Audio / zai-org

GLM ASR Nano 2512

GLM ASR Nano 2512 is an open-source automatic speech recognition model from zai-org with MIT license and 1.5B parameters. It supports English, Chinese, and Cantonese/dialects, with claimed robustness for low-volume speech and benchmark performance compared to Whisper V3 and other open-source models. It is intended for use with transformers and planned support for vLLM and SGLang.

EstimatedRepository opened Dec 9, 2025Source checked 9/25/2026Version: 61ba4e0b
LOCALRENTED GPUOPEN WEIGHTS
Freshness 27 out of 100. Repository age: 340 days old, Dec 9, 2025. Measures age, not quality.
Popularity 59 out of 100. 19,644 downloads in 30 days; 389 lifetime likes. Checked Nov 13, 2026. Measures adoption, not quality.
What do these scores mean?

Two ways to compare models. Neither score measures answer quality, benchmark performance, or whether a model fits your machine.

Freshness · release age

A new release starts at 100. The score halves every 180 days: 100 × 2^(−age in days / 180). When the release date is unavailable, a repository date is a fallback—not proof of when the model was released. Missing dates stay unscored.

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Popularity · adoption

Downloads over the past 30 days carry 80% of the score; lifetime likes carry 20%. Both use a logarithmic scale, capped at 10 million downloads and 100,000 likes. If likes are unavailable, downloads carry all the weight. Missing download statistics stay unscored.

round(80 × min(1, log10(1 + downloads) / 7) + 20 × min(1, log10(1 + likes) / 5))

Check the statistics source
Scores as of Nov 14, 2026. Statistics checked Nov 13, 2026. Statistics are marked for refresh after 72 hours.Full scoring method

At a glance

Parameters
2.26B
Architecture
glmasr
License
mit
Disk space
4.2 GB
Software
Transformers, Transformers
View the model source
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Best for

  • Transcribing English or Mandarin audio
  • Transcribing Cantonese or dialectal speech
  • Recognizing very quiet or whispered speech
  • Processing speech in noisy meeting-like environments
  • Batched automatic speech recognition with transformers

Tradeoffs

  • Only English, Chinese, and Cantonese/dialect language support are stated; other languages are not confirmed
  • Does not include full deployment or framework support beyond transformers in the source
  • Performance claims are source-provided and not independently verified here
  • No information about hardware requirements, speed, or compatibility beyond the model card
  • No license or usage guidance beyond the stated MIT license

Ways to run it

Transformers · See official guide

Repository-specific command in publisher documentation; confirm dependencies and hardware in the source.

from transformers import AutoModelForSeq2SeqLM, AutoProcessor processor = AutoProcessor.from_pretrained("zai-org/GLM-ASR-Nano-2512") model = AutoModelForSeq2SeqLM.from_pretrained("zai-org/GLM-ASR-Nano-2512", dtype="auto", device_map="auto") inputs = processor.apply_transcription_request("https://huggingface.co/datasets/hf-internal-testing/dummy-audio-samples/resolve/main/bcn_weather.mp3") inputs = inputs.to(model.device, dtype=model.dtype) outputs = model.generate(**inputs, do_sample=False, max_new_tokens=500) decoded_outputs = processor.batch_decode(outputs[:, inputs.input_ids.shape[1] :], skip_special_tokens=True) print(decoded_outputs)
Official instructions

Transformers · See official guide

Repository-specific command in publisher documentation; confirm dependencies and hardware in the source.

from transformers import AutoModel, AutoProcessor import torch device = "cuda" if torch.cuda.is_available() else "cpu" repo_id = "zai-org/GLM-ASR-Nano-2512" processor = AutoProcessor.from_pretrained(repo_id) model = AutoModel.from_pretrained(repo_id, dtype=torch.bfloat16, device_map=device) messages = [ { "role": "user", "content": [ { "type": "audio", "url": "example_zh.wav", }, {"type": "text", "text": "Please transcribe this audio into text"}, ], } ] inputs = processor.apply_chat_template( messages, tokenize=True, add_generation_prompt=True, return_dict=True, return_tensors="pt" ) inputs = inputs.to(device, dtype=torch.bfloat16) outputs = model.generate(**inputs, max_new_tokens=128, do_sample=False) print(processor.batch_decode(outputs[:, inputs.input_ids.shape[1]:], skip_special_tokens=True))
Official instructions

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