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docs(livekit): add word_timestamps to TTS parameters table #195

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Merged
harshitajain165 merged 4 commits into from
Jul 3, 2026
Merged

docs(livekit): add word_timestamps to TTS parameters table #195

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a9f4673
docs(livekit): add word_timestamps to TTS parameters table
harshitajain165 Jun 1, 2026
fd8c44d
docs(livekit): remove eou_timeout_ms note from Notes section
harshitajain165 Jun 3, 2026
61cdb86
fix(livekit): update STT endpoints and correct word_timestamps default
harshitajain165 Jun 3, 2026
2fe9bf3
fix(livekit): change TTS word_timestamps default to False (opt-in)
harshitajain165 Jun 3, 2026
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4 changes: 2 additions & 2 deletions fern/products/waves/pages/v4.0.0/integrations/livekit.mdx
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Expand Up @@ -90,7 +90,7 @@ stt = smallestai.STT(language="en", diarize=True)
| `diarize` | `bool` | `False` | Enable speaker diarization — each word includes a speaker ID |
| `eou_timeout_ms` | `int` | `0` | Milliseconds of silence before the server emits a final transcript. `0` disables server-side end-of-utterance detection (recommended — lets LiveKit's built-in turn detection control timing) |

The STT service connects to `wss://api.smallest.ai/waves/v1/pulse/get_text` for streaming and `https://api.smallest.ai/waves/v1/pulse/get_text` for batch. Interim and final transcripts are both supported. `START_OF_SPEECH` is inferred from the first non-empty transcript.
The STT service connects to `wss://api.smallest.ai/waves/v1/stt/live?model=pulse` for streaming and `https://api.smallest.ai/waves/v1/stt/?model=pulse` for batch. Interim and final transcripts are both supported. `START_OF_SPEECH` is inferred from the first non-empty transcript.

---

Expand Down Expand Up @@ -119,6 +119,7 @@ smallest_tts = smallestai.TTS(
| `sample_rate` | `int` | `24000` | Output audio sample rate in Hz. Supported: `8000`, `16000`, `24000`, `44100` |
| `output_format` | `str` | `"pcm"` | Output encoding for HTTP synthesis: `"pcm"`, `"mp3"`, `"wav"`, `"ulaw"`, `"alaw"`. WebSocket streaming always returns PCM. |
| `ws_url` | `str` | `wss://api.smallest.ai/waves/v1/tts/live` | WebSocket endpoint for low-latency streaming synthesis |
| `word_timestamps` | `bool` | `False` | Opt in to per-word timed transcript entries aligned to audio playback. Supported on base-queue English + Hindi voices (`meher`, `devansh`, `kartik`, `maithili`, `liam`, `avery`); other voices silently emit no word events. |

---

Expand Down Expand Up @@ -195,6 +196,5 @@ The pipeline is fully interruptible — if the user speaks while the bot is talk

## Notes

- Set `eou_timeout_ms=0` (the default) when using LiveKit's built-in turn detection. Setting it to a non-zero value adds server-side silence detection on top of LiveKit's own logic, which increases end-of-turn latency.
- Call `tts.prewarm()` during worker startup to pre-warm the WebSocket connection pool and reduce first-audio latency on the initial request.
- For issues or questions, open an issue in the [cookbook repository](https://github.com/smallest-inc/cookbook) or reach out on [Discord](https://discord.gg/9WtSXv26WE).
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