Real-time Streaming

Audixa provides a WebSocket endpoint for real-time text-to-speech streaming, allowing you to play audio as it is being generated.

Compatible with wav and mp3 output formats.

Connection

Endpoint

wss://api.audixa.ai/v3/tts/stream

Authentication

Pass your API key as a query parameter:

wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_API_KEY

Protocol

1. Send Request

Once connected, send a JSON payload with your generation details:

{
  "text": "Hello, world! This is a streaming test.",  // Required
  "voice_id": "am_ethan",                               // Required
  "model": "base",                                  // Optional, default: "base"
  "speed": 1.0,                                     // Optional, default: 1.0
  "audio_format": "wav",                            // Optional: "wav" or "mp3"
  "cfg_weight": 2.5,                                // Optional: Advanced model only
  "exaggeration": 0.5                               // Optional: Advanced model only
}

2. Receive Messages

The server will send a mix of JSON control messages and binary audio data. A. Started Message (JSON) Sent when processing begins.

{
  "type": "started",
  "generation_id": "gen_abc123",
  "channel": "generation:gen_abc123"
}

B. Audio Chunks (Binary) Raw binary audio data streams immediately after the “started” message. See Audio Format below for decoding details. Append these chunks to your audio buffer or play them directly. C. Completion (JSON) Sent when the stream ends.

{
  "type": "completed",
  "audio_url": "https://cdn.audixa.ai/..." // Backup complete file URL
}

D. Error (JSON)

{
  "type": "error",
  "message": "Invalid API Key"
}

Audio Format

All binary audio chunks are streamed in a standardized format, consistent across all models (base, advanced, turbo):

Property	Value
Encoding	Raw PCM (uncompressed)
Sample Format	32-bit float (`float32`)
Sample Rate	24,000 Hz
Channels	1 (Mono)
Byte Order	Little-endian

The streamed chunks are raw PCM float32 — not WAV or MP3. Each sample is a 4-byte IEEE 754 float, typically in the range [-1.0, 1.0]. You must decode them as float32, not int16, or the audio will sound like static noise.

The completed message includes an audio_url pointing to the fully encoded file (WAV/MP3) on the CDN. You can use this as a fallback or for saving a complete copy.

Examples

Real-Time Playback

Play audio chunks as they arrive for the lowest perceived latency.

import websockets
import json
import asyncio
import numpy as np
import sounddevice as sd

# Audio format — must match server output
RATE = 24000        # 24 kHz
CHANNELS = 1        # Mono
DTYPE = 'float32'   # 32-bit float PCM

async def stream_tts():
    audio_stream = sd.RawOutputStream(
        samplerate=RATE,
        channels=CHANNELS,
        dtype=DTYPE,
    )
    audio_stream.start()

    uri = "wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY"
    try:
        async with websockets.connect(uri) as websocket:
            await websocket.send(json.dumps({
                "text": "Hello from the stream!",
                "voice_id": "am_ethan",
                "model": "base"
            }))

            while True:
                msg = await websocket.recv()
                if isinstance(msg, str):
                    data = json.loads(msg)
                    if data["type"] == "started":
                        print("Stream started!")
                    elif data["type"] == "completed":
                        print(f"Done! CDN URL: {data.get('audio_url')}")
                        break
                    elif data["type"] == "error":
                        print(f"Error: {data['message']}")
                        break
                else:
                    # Decode float32 PCM and play immediately
                    audio_stream.write(np.frombuffer(msg, dtype=DTYPE))
    finally:
        audio_stream.stop()
        audio_stream.close()

asyncio.run(stream_tts())

import WebSocket from 'ws';
import Speaker from 'speaker';

const SAMPLE_RATE = 24000;

