Model reference · open weights

MOSS-Transcribe

Audio OpenMOSS-Team Speech→text 1 build Open weights 1k dl/mo

MOSS-Transcribe is an open-weight audio or speech model from OpenMOSS-Team. MOSS-Transcribe-preview-2B (BF16) weighs 4.8 GB; the smallest configuration that runs it is RTX 3060 12 GB.

  • MOSS-Transcribe is an English automatic speech recognition model developed by the OpenMOSS Team.
  • It features a Qwen3-1.7B-base language model backbone paired with a Qwen3-Omni-MoE audio encoder and a gated-MLP adapter.
  • The model contains approximately 2.4B parameters, is distributed as a single bfloat16 safetensors shard, and is released under the Apache-2.0 license.

Summary of the OpenMOSS-Team/MOSS-Transcribe-preview-2B model card, 2026-10-01

What it is

Released byOpenMOSS-Team
Released2026-06-26
Parameters2.4B
VRAM4.8 GB for the weights

What it runs on

Memory and cards for MOSS-Transcribe-preview-2B (BF16)

4.8 GBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
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-Transcribe

Read the model card

    

MOSS-Transcribe-preview-2B is an English speech-to-text model that pairs a Qwen3-1.7B-base language-model backbone with a Qwen3-Omni-MoE audio encoder. A gated-MLP adapter projects audio features into the language-model embedding space. The model is trained on public English ASR corpora and fine-tuned with reinforcement learning on the Open ASR Leaderboard training splits.

The model has approximately 2.4B parameters and is distributed as a single bfloat16 safetensors shard of approximately 4.84 GB.

Model Details

  • Developed by: OpenMOSS Team
  • Model type: Automatic Speech Recognition / speech-to-text model
  • Language: English
  • License: Apache-2.0
  • Library: Transformers
  • Backbone: Qwen3-1.7B-base, 28 layers, hidden size 2048
  • Audio encoder: Qwen3-Omni-MoE audio encoder
  • Adapter: Gated-MLP adapter, hidden size 8192
  • Parameter size: approximately 2.4B
  • Checkpoint format: bfloat16 safetensors

Intended Use

This model is intended for English automatic speech recognition, including transcription of English speech audio for research and evaluation purposes.

Evaluation

Evaluated on the Open ASR Leaderboard test sets. Predictions are produced with greedy decoding (num_beams=1, max_new_tokens=512), a single dataset-agnostic chat template, and scored with the leaderboard's standardized scoring (English normalizer + word-level edit distance with compound merging). TED-LIUM is not currently part of the leaderboard run and is therefore excluded.

DatasetWER (%)
AMI8.37
Earnings227.84
GigaSpeech6.78
LibriSpeech test.clean1.21
LibriSpeech test.other2.84
SPGISpeech1.63
VoxPopuli5.39
Average4.87

Inference

import librosa
import torch
from huggingface_hub import hf_hub_download
from transformers import AutoModelForCausalLM, AutoTokenizer
from transformers.dynamic_module_utils import get_class_from_dynamic_module

REPO = "OpenMOSS-Team/MOSS-Transcribe-preview-2B"
DEVICE = "cuda:0"

model = AutoModelForCausalLM.from_pretrained(
    REPO, dtype=torch.bfloat16, trust_remote_code=True
).to(DEVICE).eval()
tokenizer = AutoTokenizer.from_pretrained(REPO, trust_remote_code=True)

MossProcessor = get_class_from_dynamic_module("processing_Moss.MossProcessor", REPO)
MelConfig = get_class_from_dynamic_module("processing_Moss.MelConfig", REPO)

mel_cfg = MelConfig(
    mel_sr=16000,
    mel_dim=128,
    mel_n_fft=400,
    mel_hop_length=160,
)
processor = MossProcessor(tokenizer, config=mel_cfg, enable_time_marker=False)
processor.load_template(hf_hub_download(REPO, "chat_template_default.py"))

waveform, _ = librosa.load("your_audio.wav", sr=16000)
inputs = processor(audio=waveform, return_tensors="pt").to(DEVICE)
inputs["audio_data"] = inputs["audio_data"].to(model.dtype)

with torch.no_grad():
    out_ids = model.generate(
        **inputs,
        max_new_tokens=512,
        do_sample=False,
        num_beams=1,
        use_cache=True,
        eos_token_id=[processor.end_token_id],
    )

new_ids = out_ids[:, inputs["input_ids"].shape[1]:]
transcript = processor.batch_decode(new_ids, skip_special_tokens=True)[0].strip()
print(transcript)

Audio Frontend

  • Sample rate: 16 kHz
  • Features: Whisper log-mel filterbank
  • Mel bins: 128
  • FFT size: 400
  • Hop length: 160

Training

The model was trained on public English ASR corpora and fine-tuned with reinforcement learning on the Open ASR Leaderboard training splits.

Limitations

The model is designed for English ASR. It may perform worse on non-English speech, heavy accents, noisy recordings, overlapping speakers, far-field audio, domain-specific terminology, or audio conditions that differ significantly from the training and evaluation data. The output should be manually reviewed before use in high-stakes settings.

Citation

@misc{moss_transcribe_2025,
  title        = {{MOSS-Transcribe-preview-2B}},
  author       = {{OpenMOSS Team}},
  year         = {2025},
  howpublished = {\url{https://huggingface.co/OpenMOSS-Team/MOSS-Transcribe-preview-2B}}
}

License

This model is released under the Apache-2.0 license.

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

Benchmarks

Reported results

As published on the model card: the maker's own numbers, not measured by AxForge.

TaskDatasetMetricScore
automatic-speech-recognitionOpen ASR LeaderboardAverage WER4.870
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