps

1. **Map the app**
   - Where assistant messages render, where per-message actions live (copy, regenerate), how the reply stream ends, server framework, package manager.
   - The **speaker icon belongs to read aloud**. Waveform is voice mode (`/add-voice`), microphone is dictation (`/add-dictation`). Put a ghost speaker button in the message action row; loading shows a spinner, playing shows a stop square. One utterance at a time: starting a new one stops the current one.
   - Align the action row to the reply’s **text edge**, not the button’s box: an icon button centres its glyph, so if the assistant bubble has no padding pull the row left by that inset (e.g. `-ml-[5px]` for a 12–14 px icon in a 24 px button). Measure in the browser; `getBoundingClientRect` on the `<p>` and the `<svg>` should share a left edge.
   - Render the button only once the reply has **finished streaming**; on a live message it would speak a partial reply.
   - With many messages on screen, keep player state (active message id, `loading | playing`, last error) in one shared store (`useSyncExternalStore`, a signal, whatever the app uses) so every button reflects it and errors can surface in the app's existing status area. A per-button `let current` is not enough.
   - Auto-speak: opt-in toggle, off by default, and only after a user gesture on the page (autoplay policy). Never auto-speak on load.
   - If `/add-voice` is installed, its `AudioContext` and PCM player can play streaming TTS; do not add a second audio graph.

2. **Prepare the text**
   - Speak prose, not markup. Strip markdown: headings → text, `**bold**` → text, links → link text, inline code → the code, fenced blocks → `[pause] Code block omitted.`, tables → one sentence per row or omit. Keep punctuation; it drives pacing.
   - Neutralise speech tags that arrive inside the reply (`[laugh]`, `<whisper>`…) so the model’s text cannot steer delivery. Strip only the **documented tag names** (list in step 5), not every bracket: `[1]` citations and `[note]` must survive.
   - Batch limit is **15,000 characters per request**. Split longer text on paragraph, then sentence, then word boundaries and play the parts in order; fetch part N+1 while N plays or there is a silent gap at every boundary. Or use streaming.
   - Cache by `hash(text + voice_id + language + speed)`; the same reply is often replayed.

3. **Batch path (default)**
   - Server: your route takes `{ text, voice_id?, language? }`, validates the shape of each (`voice_id` `^[a-z0-9-]{1,64}$`, `language` BCP-47 or `auto`), forwards JSON, streams the body back with the upstream `Content-Type` and `Cache-Control: no-store`. Map upstream 404 to “unknown voice” so the client gets a readable error. Default output is MP3 at 24 kHz / 128 kbps, playable everywhere in the browser.
   - `language`: default to `"auto"` for a chat app, where replies follow the user’s language; pin `"en"` etc. only for fixed-language products.

```ts
// server (any runtime with fetch)
export async function speak(text: string, voice_id = "eve", language = "auto") {
  if (!text.trim() || text.length > 15_000) throw new Error("TTS text must be 1–15,000 chars");
  const res = await fetch("https://api.x.ai/v1/tts", {
    method: "POST",
    headers: { Authorization: `Bearer ${process.env.XAI_API_KEY}`, "Content-Type": "application/json" },
    body: JSON.stringify({
      text,
      voice_id,
      language,                       // required: `auto` or BCP-47 (`en`, `pt-BR`); omitting it → 422
      // output_format: { codec: "mp3", sample_rate: 24000, bit_rate: 128000 }, // default
      // speed: 1.0,                  // 0.7–1.5
      // text_normalization: true,    // "$5" → "five dollars"
      // replace: { nginx: "/ˈɛndʒɪn ˈɛks/" },
    }),
  });
  if (!res.ok) throw new Error(`TTS ${res.status}`); // 400 bad text/format, 401 key, 404 unknown voice_id, 422 missing required field (e.g. language), 429/500/503 back off and retry
  return new Response(res.body, { headers: { "Content-Type": res.headers.get("content-type") ?? "audio/mpeg" } });
}
```

```ts
// client
let current: HTMLAudioElement | null = null;
async function readAloud(text: string, voiceId = "eve") {
  current?.pause(); current = null;                       // one utterance at a time
  const res = await fetch("/api/tts", { method: "POST", headers: { "Content-Type": "application/json" }, body: JSON.stringify({ text, voice_id: voiceId }) });
  if (!res.ok) throw new Error("TTS request failed");
  const url = URL.createObjectURL(await res.blob());
  const audio = new Audio(url);
  audio.addEventListener("ended", () => URL.revokeObjectURL(url)); // avoid blob leaks
  current = audio;
  await audio.play();                                      // call from the click handler’s promise chain
}
function stop() { current?.pause(); current = null; }
```

   - Safari: `audio.duration` is `Infinity` on blob URLs. If you need a progress bar, decode with `AudioContext.decodeAudioData(buf.slice(0))` or request `with_timestamps: true` and read `duration` from the JSON envelope (audio is then base64 in `audio`).
   - Safari suspends an `AudioContext` created outside a gesture for good. Create it synchronously in the click handler, before any `await`.

