Add Read Aloud

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

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.

tsCopy code
// 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" } });
}

tsCopy code
// 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.

tsCopy code
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.

tsCopy code
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)

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

pythonCopy code
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.deltas, 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