Moss TTS Nano Speech

MOSS-TTS-Nano Speech Generation Skill

Skill by ara.so — Daily 2026 Skills collection.

MOSS-TTS-Nano is an open-source multilingual tiny TTS model (0.1B parameters) from MOSI.AI and the OpenMOSS team. It uses an Audio Tokenizer + LLM autoregressive pipeline to generate 48 kHz stereo speech in real time, supports 20 languages, voice cloning, streaming inference, and runs on CPU without a GPU.

Installation

Conda (recommended)

conda create -n moss-tts-nano python=3.12 -y
conda activate moss-tts-nano

git clone https://github.com/OpenMOSS/MOSS-TTS-Nano.git
cd MOSS-TTS-Nano

pip install -r requirements.txt
pip install -e .

Fix WeTextProcessing if it fails

conda install -c conda-forge pynini=2.1.6.post1 -y
pip install git+https://github.com/WhizZest/WeTextProcessing.git

After pip install -e . the moss-tts-nano CLI command is available in the active environment.

Model Weights

Models are auto-downloaded from Hugging Face on first run:

• TTS model: OpenMOSS-Team/MOSS-TTS-Nano
• Audio tokenizer: OpenMOSS-Team/MOSS-Audio-Tokenizer-Nano

ModelScope mirrors are available at openmoss/MOSS-TTS-Nano and openmoss/MOSS-Audio-Tokenizer-Nano.

CLI Commands

Generate speech (voice clone mode)

moss-tts-nano generate \
  --prompt-speech assets/audio/zh_1.wav \
  --text "欢迎关注模思智能、上海创智学院与复旦大学自然语言处理实验室。"

Output defaults to generated_audio/moss_tts_nano_output.wav.

Generate from a text file (long-form)

moss-tts-nano generate \
  --prompt-speech assets/audio/zh_1.wav \
  --text-file my_script.txt \
  --output output.wav

Launch local web demo

moss-tts-nano serve
# or directly:
python app.py

Opens at http://127.0.0.1:18083 — model stays loaded in memory for fast repeated requests.

Direct Python entrypoint

python infer.py \
  --prompt-audio-path assets/audio/zh_1.wav \
  --text "Hello, this is a test of MOSS-TTS-Nano."

Output: generated_audio/infer_output.wav

Python API Usage

Basic voice clone inference

from infer import MossTTSNanoInference

# Initialize once (downloads weights on first run)
tts = MossTTSNanoInference()

# Voice clone: synthesize text in the style of the reference audio
audio = tts.infer(
    text="欢迎使用MOSS语音合成系统。",
    prompt_audio_path="assets/audio/zh_1.wav",
)

# Save output
import soundfile as sf
sf.write("output.wav", audio, samplerate=48000)

English voice clone

from infer import MossTTSNanoInference

tts = MossTTSNanoInference()

audio = tts.infer(
    text="Welcome to MOSS TTS Nano, a tiny but capable text to speech model.",
    prompt_audio_path="assets/audio/en_sample.wav",
)

import soundfile as sf
sf.write("english_output.wav", audio, samplerate=48000)

Streaming inference (low latency)

from infer import MossTTSNanoInference
import soundfile as sf
import numpy as np

tts = MossTTSNanoInference()

chunks = []
for audio_chunk in tts.infer_stream(
    text="This sentence is generated chunk by chunk for low latency playback.",
    prompt_audio_path="assets/audio/en_sample.wav",
):
    chunks.append(audio_chunk)
    # process or play chunk in real time here

full_audio = np.concatenate(chunks)
sf.write("streamed_output.wav", full_audio, samplerate=48000)

Long-text synthesis with chunked voice cloning

from infer import MossTTSNanoInference

tts = MossTTSNanoInference()

long_text = """
MOSS-TTS-Nano supports long-form synthesis through automatic chunking.
Each chunk uses the same reference voice, producing consistent speaker identity
across the entire output even for multi-paragraph documents.
"""

audio = tts.infer(
    text=long_text,
    prompt_audio_path="assets/audio/en_sample.wav",
)

import soundfile as sf
sf.write("long_form_output.wav", audio, samplerate=48000)

FastAPI HTTP endpoint usage

When the server is running (moss-tts-nano serve or python app.py):

import requests
import base64
import soundfile as sf
import io
import numpy as np

# Read reference audio as base64
with open("assets/audio/zh_1.wav", "rb") as f:
    ref_audio_b64 = base64.b64encode(f.read()).decode()

response = requests.post(
    "http://127.0.0.1:18083/generate",
    json={
        "text": "你好，这是一个语音合成测试。",
        "prompt_audio_base64": ref_audio_b64,
    },
)

data = response.json()
audio_bytes = base64.b64decode(data["audio_base64"])

audio_array, sr = sf.read(io.BytesIO(audio_bytes))
sf.write("api_output.wav", audio_array, samplerate=sr)

Streaming HTTP response (real-time web playback)

import requests

with open("assets/audio/zh_1.wav", "rb") as f:
    ref_audio_b64 = __import__("base64").b64encode(f.read()).decode()

with requests.post(
    "http://127.0.0.1:18083/generate_stream",
    json={
        "text": "流式语音合成示例，适合实时播放场景。",
        "prompt_audio_base64": ref_audio_b64,
    },
    stream=True,
) as resp:
    with open("stream_output.wav", "wb") as out:
        for chunk in resp.iter_content(chunk_size=4096):
            out.write(chunk)

Supported Languages

The language is inferred automatically from the input text and the reference audio. No explicit language code parameter is required for basic usage.

Architecture Overview

• Pipeline: Audio Tokenizer + LLM (pure autoregressive)
• Audio Tokenizer: MOSS-Audio-Tokenizer-Nano (~20M params), CNN-free causal Transformer (Cat architecture)
• Output: 48 kHz, 2-channel (stereo)
• Token rate: 12.5 Hz token stream
• Codebooks: RVQ with 16 codebooks (0.125 kbps – 2 kbps)
• LLM: ~0.1B parameters total

Key CLI Flags

Common Patterns

Pattern: Batch synthesis with one reference voice

from infer import MossTTSNanoInference
import soundfile as sf

tts = MossTTSNanoInference()
ref = "assets/audio/zh_1.wav"

sentences = [
    "第一句话，用于批量合成测试。",
    "第二句话，保持相同的音色。",
    "第三句话，输出独立的音频文件。",
]

for i, sentence in enumerate(sentences):
    audio = tts.infer(text=sentence, prompt_audio_path=ref)
    sf.write(f"output_{i:02d}.wav", audio, samplerate=48000)
    print(f"Saved output_{i:02d}.wav")

Pattern: Real-time playback with sounddevice

import sounddevice as sd
import numpy as np
from infer import MossTTSNanoInference

tts = MossTTSNanoInference()

buffer = []
for chunk in tts.infer_stream(
    text="Real-time playback example using sounddevice.",
    prompt_audio_path="assets/audio/en_sample.wav",
):
    buffer.append(chunk)

audio = np.concatenate(buffer)
sd.play(audio, samplerate=48000)
sd.wait()

Pattern: Gradio integration

import gradio as gr
import soundfile as sf
import numpy as np
import io
from infer import MossTTSNanoInference

tts = MossTTSNanoInference()

def synthesize(reference_audio_path: str, text: str):
    audio = tts.infer(text=text, prompt_audio_path=reference_audio_path)
    # Return as (sample_rate, numpy_array) tuple for Gradio Audio component
    return (48000, audio)

demo = gr.Interface(
    fn=synthesize,
    inputs=[