moss-tts-nano-speech
Expert skill for using MOSS-TTS-Nano, a 0.1B parameter multilingual real-time TTS model that runs on CPU with voice cloning support.
How do I install this agent skill?
npx skills add https://github.com/reason-machines/trending-skills --skill moss-tts-nano-speechIs this agent skill safe to install?
- Gen Agent Trust Hubpass
The skill is safe and provides instructions for deploying and using the MOSS-TTS-Nano text-to-speech model. It facilitates local inference and voice cloning through standard installation procedures and well-known model repositories.
- Socketpass
No alerts
- Snykpass
Risk: LOW · No issues
What does this agent skill do?
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
| Code | Language | Code | Language | Code | Language |
|---|---|---|---|---|---|
| zh | Chinese | en | English | de | German |
| es | Spanish | fr | French | ja | Japanese |
| it | Italian | hu | Hungarian | ko | Korean |
| ru | Russian | fa | Persian | ar | Arabic |
| pl | Polish | pt | Portuguese | cs | Czech |
| da | Danish | sv | Swedish | el | Greek |
| tr | Turkish |
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
| Flag | Alias | Description |
|---|---|---|
--prompt-audio-path | — | Path to reference WAV for voice cloning (infer.py) |
--prompt-speech | — | Same purpose in moss-tts-nano generate CLI |
--text | — | Input text string |
--text-file | — | Path to plain text file for long-form synthesis |
--output | — | Output WAV file path (default varies by entrypoint) |
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=[
gr.Audio(type="filepath", label="Reference Voice"),
gr.Textbox(label="Text to synthesize"),
],
outputs=gr.Audio(label="Generated Speech"),
title="MOSS-TTS-Nano Voice Clone",
)
demo.launch()
Troubleshooting
WeTextProcessing install fails
# Use conda to get pynini, then install from source
conda install -c conda-forge pynini=2.1.6.post1 -y
pip install git+https://github.com/WhizZest/WeTextProcessing.git
Model download is slow or fails
Set HF_ENDPOINT to a mirror if Hugging Face is unreachable:
export HF_ENDPOINT=https://hf-mirror.com
python infer.py --prompt-audio-path assets/audio/zh_1.wav --text "测试"
Or use ModelScope:
pip install modelscope
Then point model paths to openmoss/MOSS-TTS-Nano and openmoss/MOSS-Audio-Tokenizer-Nano.
Out of memory on CPU
- Use streaming inference (
infer_stream) to reduce peak memory. - Reduce chunk size for long text inputs — the model handles chunked voice cloning automatically.
- Close other applications; the model needs ~1–2 GB RAM.
Audio output is silent or corrupt
- Ensure the reference WAV is a clean mono or stereo file, 16-bit or float32, any sample rate (it will be resampled).
- Minimum reference audio duration: ~3–5 seconds for reliable voice cloning.
- Avoid reference audio with heavy background noise.
moss-tts-nano command not found
# Re-run editable install inside the active conda env
pip install -e .
which moss-tts-nano # should resolve now
Port conflict for web demo
# Default port is 18083; check what occupies it
lsof -i :18083
# Kill if needed, then relaunch
moss-tts-nano serve
Output Defaults
| Entrypoint | Default output path |
|---|---|
python infer.py | generated_audio/infer_output.wav |
moss-tts-nano generate | generated_audio/moss_tts_nano_output.wav |
python app.py / moss-tts-nano serve | returned via HTTP response |
The generated_audio/ directory is created automatically if it does not exist.
How can the creator link this skill?
Add the canonical catalog link to the repository README so users can inspect current installs and available audits. The publishing guide covers the complete discovery path.
<a href="https://skillzs.dev/skills/reason-machines/trending-skills/moss-tts-nano-speech">View moss-tts-nano-speech on skillZs</a>