Real-time Streaming

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Connection

wss://api.lokutor.com/ws/tts?api_key=your-api-key

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Request

{
  "text": "Hello, how can I help you today?",
  "voice": "M1",
  "language": "en",
  "speed": 1.0,
  "steps": 5,
  "visemes": false
}

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Response

• Binary messages: Raw Int16 PCM audio chunks (44.1 kHz, mono). Large chunks are split into ≤512KB WebSocket frames.
• Text message "EOS": Synthesis complete for this request — the connection stays open for further requests.
• JSON visemes: A bare JSON array of { v, c, t, id } objects (not wrapped in a {"type":...} envelope) — sent only if visemes: true and only when the model actually produced viseme data for that chunk.
• JSON timing (optional, streaming mode only): { "type": "timing", "timing": { ... RTF, IsFirst, ... }, "ttfb_ms"?: number } — internal performance telemetry sent alongside some chunks; ttfb_ms is only present on the first chunk. Treat this as informational, not part of the audio contract.
• JSON error: { "type": "error", "data": { "code": "...", "message": "...", "detail": "...", "retryable": true|false } } — same structured shape as every other Lokutor error; see the Error Catalog. This is not a bare {"type":"error","message":"..."} — the payload is under data, and it’s the full structured object.

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Best practices

• Keep the WebSocket open for multiple requests to avoid handshake latency
• The stream runs at 6 denoising steps, tuned for live conversation; for offline audio at up to 8, use REST POST /tts/synthesize
• Validate inputs against GET /config before sending
• Always pass voice and language explicitly — neither has a server-side default on this endpoint

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Languages