efficient-speech/lite-whisper-large-v3-turbo-acc

Model

13

stars

12

commits

2

repos using this model

1

linked in READMEs

Jul 6, 2026

updated

audio
automatic-speech-recognition
custom_code
eval-results
feature-extraction
hf-asr-leaderboard
lite-whisper
safetensors
transformers
whisper
Browse cluster: Speech Recognition and Audio Processing

README

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:

ModelAverage WER (↓)Encoder SizeDecoder Size
whisper-large-v310.1635M907M
lite-whisper-large-v3-acc10.1429M907M
lite-whisper-large-v310.2377M907M
lite-whisper-large-v3-fast11.3308M907M
    
whisper-large-v3-turbo10.1635M172M
lite-whisper-large-v3-turbo-acc10.2421M172M
lite-whisper-large-v3-turbo12.6374M172M
lite-whisper-large-v3-turbo-fast20.1313M172M
    
whisper-medium14.8306M457M

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}, 
}

Contributors

kamahori

8 commits

SaylorTwift

2 commits

nielsr

1 commits

Xenova

1 commits

efficient-speech/lite-whisper-large-v3-turbo-acc

Model

13

stars

12

commits

2

repos using this model

1

linked in READMEs

Jul 6, 2026

updated

audio
automatic-speech-recognition
custom_code
eval-results
feature-extraction
hf-asr-leaderboard
lite-whisper
safetensors
transformers
whisper
Browse cluster: Speech Recognition and Audio Processing

README

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:

ModelAverage WER (↓)Encoder SizeDecoder Size
whisper-large-v310.1635M907M
lite-whisper-large-v3-acc10.1429M907M
lite-whisper-large-v310.2377M907M
lite-whisper-large-v3-fast11.3308M907M
    
whisper-large-v3-turbo10.1635M172M
lite-whisper-large-v3-turbo-acc10.2421M172M
lite-whisper-large-v3-turbo12.6374M172M
lite-whisper-large-v3-turbo-fast20.1313M172M
    
whisper-medium14.8306M457M

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}, 
}

Contributors

kamahori

8 commits

SaylorTwift

2 commits

nielsr

1 commits

Xenova

1 commits