--- base_model: openai/whisper-tiny library_name: transformers license: apache-2.0 pipeline_tag: automatic-speech-recognition tags: - audio - automatic-speech-recognition - whisper - hf-asr-leaderboard --- Modern automatic speech recognition (ASR) models, such as OpenAI's Whisper, rely on deep encoder-decoder architectures, and their encoders are a critical bottleneck for efficient deployment due to high computational intensity. We introduce **LiteASR**, a low-rank compression scheme for ASR encoders that significantly reduces inference costs while maintaining transcription accuracy. Our approach leverages the strong low-rank properties observed in intermediate activations: by applying principal component analysis (PCA) with a small calibration dataset, we approximate linear transformations with a chain of low-rank matrix multiplications, and further optimize self-attention to work in reduced dimensionality. Evaluation results show that our method can compress Whisper large-v3's encoder size by over 50%, matching Whisper medium's size with better transcription accuracy, thereby establishing a new Pareto frontier of accuracy and efficiency. For more technical details, see our [GitHub repository](https://github.com/efeslab/LiteASR) and [paper](https://arxiv.org/abs/2502.20583). ## Sample Usage 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](https://huggingface.co/efficient-speech). ```python 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) ``` ## Benchmark Results Following is the average word error rate (WER) evaluated on the [ESB datasets](https://huggingface.co/datasets/hf-audio/esb-datasets-test-only-sorted): | Model | Average WER (↓) | Encoder Size | Decoder Size | |-------|----------------|--------------|--------------| | [whisper-tiny](https://huggingface.co/openai/whisper-tiny) | 22.01 | 7.63M | 29.55M | | [lite-whisper-tiny-acc](https://huggingface.co/efficient-speech/lite-whisper-tiny-acc) | 22.97 | 7.41M | 29.55M | | [lite-whisper-tiny](https://huggingface.co/efficient-speech/lite-whisper-tiny) | 23.95 | 7.00M | 29.55M | | [lite-whisper-tiny-fast](https://huggingface.co/efficient-speech/lite-whisper-tiny-fast) | 27.09 | 6.48M | 29.55M | |   |   |   |   | | [whisper-base](https://huggingface.co/openai/whisper-base) | 17.67 | 19.82M | 52.00M | | [lite-whisper-base-acc](https://huggingface.co/efficient-speech/lite-whisper-base-acc) | 19.07 | 18.64M | 52.00M | | [lite-whisper-base](https://huggingface.co/efficient-speech/lite-whisper-base) | 19.71 | 17.44M | 52.00M | | [lite-whisper-base-fast](https://huggingface.co/efficient-speech/lite-whisper-base-fast) | 23.05 | 16.07M | 52.00M | |   |   |   |   | | [whisper-small](https://huggingface.co/openai/whisper-small) | 15.89 | 87.00M | 153.58M | | [lite-whisper-small-acc](https://huggingface.co/efficient-speech/lite-whisper-small-acc) | 15.37 | 76.99M | 153.58M | | [lite-whisper-small](https://huggingface.co/efficient-speech/lite-whisper-small) | 14.96 | 70.16M | 153.58M | | [lite-whisper-small-fast](https://huggingface.co/efficient-speech/lite-whisper-small-fast) | 14.92 | 63.11M | 153.58M | |   |   |   |   | | [whisper-medium](https://huggingface.co/openai/whisper-medium) | 15.12 | 305.68M | 456.64M | | [lite-whisper-medium-acc](https://huggingface.co/efficient-speech/lite-whisper-medium-acc) | 13.46 | 269.93M | 456.64M | | [lite-whisper-medium](https://huggingface.co/efficient-speech/lite-whisper-medium) | 14.50 | 239.99M | 456.64M | | [lite-whisper-medium-fast](https://huggingface.co/efficient-speech/lite-whisper-medium-fast) | 14.52 | 215.31M | 456.64M | ## 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}, } ```