// npm install speaker — streams raw PCM to system audio
const speaker = new Speaker({
    channels: 1,
    bitDepth: 32,
    sampleRate: SAMPLE_RATE,
    float: true,
    signed: true,
});

const ws = new WebSocket('wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY');

ws.on('open', () => {
    ws.send(JSON.stringify({
        text: 'Hello from the stream!',
        voice_id: 'am_ethan',
        model: 'base'
    }));
});

ws.on('message', (data: WebSocket.Data, isBinary: boolean) => {
    if (isBinary) {
        // Write raw float32 PCM directly to speaker
        speaker.write(data as Buffer);
    } else {
        const msg = JSON.parse(data.toString());
        if (msg.type === 'started') {
            console.log(`Stream started: ${msg.generation_id}`);
        } else if (msg.type === 'completed') {
            console.log(`Done! CDN URL: ${msg.audio_url}`);
            speaker.end();
            ws.close();
        } else if (msg.type === 'error') {
            console.error('Error:', msg.message);
            ws.close();
        }
    }
});

// Real-time playback in the browser using Web Audio API
const SAMPLE_RATE = 24000;

async function streamTTS(apiKey) {
    const ctx = new AudioContext({ sampleRate: SAMPLE_RATE });
    const ws = new WebSocket(`wss://api.audixa.ai/v3/tts/stream?api_key=${apiKey}`);

    let nextStartTime = 0;

    ws.onopen = () => {
        ws.send(JSON.stringify({
            text: 'Hello from the stream!',
            voice_id: 'am_ethan',
            model: 'base'
        }));
    };

    ws.onmessage = (event) => {
        if (typeof event.data === 'string') {
            const msg = JSON.parse(event.data);
            if (msg.type === 'started') {
                console.log('Stream started:', msg.generation_id);
                nextStartTime = ctx.currentTime;
            } else if (msg.type === 'completed') {
                console.log('Done! CDN URL:', msg.audio_url);
            } else if (msg.type === 'error') {
                console.error('Error:', msg.message);
                ws.close();
            }
        } else {
            // Decode float32 PCM and schedule seamless playback
            event.data.arrayBuffer().then((arrayBuf) => {
                const float32Data = new Float32Array(arrayBuf);
                const audioBuffer = ctx.createBuffer(1, float32Data.length, SAMPLE_RATE);
                audioBuffer.getChannelData(0).set(float32Data);

                const source = ctx.createBufferSource();
                source.buffer = audioBuffer;
                source.connect(ctx.destination);

                const startAt = Math.max(ctx.currentTime, nextStartTime);
                source.start(startAt);
                nextStartTime = startAt + audioBuffer.duration;
            });
        }
    };
}

// Must be called from a user gesture (e.g., button click)
// streamTTS('YOUR_KEY');

Save to File

Collect all audio chunks and save a complete file when the stream ends.

import websockets
import json
import asyncio
import numpy as np
import soundfile as sf

async def stream_tts():
    uri = "wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY"
    async with websockets.connect(uri) as websocket:
        await websocket.send(json.dumps({
            "text": "Hello from the stream!",
            "voice_id": "am_ethan",
            "model": "base"
        }))

        audio_chunks = []
        while True:
            msg = await websocket.recv()
            if isinstance(msg, str):
                data = json.loads(msg)
                if data["type"] == "completed":
                    print(f"Done! CDN URL: {data.get('audio_url')}")
                    break
                elif data["type"] == "error":
                    print(f"Error: {data['message']}")
                    break
            else:
                # Decode float32 PCM
                audio_chunks.append(np.frombuffer(msg, dtype=np.float32))

