metadata

base_model: openai/whisper-large-v3-turbo
library_name: transformers
license: apache-2.0
pipeline_tag: automatic-speech-recognition
tags:
  - audio
  - automatic-speech-recognition
  - whisper
  - hf-asr-leaderboard

Model Card for Lite-Whisper large-v3-turbo-acc

Lite-Whisper is a compressed version of OpenAI Whisper with LiteASR. See our GitHub repository and paper for details.

Benchmark Results

Following is the average word error rate (WER) evaluated on the ESB datasets:

Model	Average WER (↓)	Encoder Size	Decoder Size
whisper-large-v3	10.1	635M	907M
lite-whisper-large-v3-acc	10.1	429M	907M
lite-whisper-large-v3	10.2	377M	907M
lite-whisper-large-v3-fast	11.3	308M	907M

whisper-large-v3-turbo	10.1	635M	172M
lite-whisper-large-v3-turbo-acc	10.2	421M	172M
lite-whisper-large-v3-turbo	12.6	374M	172M
lite-whisper-large-v3-turbo-fast	20.1	313M	172M

whisper-medium	14.8	306M	457M

Quick Start

The easiest way to run our model is to use our integration with HuggingFace Transformers library. We provide model weights for the compressed version of OpenAI Whisper series here.

import librosa 
import torch
from transformers import AutoProcessor, AutoModel

device = "cuda:0"
dtype = torch.float16

# load the compressed Whisper model
model = AutoModel.from_pretrained(
    "efficient-speech/lite-whisper-large-v3-turbo", 
    trust_remote_code=True, 
)
model.to(dtype).to(device)

# we use the same processor as the original model
processor = AutoProcessor.from_pretrained("openai/whisper-large-v3")

# set the path to your audio file
path = "path/to/audio.wav"
audio, _ = librosa.load(path, sr=16000)

input_features = processor(audio, sampling_rate=16000, return_tensors="pt").input_features
input_features = input_features.to(dtype).to(device)

predicted_ids = model.generate(input_features)
transcription = processor.batch_decode(
    predicted_ids, 
    skip_special_tokens=True
)[0]

print(transcription)

Citation

If you use LiteASR in your research, please cite the following paper:

@misc{kamahori2025liteasrefficientautomaticspeech,
      title={LiteASR: Efficient Automatic Speech Recognition with Low-Rank Approximation}, 
      author={Keisuke Kamahori and Jungo Kasai and Noriyuki Kojima and Baris Kasikci},
      year={2025},
      eprint={2502.20583},
      archivePrefix={arXiv},
      primaryClass={cs.LG},
      url={https://arxiv.org/abs/2502.20583}, 
}