--- language: - de thumbnail: null pipeline_tag: automatic-speech-recognition tags: - whisper - pytorch - speechbrain - Transformer - hf-asr-leaderboard license: apache-2.0 datasets: - RescueSpeech metrics: - wer - cer model-index: - name: rescuespeech_whisper results: - task: name: Automatic Speech Recognition type: automatic-speech-recognition dataset: name: RescueSpeech config: de split: test args: language: de metrics: - name: Test WER type: wer value: '23.14' ---

# Whisper large-v2 fine-tuned on RescueSpeech dataset. This repository provides all the necessary tools to perform automatic speech recognition from an end-to-end whisper model fine-tuned on the RescueSpeech dataset within SpeechBrain. For a better experience, we encourage you to learn more about [SpeechBrain](https://speechbrain.github.io). The performance of the model is the following: | Release | Test CER | Test WER | GPUs | |:-------------:|:--------------:|:--------------:| :--------:| | 01-07-23 | 10.82 | 23.14 | 1xA100 80 GB | ## Pipeline description This ASR system is composed of whisper encoder-decoder blocks: - The pretrained whisper-large-v2 encoder is frozen. - The pretrained Whisper tokenizer is used. - A pretrained Whisper-large-v2 decoder ([openai/whisper-large-v2](https://huggingface.co/openai/whisper-large-v2)) is finetuned on RescueSpeech dataset. The obtained final acoustic representation is given to the greedy decoder. The system is trained with recordings sampled at 16kHz (single channel). The code will automatically normalize your audio (i.e., resampling + mono channel selection) when calling *transcribe_file* if needed. ## Install SpeechBrain First of all, please install tranformers and SpeechBrain with the following command: ``` pip install speechbrain transformers==4.26.0 ``` Please notice that we encourage you to read our tutorials and learn more about [SpeechBrain](https://speechbrain.github.io). ### Transcribing your own audio files (in French) ```python from speechbrain.pretrained import WhisperASR asr_model = WhisperASR.from_hparams(source="speechbrain/rescuespeech_whisper", savedir="pretrained_models/rescuespeech_whisper") asr_model.transcribe_file("speechbrain/rescuespeech_whisper/example_de.wav") ``` ### Inference on GPU To perform inference on the GPU, add `run_opts={"device":"cuda"}` when calling the `from_hparams` method. You can find our training results (models, logs, etc) [here](https://www.dropbox.com/sh/45wk44h8e0wkc5f/AABjEJJJ_OJp2fDYz3zEihmPa?dl=0). ### Limitations The SpeechBrain team does not provide any warranty on the performance achieved by this model when used on other datasets. #### Referencing SpeechBrain ``` @misc{SB2021, author = {Ravanelli, Mirco and Parcollet, Titouan and Rouhe, Aku and Plantinga, Peter and Rastorgueva, Elena and Lugosch, Loren and Dawalatabad, Nauman and Ju-Chieh, Chou and Heba, Abdel and Grondin, Francois and Aris, William and Liao, Chien-Feng and Cornell, Samuele and Yeh, Sung-Lin and Na, Hwidong and Gao, Yan and Fu, Szu-Wei and Subakan, Cem and De Mori, Renato and Bengio, Yoshua }, title = {SpeechBrain}, year = {2021}, publisher = {GitHub}, journal = {GitHub repository}, howpublished = {\\\\url{https://github.com/speechbrain/speechbrain}}, } ``` ### Referencing RescueSpeech ```bibtex @misc{sagar2023rescuespeech, title={RescueSpeech: A German Corpus for Speech Recognition in Search and Rescue Domain}, author={Sangeet Sagar and Mirco Ravanelli and Bernd Kiefer and Ivana Kruijff Korbayova and Josef van Genabith}, year={2023}, eprint={2306.04054}, archivePrefix={arXiv}, primaryClass={eess.AS} } ``` #### About SpeechBrain SpeechBrain is an open-source and all-in-one speech toolkit. It is designed to be simple, extremely flexible, and user-friendly. Competitive or state-of-the-art performance is obtained in various domains. Website: https://speechbrain.github.io/ GitHub: https://github.com/speechbrain/speechbrain