        # Combine chunks and save as WAV
        audio = np.concatenate(audio_chunks)
        sf.write("streamed_output.wav", audio, 24000)
        print("Saved to streamed_output.wav")

asyncio.run(stream_tts())

import WebSocket from 'ws';
import * as fs from 'fs';
import { WaveFile } from 'wavefile';

const SAMPLE_RATE = 24000;
const audioChunks: Buffer[] = [];

const ws = new WebSocket('wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY');

ws.on('open', () => {
    ws.send(JSON.stringify({
        text: 'Hello from the stream!',
        voice_id: 'am_ethan',
        model: 'base'
    }));
});

ws.on('message', (data: WebSocket.Data, isBinary: boolean) => {
    if (isBinary) {
        const buffer = data as Buffer;
        console.log(`Received ${buffer.length / 4} samples`);
        audioChunks.push(buffer);
    } else {
        const msg = JSON.parse(data.toString());
        if (msg.type === 'completed') {
            console.log(`Done! CDN URL: ${msg.audio_url}`);

            // Combine and save as WAV using wavefile
            const combined = Buffer.concat(audioChunks);
            const float32 = new Float32Array(
                combined.buffer, combined.byteOffset, combined.length / 4
            );
            const wav = new WaveFile();
            wav.fromScratch(1, SAMPLE_RATE, '32f', float32);
            fs.writeFileSync('streamed_output.wav', wav.toBuffer());
            console.log('Saved to streamed_output.wav');
            ws.close();
        } else if (msg.type === 'error') {
            console.error('Error:', msg.message);
            ws.close();
        }
    }
});

const WebSocket = require('ws');
const fs = require('fs');

const SAMPLE_RATE = 24000;
const audioChunks = [];

const ws = new WebSocket('wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY');

ws.on('open', () => {
    ws.send(JSON.stringify({
        text: 'Hello from the stream!',
        voice_id: 'am_ethan',
        model: 'base'
    }));
});

ws.on('message', (data, isBinary) => {
    if (isBinary) {
        // Raw float32 PCM: 4 bytes per sample
        console.log(`Received ${data.length / 4} samples (${data.length} bytes)`);
        audioChunks.push(Buffer.from(data));
    } else {
        const msg = JSON.parse(data.toString());
        if (msg.type === 'completed') {
            console.log(`Done! CDN URL: ${msg.audio_url}`);

            // Save raw PCM — convert to WAV with ffmpeg:
            // ffmpeg -f f32le -ar 24000 -ac 1 -i output.pcm output.wav
            const combined = Buffer.concat(audioChunks);
            fs.writeFileSync('output.pcm', combined);
            console.log('Saved raw PCM to output.pcm');
            ws.close();
        } else if (msg.type === 'error') {
            console.error('Error:', msg.message);
            ws.close();
        }
    }
});

Other Languages & CLI

WebSocket is a bidirectional protocol — standard HTTP tools like cURL cannot be used for streaming. For CLI testing, you can use websocat, though binary audio handling is limited in terminal tools.For any language with a WebSocket client, the decoding rules are the same:

Connect to wss://api.audixa.ai/v3/tts/stream?api_key=YOUR_KEY
Send a JSON request, then listen for messages
Text messages → JSON control events (started, completed, error)
Binary messages → Raw PCM audio: float32, little-endian, 24 kHz, mono (4 bytes per sample)

WebSocket Close Codes

1003 Invalid Payload

The server received data it cannot accept (e.g., malformed JSON, missing fields, or validation errors).

1008 Policy Violation

Authentication failed (e.g., invalid or missing API key).

1011 Internal Error

An unexpected condition prevented the server from fulfilling the request (e.g., database or Redis connection failure).

Overview

Endpoints

Streaming

Connection

Authentication

Protocol

1. Send Request

2. Receive Messages

Audio Format

Examples

Real-Time Playback

Save to File

Other Languages & CLI

WebSocket Close Codes

​Connection

​Authentication

​Protocol

​1. Send Request

​2. Receive Messages

​Audio Format

​Examples

​Real-Time Playback

​Save to File

​Other Languages & CLI

​WebSocket Close Codes

Connection

Authentication

Protocol

1. Send Request

2. Receive Messages

Audio Format

Examples

Real-Time Playback

Save to File

Other Languages & CLI

WebSocket Close Codes