4. **Streaming path**
   - Relay: server holds the key, upgrades the browser socket, builds the query string, forwards client JSON up and server JSON down. Request `codec=pcm` for the browser: raw PCM16 chunks can be scheduled as they arrive, while MP3 chunks cannot be decoded piecemeal without `MediaSource`.

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

new WebSocketServer({ port: 8789 }).on("connection", (client) => {
  const q = new URLSearchParams({ language: "en", voice: "eve", codec: "pcm", sample_rate: "24000" /* optimize_streaming_latency: "1" */ });
  const up = new WebSocket(`wss://api.x.ai/v1/tts?${q}`, { headers: { Authorization: `Bearer ${process.env.XAI_API_KEY}` } });
  up.on("message", (d) => client.send(d.toString()));                                   // audio.delta, audio.done, audio.clear, session.updated, error
  client.on("message", (d) => up.readyState === WebSocket.OPEN && up.send(d.toString())); // text.delta, text.done, text.clear, session.update
  const end = () => { client.close(); up.close(); };
  up.on("close", end); up.on("error", end); client.on("close", end);
});
```

   - Client: one socket per chat session (it stays open across utterances; 50 concurrent sessions per team, permit TTL 600 s, so reconnect on close). Forward LLM tokens as `text.delta` (each ≤ 15,000 chars), send `text.done` when the reply finishes, `text.clear` on stop or when a new reply starts; drop queued audio on `audio.clear`.

```ts
const ws = new WebSocket(relayUrl);
const ctx = new AudioContext({ sampleRate: 24000 }); // create in the click handler; resume if suspended
let playhead = 0, sources: AudioBufferSourceNode[] = [];

ws.addEventListener("message", (e) => {
  const ev = JSON.parse(e.data);
  if (ev.type === "audio.delta") {
    const bytes = Uint8Array.from(atob(ev.delta), (c) => c.charCodeAt(0));
    const pcm = new Int16Array(bytes.buffer, 0, bytes.byteLength >> 1);
    const buf = ctx.createBuffer(1, pcm.length, 24000);
    const ch = buf.getChannelData(0);
    for (let i = 0; i < pcm.length; i++) ch[i] = pcm[i] / 32768;
    const src = ctx.createBufferSource(); src.buffer = buf; src.connect(ctx.destination);
    playhead = Math.max(playhead, ctx.currentTime + 0.15);   // ~150 ms lead so chunks butt together
    src.start(playhead); playhead += buf.duration; sources.push(src);
  } else if (ev.type === "audio.done") { /* utterance finished; socket stays open */ }
  else if (ev.type === "audio.clear") { sources.forEach((s) => s.stop()); sources = []; playhead = 0; }
  else if (ev.type === "error") showError(ev.message);
});
// on each LLM token:  ws.send(JSON.stringify({ type: "text.delta", delta: token }))
// on reply finished:  ws.send(JSON.stringify({ type: "text.done" }))
// on stop / barge-in: ws.send(JSON.stringify({ type: "text.clear" }))  → wait for audio.clear before the next text.delta
```

   - Words split across `text.delta` boundaries are fine; matching and synthesis run across deltas.

5. **Options** (JSON fields for batch, query params for streaming)

| Want | Set |
| --- | --- |
| Different voice | `voice_id` (batch) / `voice` (streaming). Built-ins from `GET /v1/tts/voices`: `eve` (default), `ara`, `rex`, `leo`, `luna`, `atlas`, `aurora`, `orion`, … 28 total, all multilingual, case-insensitive. Custom voice: 8-char id from the console or `GET /v1/custom-voices` |
| Non-English or mixed | `language`: `en`, `ar-EG`, `ar-SA`, `ar-AE`, `bn`, `zh`, `fr`, `de`, `hi`, `id`, `it`, `ja`, `ko`, `pt-BR`, `pt-PT`, `ru`, `es-MX`, `es-ES`, `tr`, `vi`, or `auto` |
| Faster or slower | `speed` 0.7–1.5 |
| “$5”, “Dr.”, “3/4” spoken as words | `text_normalization: true` |
| Brand names, acronyms, jargon | `replace: { "Acme Mobile": "Acme Mobull", "nginx": "/ˈɛndʒɪn ˈɛks/" }`; ≤200 entries, keys ≤100 chars (letters, digits, apostrophes, spaces), values ≤128; whole-word, case-insensitive, longest match wins. Streaming: `session.update { replace }` before the first `text.delta` |
| Expressive delivery | Inline `[pause]`, `[long-pause]`, `[laugh]`, `[chuckle]`, `[giggle]`, `[cry]`, `[sigh]`, `[breath]`, `[inhale]`, `[exhale]`, `[tsk]`, `[tongue-click]`, `[lip-smack]`, `[hum-tune]`. Wrapping `<whisper>`, `<soft>`, `<loud>`, `<emphasis>`, `<build-intensity>`, `<decrease-intensity>`, `<slow>`, `<fast>`, `<higher-pitch>`, `<lower-pitch>`, `<singing>`, `<sing-song>` around whole phrases |
| Captions, karaoke, lip-sync | `with_timestamps: true` → JSON `{ audio (base64), content_type, duration, audio_timestamps: { graph_chars[], graph_times[][start,end] } }`; step through `graph_chars` in order, never slice input by index |
| First audio sooner (streaming) | `optimize_streaming_latency=1` (docs also list `2`; API reference lists `0`/`1`) |
| Telephony / IVR | `output_format: { codec: "mulaw" \| "alaw", sample_rate: 8000 }`; not playable in browsers |
| Editing, post-production | `codec: "wav"`, `sample_rate: 44100` or `48000` |
| Smaller files | `codec: "mp3"`, `bit_rate: 64000` |

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

