Websockets

Mist v3 plain-text WebSocket (/ws): send text, receive raw audio bytes.

Python

import asyncio
import websockets

class RimeClient:
    def __init__(self, speaker, api_key):
        self.url = f"wss://users-ws.rime.ai/ws?speaker={speaker}&modelId=mistv3&audioFormat=mp3"
        self.auth_headers = {
            "Authorization": f"Bearer {api_key}"
        }
        self.audio_data = b''

    async def send_tokens(self, websocket, message):
        for token in message:
            await websocket.send(token)

    async def handle_audio(self, websocket):
        while True:
            try:
                audio = await websocket.recv()
            except websockets.exceptions.ConnectionClosedOK:
                break
            self.audio_data += audio

    async def run(self, message):
        async with websockets.connect(self.url, additional_headers=self.auth_headers) as websocket:
            await asyncio.gather(
                self.send_tokens(websocket, message),
                self.handle_audio(websocket),
            )

    def save_audio(self, file_path):
        with open(file_path, 'wb') as f:
            f.write(self.audio_data)

message = [
    "This ",
    "is ",
    "a ",
    "sentence, ",
    "that ",
    "will ",
    "produce ",
    "audio.",
    "<EOS>",
]

client = RimeClient("cove", api_key="YOUR_API_KEY")
asyncio.run(client.run(message))

client.save_audio("output.mp3")

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.

Overview

This WebSocket accepts plain text and returns raw audio bytes in the format you set with audioFormat. You pass every synthesis parameter as a query parameter when you open the connection, and those settings apply for the life of the connection.

The API buffers input until it detects a sentence boundary, typically at ., ?, or !, so synthesis starts once it has enough text for natural prosody. The wait is most noticeable on the first sentence. After that, later text usually arrives while earlier audio is still playing, and the buffering becomes largely invisible. To change when synthesis starts, set segment.

Messages

Send

Send bare text, not JSON.

This will be converted to audio via websockets

Receive

Your client receives raw audio bytes in the audioFormat set at connection time. Append the messages in order to rebuild the audio.

Commands

Send these strings as text messages to control the buffer.

<CLEAR>

Discards buffered text that hasn’t been synthesized yet. Send it when the user interrupts. It doesn’t cancel a synthesis already in progress, so stop playback in your client and drop any audio that still arrives.

<FLUSH>

Synthesizes whatever text is buffered, without waiting for a sentence boundary, and sends the audio.

<EOS>

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

Variable parameters

speaker · string · required

Must be a voice from the Rime voice catalog.

modelId · string

Set to mistv3.

audioFormat · 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.

lang · string · default: "eng"

If provided, the language must match the language spoken by the selected speaker. Verify the pairing in the Rime voice catalog.

pauseBetweenBrackets · bool · default: "false"

When set to true, adds pauses between words enclosed in angle brackets. The number inside the brackets specifies the pause duration in milliseconds. Example: Hi. <200> I'd love to have a conversation with you. adds a 200ms pause. Learn more about custom pauses.

samplingRate · int

The sampling rate in Hz. Default: 24000. Values above 24000 are upsampled.

inlineSpeedAlpha · string

Comma-separated list of speed values applied to words in square brackets. Values < 1.0 speed up speech, > 1.0 slow it down. Example: “This is [slow] and [fast]”, use “3, 0.5” to make “slow” slower and “fast” faster.

speedAlpha · float · default: "1.0"

Adjusts the speed of speech. Higher than 1.0 is faster and lower than 1.0 is slower.

segment · 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

Python

import asyncio
import websockets

class RimeClient:
    def __init__(self, speaker, api_key):
        self.url = f"wss://users-ws.rime.ai/ws?speaker={speaker}&modelId=mistv3&audioFormat=mp3"
        self.auth_headers = {
            "Authorization": f"Bearer {api_key}"
        }
        self.audio_data = b''

    async def send_tokens(self, websocket, message):
        for token in message:
            await websocket.send(token)

    async def handle_audio(self, websocket):
        while True:
            try:
                audio = await websocket.recv()
            except websockets.exceptions.ConnectionClosedOK:
                break
            self.audio_data += audio

    async def run(self, message):
        async with websockets.connect(self.url, additional_headers=self.auth_headers) as websocket:
            await asyncio.gather(
                self.send_tokens(websocket, message),
                self.handle_audio(websocket),
            )

    def save_audio(self, file_path):
        with open(file_path, 'wb') as f:
            f.write(self.audio_data)

message = [
    "This ",
    "is ",
    "a ",
    "sentence, ",
    "that ",
    "will ",
    "produce ",
    "audio.",
    "<EOS>",
]

client = RimeClient("cove", api_key="YOUR_API_KEY")
asyncio.run(client.run(message))

client.save_audio("output.mp3")