Stream Speech (WebSocket)

# Live TTS WebSocket — `/waves/v1/tts/live` Real-time text-to-speech over a persistent WebSocket connection. The `model` field in the request payload selects which Lightning pool serves the synthesis. ## When to use this - **Use this** when text arrives incrementally (LLM token streams, live captioning, conversational pipelines where playback should start as soon as the first chunk is ready). - POST to `/waves/v1/tts/live` (SSE) when you have the full text up front but still want chunked playback. (Same URL, different protocol — HTTP POST gets you SSE; WSS connect gets you WebSocket.) - Use `/waves/v1/tts` (sync) when total latency doesn't matter. ## Selecting the model Pass `"model": "lightning_v3.1"` (default) or `"model": "lightning_v3.1_pro"` on each request. Concurrency and latency are identical across both. Voice catalogs differ — see the [Lightning v3.1](/waves/model-cards/text-to-speech/lightning-v-3-1) and [Lightning v3.1 Pro](/waves/model-cards/text-to-speech/lightning-v-3-1-pro) model cards for the per-model catalog. ## Migrating from `/waves/v1/lightning-v3.1/get_speech/stream` Same protocol, same payload shape — only the URL changes. Existing clients should: 1. Update the WebSocket URL to `wss://api.smallest.ai/waves/v1/tts/live`. 2. Optionally add `"model": "lightning_v3.1_pro"` to route to the Pro pool. Omitting `model` keeps the existing standard-pool behavior. Voice IDs, sample rates, auth, and the response/streaming format are unchanged, so downstream audio handling, jitter buffers, and barge-in logic stay the same.