AI Voices for Live Narration & Streaming
Generation runs at 3.3 times real time, so audio renders faster than it plays and the buffer stays ahead. First audio arrives in about 200 milliseconds over SSE or WebSocket.

VOICES FOR THIS USE CASE
Generation outruns playback
Synthesis runs at 3.3 times real time, so audio renders faster than it plays and the buffer stays ahead of the listener. First audio arrives in around 200 milliseconds. For live commentary that margin is what makes continuous delivery possible rather than merely fast.
Planning for a dropped connection
A socket closing mid stream leaves playback truncated, and that decision belongs in your application. Retry, fall back to a cached holding clip, or fail silently are the options. For live narration a short pre-rendered holding line is usually better than dead air while a retry completes.
Transport decides the experience
The synchronous endpoint waits for a complete clip, which is fine for files and wrong for live work. Use SSE for one directional streaming, or WebSocket where text arrives incrementally and playback should begin mid sentence. On live material the pause before audio is the thing listeners notice.
Writing for a 250 character window
Requests cap at 250 characters, so continuous narration is a sequence of short calls rather than one long stream. That suits live commentary, which is naturally clause by clause, but it does mean your text pipeline needs to segment sensibly rather than cutting mid phrase.
SPECIFICATION
Sample rate
44.1 kHz native, resampled to 8 kHz for telephony
Latency
~200 ms to first byte (p50, warm region)
Output formats
ulaw, alaw, PCM 16-bit, WAV, MP3
Streaming transports
WebSocket, HTTP chunked transfer
Speed range
0.5× – 2.0×, set per request
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AI Voices for Live Narration & Streaming
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