Automatic Speech Recognition
Transformers
PyTorch
TensorBoard
Safetensors
Bengali
whisper
audio
hf-asr-leaderboard
Eval Results (legacy)
Instructions to use arijitx/whisper-small-bn with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use arijitx/whisper-small-bn with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="arijitx/whisper-small-bn")# Load model directly from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq processor = AutoProcessor.from_pretrained("arijitx/whisper-small-bn") model = AutoModelForSpeechSeq2Seq.from_pretrained("arijitx/whisper-small-bn") - Notebooks
- Google Colab
- Kaggle
| language: | |
| - bn | |
| tags: | |
| - audio | |
| - automatic-speech-recognition | |
| - hf-asr-leaderboard | |
| pipeline_tag: automatic-speech-recognition | |
| license: apache-2.0 | |
| model-index: | |
| - name: whisper-small-bn | |
| results: | |
| - task: | |
| type: automatic-speech-recognition | |
| name: Automatic Speech Recognition | |
| dataset: | |
| name: Common Voice 11.0 | |
| type: mozilla-foundation/common_voice_11_0 | |
| config: bn | |
| split: test | |
| args: | |
| language: bn | |
| metrics: | |
| - type: wer | |
| value: 35.14 | |
| name: Test WER | |
| # Whisper | |
| Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation. Trained on 680k hours | |
| of labelled data, Whisper models demonstrate a strong ability to generalise to many datasets and domains **without** the need | |
| for fine-tuning. | |
| Whisper was proposed in the paper [Robust Speech Recognition via Large-Scale Weak Supervision](https://arxiv.org/abs/2212.04356) | |
| by Alec Radford et al from OpenAI. The original code repository can be found [here](https://github.com/openai/whisper). | |
| # Usage | |
| To transcribe audio samples, the model has to be used alongside a [`WhisperProcessor`](https://huggingface.co/docs/transformers/model_doc/whisper#transformers.WhisperProcessor). | |
| The `WhisperProcessor` is used to: | |
| 1. Pre-process the audio inputs (converting them to log-Mel spectrograms for the model) | |
| 2. Post-process the model outputs (converting them from tokens to text) | |
| The model is informed of which task to perform (transcription or translation) by passing the appropriate "context tokens". These context tokens | |
| are a sequence of tokens that are given to the decoder at the start of the decoding process, and take the following order: | |
| 1. The transcription always starts with the `<|startoftranscript|>` token | |
| 2. The second token is the language token (e.g. `<|en|>` for English) | |
| 3. The third token is the "task token". It can take one of two values: `<|transcribe|>` for speech recognition or `<|translate|>` for speech translation | |
| 4. In addition, a `<|notimestamps|>` token is added if the model should not include timestamp prediction | |
| Thus, a typical sequence of context tokens might look as follows: | |
| ``` | |
| <|startoftranscript|> <|en|> <|transcribe|> <|notimestamps|> | |
| ``` | |
| Which tells the model to decode in English, under the task of speech recognition, and not to predict timestamps. | |
| These tokens can either be forced or un-forced. If they are forced, the model is made to predict each token at | |
| each position. This allows one to control the output language and task for the Whisper model. If they are un-forced, | |
| the Whisper model will automatically predict the output langauge and task itself. | |
| The context tokens can be set accordingly: | |
| ```python | |
| model.config.forced_decoder_ids = WhisperProcessor.get_decoder_prompt_ids(language="english", task="transcribe") | |
| ``` | |
| Which forces the model to predict in English under the task of speech recognition. | |
| ## Training Data | |
| Common Voice 11.0 Bengali Train | |
| OpenSLR 53 Bengali Train | |
| ### BibTeX entry and citation info | |
| ```bibtex | |
| @misc{radford2022whisper, | |
| doi = {10.48550/ARXIV.2212.04356}, | |
| url = {https://arxiv.org/abs/2212.04356}, | |
| author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya}, | |
| title = {Robust Speech Recognition via Large-Scale Weak Supervision}, | |
| publisher = {arXiv}, | |
| year = {2022}, | |
| copyright = {arXiv.org perpetual, non-exclusive license} | |
| } | |
| ``` | |