```python
import os, requests
r = requests.post(
    "https://api.x.ai/v1/tts",
    headers={"Authorization": f"Bearer {os.environ['XAI_API_KEY']}"},
    json={"text": text, "voice_id": "eve", "language": "en"},
)
r.raise_for_status(); audio_bytes = r.content  # audio/mpeg
# streaming: websockets.connect(url, additional_headers={"Authorization": f"Bearer {key}"}); send {"type":"text.delta",...}, {"type":"text.done"}; read audio.delta / audio.done
```

7. **Smoke**
   - Batch: `curl -X POST https://api.x.ai/v1/tts -H "Authorization: Bearer $XAI_API_KEY" -H "Content-Type: application/json" -d '{"text":"Hello from read aloud.","voice_id":"eve","language":"en"}' --output /tmp/hello.mp3` → 200 `audio/mpeg` (MP3, 24 kHz, 128 kbps, mono), plays. Omit `language` → 422 (observed; the docs’ table only lists 400). `voice_id: "nope"` → 404.
   - In the app: no speaker while a reply streams; it appears when the reply ends. Click → spinner → stop square → back to speaker when audio ends; click again mid-play → stops at once; click a second reply while the first plays → first stops, second plays. Safari: first play works from the click, `URL.revokeObjectURL` fires on `ended`.
   - Text prep: a reply with a fenced block, a table, a `[1]` citation and a stray `[laugh]` → spoken as “Code block omitted”, one sentence per row, the citation intact, the tag gone. Unit-test this; it is pure.
   - Streaming: send a two-sentence reply token by token; audio should start before `text.done`. Send `text.clear` mid-utterance → `audio.clear`, playback stops with nothing stale. Second utterance on the same socket → fresh `audio.delta`s, no bleed from the first.
   - Search the client bundle for `XAI_API_KEY`; it must not be there. Against a running dev server, fetch `/`, collect the `/_next/static/chunks/*.js` (or equivalent) URLs it references, and grep each; do not rely on a production build you have not made.
   - Debug from logs with `/debug-voice`; swap its hook points to `audio.delta` (byte counts), `audio.done`, `audio.clear`, `error`.

## Out of scope

- A voice agent that listens and answers (`/add-voice`), speech to text (`/add-dictation`)
- Voice cloning beyond passing an existing custom `voice_id`
- Inventing a TTS token flow, a `stream` flag on `POST /v1/tts`, or event names not in the docs