README.md

---
language:
- en
- zh
- de
- es
- ru
- ko
- fr
- ja
- pt
- tr
- pl
- ca
- nl
- ar
- sv
- it
- id
- hi
- fi
- vi
- he
- uk
- el
- ms
- cs
- ro
- da
- hu
- ta
- 'no'
- th
- ur
- hr
- bg
- lt
- la
- mi
- ml
- cy
- sk
- te
- fa
- lv
- bn
- sr
- az
- sl
- kn
- et
- mk
- br
- eu
- is
- hy
- ne
- mn
- bs
- kk
- sq
- sw
- gl
- mr
- pa
- si
- km
- sn
- yo
- so
- af
- oc
- ka
- be
- tg
- sd
- gu
- am
- yi
- lo
- uz
- fo
- ht
- ps
- tk
- nn
- mt
- sa
- lb
- my
- bo
- tl
- mg
- as
- tt
- haw
- ln
- ha
- ba
- jw
- su
tags:
- audio
- automatic-speech-recognition
- hf-asr-leaderboard
widget:
- example_title: Librispeech sample 1
  src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
- example_title: Librispeech sample 2
  src: https://cdn-media.huggingface.co/speech_samples/sample2.flac
pipeline_tag: automatic-speech-recognition
license: apache-2.0
datasets:
- ivrit-ai/whisper-training
---

# Whisper

Whisper is a pre-trained model for automatic speech recognition (ASR) and speech translation.
More details about it are available [here](https://huggingface.co/openai/whisper-large-v2).

**whisper-v2-d3-e3** is a version of whisper-large-v2, fine-tuned by [ivrit.ai](https://www.ivrit.ai) to improve Hebrew ASR using crowd-sourced labeling.

## Model details

This model comes as a single checkpoint, whisper-v2-d3-e3.
It is a 1550M parameters multi-lingual ASR solution.

# Usage

To transcribe audio samples, the model has to be used alongside a [`WhisperProcessor`](https://huggingface.co/docs/transformers/model_doc/whisper#transformers.WhisperProcessor).

```python
import torch
from transformers import WhisperProcessor, WhisperForConditionalGeneration

SAMPLING_RATE = 16000

has_cuda = torch.cuda.is_available()
model_path = 'ivrit-ai/whisper-v2-d3-e3'

model = WhisperForConditionalGeneration.from_pretrained(model_path)
if has_cuda:
    model.to('cuda:0')

processor = WhisperProcessor.from_pretrained(model_path)

# audio_resample based on entry being part of an existing dataset.
# Alternatively, this can be loaded from an audio file.
audio_resample = librosa.resample(entry['audio']['array'], orig_sr=entry['audio']['sampling_rate'], target_sr=SAMPLING_RATE)

input_features = processor(audio_resample, sampling_rate=SAMPLING_RATE, return_tensors="pt").input_features
if has_cuda:
  input_features = input_features.to('cuda:0')

predicted_ids = model.generate(input_features, language='he', num_beams=5)
transcript = processor.batch_decode(predicted_ids, skip_special_tokens=True)

print(f'Transcript: {transcription[0]}')
```

## Evaluation

You can use the [evaluate_model.py](https://github.com/yairl/ivrit.ai/blob/master/evaluate_model.py) reference on GitHub to evalute the model's quality.

## Long-Form Transcription

The Whisper model is intrinsically designed to work on audio samples of up to 30s in duration. However, by using a chunking 
algorithm, it can be used to transcribe audio samples of up to arbitrary length. This is possible through Transformers 
[`pipeline`](https://huggingface.co/docs/transformers/main_classes/pipelines#transformers.AutomaticSpeechRecognitionPipeline) 
method. Chunking is enabled by setting `chunk_length_s=30` when instantiating the pipeline. With chunking enabled, the pipeline 
can be run with batched inference. It can also be extended to predict sequence level timestamps by passing `return_timestamps=True`:

```python
>>> import torch
>>> from transformers import pipeline
>>> from datasets import load_dataset

>>> device = "cuda:0" if torch.cuda.is_available() else "cpu"

>>> pipe = pipeline(
>>>   "automatic-speech-recognition",
>>>   model="ivrit-ai/whisper-v2-d3-e3",
>>>   chunk_length_s=30,
>>>   device=device,
>>> )

>>> ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
>>> sample = ds[0]["audio"]

>>> prediction = pipe(sample.copy(), batch_size=8)["text"]
" Mr. Quilter is the apostle of the middle classes, and we are glad to welcome his gospel."

>>> # we can also return timestamps for the predictions
>>> prediction = pipe(sample.copy(), batch_size=8, return_timestamps=True)["chunks"]
[{'text': ' Mr. Quilter is the apostle of the middle classes and we are glad to welcome his gospel.',
  'timestamp': (0.0, 5.44)}]
```

Refer to the blog post [ASR Chunking](https://huggingface.co/blog/asr-chunking) for more details on the chunking algorithm.



### BibTeX entry and citation info

**ivrit.ai: A Comprehensive Dataset of Hebrew Speech for AI Research and Development**
```bibtex
@misc{marmor2023ivritai,
      title={ivrit.ai: A Comprehensive Dataset of Hebrew Speech for AI Research and Development}, 
      author={Yanir Marmor and Kinneret Misgav and Yair Lifshitz},
      year={2023},
      eprint={2307.08720},
      archivePrefix={arXiv},
      primaryClass={eess.AS}
}
```

**Whisper: Robust Speech Recognition via Large-Scale Weak Supervision**
```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}
}
```