Model reference · open weights

MOSS-Audio-Tokenizer-Nano

Embeddings OpenMOSS-Team Embeddings 1 build Open weights 167k dl/mo

MOSS-Audio-Tokenizer-Nano is an open-weight embedding model from OpenMOSS-Team. MOSS-Audio-Tokenizer-Nano (FP32) weighs 44 MB; the smallest configuration that runs it is RTX 3060 12 GB.

For self-hosting, compare this build's requirements below with the GPU you plan to rent.

  • MOSS-Audio-Tokenizer-Nano is a lightweight discrete audio tokenizer developed by OpenMOSS-Team for feature extraction and use as the backbone for MOSS-TTS-Nano.
  • The model contains approximately 22M parameters and supports native 48 kHz stereo audio with a 12.5 Hz token stream.
  • It is released under the Apache-2.0 license.

Summary of the OpenMOSS-Team/MOSS-Audio-Tokenizer-Nano model card, 2026-10-01

What it is

Released byOpenMOSS-Team
Released2026-04-02
Parameters22M
VRAM44 MB for the weights

What it runs on

Memory and cards for MOSS-Audio-Tokenizer-Nano (FP32)

44 MBweights, file size
1.1 GBruntime overhead
CardRunsMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

From the model card

What OpenMOSS-Team says about MOSS-Audio-Tokenizer-Nano

Read the model card

This repository contains the Hugging Face remote-code implementation and weights for MOSS-Audio-Tokenizer-Nano, the lightweight audio tokenizer used by MOSS-TTS-Nano.

MOSS-Audio-Tokenizer-Nano is a compact discrete audio tokenizer based on the Cat (Causal Audio Tokenizer with Transformer) architecture from MOSS-Audio-Tokenizer: Scaling Audio Tokenizers for Future Audio Foundation Models. The checkpoint in this repository has 21,969,664 parameters (approximately 22M), making it much smaller than the full-size MOSS-Audio-Tokenizer while preserving the 48 kHz stereo tokenizer interface used by the MOSS-TTS family.

Key Features

  • Small model size: approximately 22M parameters, including about 10.45M encoder parameters, 10.45M decoder parameters, and 1.07M quantizer parameters.
  • Native high-resolution audio: supports 48 kHz input and output with 2-channel stereo audio, helping reduce compression loss and improve listening quality.
  • Low-frame-rate discrete codes: compresses 48 kHz stereo audio into a 12.5 Hz token stream with a downsample rate of 7,680 samples.
  • Variable bitrate reconstruction: uses a residual quantizer stack with 16 codebooks and 1,024 entries per codebook. Each codebook contributes about 0.125 kbps, for an inference range from 0.125 kbps to 2 kbps.
  • Transformer-based tokenizer: uses causal Transformer blocks and supports low-latency streaming encode/decode.
  • MOSS-TTS family interface: designed as the audio tokenizer backbone for MOSS-TTS-Nano and compatible MOSS-TTS-family workflows.

Summary: By combining a compact causal Transformer tokenizer with native 48 kHz stereo modeling, MOSS-Audio-Tokenizer-Nano reduces the deployment cost of the MOSS audio tokenizer interface while keeping high-fidelity reconstruction for speech, general audio, and music. It provides a lightweight, low-frame-rate, and streaming-friendly discrete audio representation for MOSS-TTS-Nano and other real-time speech generation workflows.

This repository contains a lightweight remote-code implementation that mirrors the current Hugging Face Transformers transformers.models.moss_audio_tokenizer module. Load it with trust_remote_code=True when needed.

Evaluation Metrics

The table below compares the reconstruction quality of MOSS-Audio-Tokenizer-Nano with open-source audio tokenizers with no more than 120M parameters on speech, audio, and music data. MOSS-Audio-Tokenizer-Nano keeps one of the smallest model sizes in the comparison while supporting 48 kHz stereo reconstruction.

