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The Rime API authenticates every request with a bearer token in the Authorization header: Authorization: Bearer YOUR_API_KEY. See API authentication for how to create a key.
This page describes the legacy /ws endpoint, which keeps working for existing integrations. For a new Coda integration, use Coda WebSocket v1.
Include modelId=coda in the connection query. If you omit it, the server routes the request to Mist v3 and speakers outside the Mist v3 catalog can fail with Speaker not found.

Overview

This endpoint accepts bare text and responds with raw audio bytes in the selected format. All synthesis arguments are query parameters on the connection URL.

Messages

Send

Send bare text, not JSON.

Receive

The server sends raw audio bytes in the audioFormat set on the connection. With pcm, a message can end mid-sample, so keep whole samples.

Commands

Send these strings as a whole message to act on the server’s text buffer.

<CLEAR>

Discards the buffered text. Use it when the user interrupts.

<FLUSH>

Synthesizes whatever text is buffered and sends the audio.

<EOS>

Synthesizes whatever text is buffered, sends the audio, then closes the connection.

Variable parameters

string
required
Must be a coda voice from the Rime voice catalog.
string
required
The text you’d like spoken. Character limit per request is 1,000 via the API and in the dashboard UI.
string
default:"mistv3"
Set this to coda. It is not strictly required, but if you omit it the server defaults to the Mist v3 backend, and speakers outside the Mist v3 catalog fail with a “Speaker not found” error.
string
One of wav, mp3 (or mpeg), ogg (Opus in OGG), webm (Opus in WebM), pcm (or l16), or mulaw (or pcmu). Unrecognized values fall back to wav.
string
default:"eng"
If provided, the language must match the language spoken by the provided speaker. Both the 2-letter ISO 639-1 and the 3-letter ISO 639-2/3 form are accepted:See the voices documentation for which speakers support each language.
int
default:"24000"
The sampling rate in Hz. Any positive integer is accepted, and values above the model’s native rate are upsampled. Common choices are 8000 (telephony), 16000, 22050, 24000, and 44100.
float
default:"1.0"
The time scaling factor. Accepted range is 0.4 to 2.5. Values outside this range are clamped without an error. Values above 1.0 slow the audio down, and values below 1.0 speed it up.
string
default:"bySentence"
Controls how text is segmented for synthesis. Available options:
  • “immediate” - Synthesizes text immediately without waiting for complete sentences
  • “never” - Never segments the text, waits for explicit flush or EOS
  • “bySentence” (default) - Waits for complete sentences before synthesis
Note: For backward compatibility, setting immediate=true in query params is equivalent to segment=immediate. If a null value is provided, it will default to “bySentence”.