BetaOur fastest Text-to-Speech model yet ・ Sub-50ms latency

Text-to-speech

Text-to-Speech under 50ms
that nails the hard parts

Your agent starts talking before you finish blinking. Emails, phone numbers and acronyms come out right, in all languages, over a single streaming API.

Enter your text ....or

No credit card required

50 ms to first audio, and none of it is silence

Coval measures when a caller actually hears sound: the first chunk, plus any silence at the start of the stream.

Perceived TTFA · Coval benchmark

TTFA P50

50 ms

Leading silence

0 ms

P25–P75 spread

5 ms

No marketing spin,just code and hard numbers

Paste one of these prompts into coding agents, it'll install the SDK and open the stream.

Add Gradium streaming text-to-speech to this project.

Read https://docs.gradium.ai/guides/text-to-speech first and follow it over
anything you assume. Use the official Gradium SDK for this codebase's
language. Read the API key from an environment variable, never hardcode it.

Then implement the flow, in this order:

1. Open a tts_realtime session with a voice ID and an output format, as an
   async context manager. Do not open a new socket per utterance.
2. Run the sender and the receiver as concurrent tasks. The sender pushes
   text as it is produced and calls send_eos when the source is done.
3. In the receiver, play or buffer audio messages as they arrive and stop on
   end_of_stream. Do not wait for the full text before starting playback.
4. Split outgoing text on whitespace only. Never split a word across two
   send_text calls and never send punctuation on its own, because the server
   inserts a space between consecutive messages.

Keep it small: one module, typed, with clear errors when the socket drops
mid-stream, when the voice ID is unknown, or when the key is missing.
Do not add a UI unless I ask for one.

Concurrent send/receive for voice agents and LLM streaming.

No distortion under load

Latency stays flat as concurrency climbs, flexible deployment that fits your setup.

Real-time API

Bidirectional WebSocket streaming. Start sending before ready returns; first audio in ~200 ms. Telephony formats built in.

in · pcm 24 kHz

CLIENT/AGENT

WEBSOCKET

Gradium API

wss://api.gradium.ai/
api/speech/s2s

BidirectionalTelephony ulaw / alaw~200 ms

out · pcm 48 kHz

VOICE ENGINE

Predictable & secure

Flat latency under load, enterprise SLAs, zero data retention, ISO 27001 certified.

1 conversationQuiet hour

I’d like to move my appointment.

50ms

Of course. Which day works?

10 000 conversationsPeak

I’d like to move my appointment.

50ms

Of course. Which day works?

Runs where you need it

Cloud API, dedicated instances, self-hosted, on-prem for data sovereignty, and cloud-provider marketplaces.

Predictableand scalable pricing

Native fluencyacross languages

Every Gradium model speaks every language we support, with the same quality. Regional variations and accents included, with seamless mid-sentence code-switching without latency.

Questions ?We’re here to help

Startbuilding

No credit card required