Arcana WebSockets (deprecated)

Deprecated Arcana plain-text WebSocket reference. Cloud Arcana requests switch to Coda on August 15, 2026 at 12:00 UTC.

Python

import asyncio
import websockets
import os
from dotenv import load_dotenv

load_dotenv()
api_key = os.environ.get('RIME_API_KEY')
if not api_key:
    raise ValueError("RIME_API_KEY environment variable is not set")

FILE_PATH = "arcana_ws.wav"

class RimeClient:
    def __init__(self, speaker, api_key):
        self.url = f"wss://users-ws.rime.ai/ws?speaker={speaker}&modelId=arcana&audioFormat=wav"
        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 ",
    "test ",
    "of ",
    "the ",
    "arcana ",
    "model ",
    "using ",
    "websockets ",
    "and ",
    "python.",
    "<EOS>",
]

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

print(f"Saving audio to {FILE_PATH}")
client.save_audio(FILE_PATH)

All requests require authentication with a bearer token in the Authorization header: Authorization: Bearer YOUR_API_KEY. See API authentication for how to create a key.

Overview

Rime’s websocket implementation accepts bare text, and responds with audio bytes of the selected format. All synthesis arguments are provided as query parameters when establishing the connection.

Messages

Send

The messages your client will send to the websocket API will be bare (non-serialized) text.

This will be converted to audio via websockets

Receive

The messages your client will receive will be raw audio bytes in the audio format specified at connection time.

<FF>^@^@^@9LAME3.100^AP^@^@^@^@^@^@^@^@^T<A0>$^D>"^@^@<A0>^@^@<A8><C0><BA><9D>G^N^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@
^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@
^@^@^@^@^@^@^@^@^@^@

Commands

Use these commands to manipulate the stored text buffer.

<CLEAR>

This clears the current buffer. Used in the event of interruptions.

<FLUSH>

This forces whatever buffer exists, if any, to be synthesized, and the generated audio to be sent over.

<EOS>

This forces whatever buffer exists, if any, to be synthesized, and for the server to close the connection after sending the generated audio.

Variable parameters

speaker · string · required

Must be an arcana voice from the Rime voice catalog.

text · string · required

The text you’d like spoken. Character limit per request is 1,000 via the API and in the dashboard UI.

modelId · string · default: "mistv3"

Set this to arcana. 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.

audioFormat · string

One of mp3, ogg, mulaw, or pcm

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.

samplingRate · int · default: "24000"

The sampling rate (Hz).

• On-cloud: Accepted values: 8000, 16000, 22050, 24000, 44100, 48000, 96000. Anything above 24000 is up sampling.
• On-prem: Any value is accepted.

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

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”.

Python

import asyncio
import websockets
import os
from dotenv import load_dotenv

load_dotenv()
api_key = os.environ.get('RIME_API_KEY')
if not api_key:
    raise ValueError("RIME_API_KEY environment variable is not set")

FILE_PATH = "arcana_ws.wav"

class RimeClient:
    def __init__(self, speaker, api_key):
        self.url = f"wss://users-ws.rime.ai/ws?speaker={speaker}&modelId=arcana&audioFormat=wav"
        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 ",
    "test ",
    "of ",
    "the ",
    "arcana ",
    "model ",
    "using ",
    "websockets ",
    "and ",
    "python.",
    "<EOS>",
]

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

print(f"Saving audio to {FILE_PATH}")
client.save_audio(FILE_PATH)