  • Speech metrics are evaluated on LibriSpeech test-clean (English) and AISHELL-2 (Chinese), reported as EN/ZH.
  • Audio metrics are evaluated on the AudioSet evaluation subset, while music metrics are evaluated on MUSDB, reported as audio/music.
  • STFT-Dist. denotes the STFT distance.
  • Higher is better for speech metrics, while lower is better for audio/music metrics (Mel-Loss, STFT-Dist.).
  • Ch. denotes the number of input/output channels supported by the audio tokenizer: ch=1 means mono audio, and ch=2 means stereo audio.
  • Nvq denotes the number of quantizers.
ModelParams (M)Sample rateCh.bpsNvqSpeech: SIM ↑ (EN/ZH)Speech: STOI ↑ (EN/ZH)Speech: PESQ-NB ↑ (EN/ZH)Speech: PESQ-WB ↑ (EN/ZH)Audio/Music: Mel-Loss ↓Audio/Music: STFT-Dist. ↓
Mimi VAE2824k1----0.75 / 0.540.91 / 0.832.92 / 2.202.30 / 1.731.35 / 1.312.70 / 2.59
DAC7744.1k186110.30 / 0.200.76 / 0.681.55 / 1.361.24 / 1.151.25 / 1.182.71 / 2.54
SpeechTokenizer12016k1100020.36 / 0.250.77 / 0.681.59 / 1.381.25 / 1.17-- / ---- / --
Mimi9624k1110080.74 / 0.590.91 / 0.852.80 / 2.242.25 / 1.781.24 / 1.192.62 / 2.49
MOSS-Audio-Tokenizer-Nano2248k275060.64 / 0.610.90 / 0.852.65 / 2.282.11 / 1.871.04 / 1.012.42 / 2.27
MOSS-Audio-Tokenizer-Nano2248k2100080.75 / 0.690.92 / 0.872.92 / 2.482.36 / 2.041.00 / 0.972.37 / 2.22
EnCodec1948k2150010.35 / 0.300.76 / 0.751.54 / 1.601.25 / 1.321.25 / 1.052.73 / 2.30
SpeechTokenizer12016k1150030.52 / 0.380.84 / 0.752.00 / 1.601.57 / 1.33-- / ---- / --
Mimi9624k11512.5110.82 / 0.670.92 / 0.883.10 / 2.502.54 / 2.001.19 / 1.142.55 / 2.42
DAC7744.1k1172320.57 / 0.470.86 / 0.802.21 / 1.851.74 / 1.491.03 / 0.992.43 / 2.26
SpeechTokenizer12016k1200040.66 / 0.500.88 / 0.802.38 / 1.791.92 / 1.49-- / ---- / --
Mimi9624k12062.5150.87 / 0.730.94 / 0.903.36 / 2.762.81 / 2.221.14 / 1.092.49 / 2.36
MOSS-Audio-Tokenizer-Nano2248k21500120.84 / 0.770.94 / 0.903.25 / 2.772.71 / 2.310.95 / 0.912.31 / 2.14
MOSS-Audio-Tokenizer-Nano2248k22000160.88 / 0.810.95 / 0.913.40 / 2.932.89 / 2.470.93 / 0.892.28 / 2.11

Usage

Quickstart

import torchaudio
from transformers import AutoModel

repo_id = "OpenMOSS-Team/MOSS-Audio-Tokenizer-Nano"
model = AutoModel.from_pretrained(repo_id, trust_remote_code=True).eval()

wav, sr = torchaudio.load("demo/demo_gt.wav")
if sr != model.sampling_rate:
    wav = torchaudio.functional.resample(wav, sr, model.sampling_rate)

# The public waveform interface expects stereo audio.
if wav.shape[0] == 1:
    wav = wav.repeat(model.config.number_channels, 1)
else:
    wav = wav[: model.config.number_channels]

wav = wav.unsqueeze(0)
en

Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.

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