Automatic Speech Recognition
Transformers
PyTorch
Safetensors
Hebrew
wav2vec2
Generated from Trainer
robust-speech-event
Instructions to use imvladikon/wav2vec2-xls-r-300m-lm-hebrew with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use imvladikon/wav2vec2-xls-r-300m-lm-hebrew with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="imvladikon/wav2vec2-xls-r-300m-lm-hebrew")# Load model directly from transformers import AutoProcessor, AutoModelForCTC processor = AutoProcessor.from_pretrained("imvladikon/wav2vec2-xls-r-300m-lm-hebrew") model = AutoModelForCTC.from_pretrained("imvladikon/wav2vec2-xls-r-300m-lm-hebrew", device_map="auto") - Notebooks
- Google Colab
- Kaggle
| language: | |
| - he | |
| license: apache-2.0 | |
| tags: | |
| - generated_from_trainer | |
| - he | |
| - robust-speech-event | |
| datasets: | |
| - imvladikon/hebrew_speech_kan | |
| - imvladikon/hebrew_speech_coursera | |
| metrics: | |
| - wer | |
| base_model: facebook/wav2vec2-xls-r-300m | |
| model-index: | |
| - name: wav2vec2-xls-r-300m-lm-hebrew | |
| results: [] | |
| # wav2vec2-xls-r-300m-lm-hebrew | |
| This model is a fine-tuned version of [facebook/wav2vec2-xls-r-300m](https://huggingface.co/facebook/wav2vec2-xls-r-300m) on the None dataset | |
| with adding ngram models according to [Boosting Wav2Vec2 with n-grams in 🤗 Transformers](https://huggingface.co/blog/wav2vec2-with-ngram) | |
| ## Usage | |
| check package: https://github.com/imvladikon/wav2vec2-hebrew | |
| or use transformers pipeline: | |
| ```python | |
| import torch | |
| from datasets import load_dataset | |
| from transformers import AutoModelForCTC, AutoProcessor | |
| import torchaudio.functional as F | |
| model_id = "imvladikon/wav2vec2-xls-r-300m-lm-hebrew" | |
| sample_iter = iter(load_dataset("google/fleurs", "he_il", split="test", streaming=True)) | |
| sample = next(sample_iter) | |
| resampled_audio = F.resample(torch.tensor(sample["audio"]["array"]), sample["audio"]["sampling_rate"], 16_000).numpy() | |
| model = AutoModelForCTC.from_pretrained(model_id) | |
| processor = AutoProcessor.from_pretrained(model_id) | |
| input_values = processor(resampled_audio, return_tensors="pt").input_values | |
| with torch.no_grad(): | |
| logits = model(input_values).logits | |
| transcription = processor.batch_decode(logits.numpy()).text | |
| print(transcription) | |
| ``` | |
| ## Intended uses & limitations | |
| More information needed | |
| ## Training and evaluation data | |
| More information needed | |
| ## Training procedure | |
| ### Training hyperparameters | |
| The following hyperparameters were used during training: | |
| - learning_rate: 0.0003 | |
| - train_batch_size: 64 | |
| - eval_batch_size: 16 | |
| - seed: 42 | |
| - gradient_accumulation_steps: 2 | |
| - total_train_batch_size: 128 | |
| - optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08 | |
| - lr_scheduler_type: linear | |
| - lr_scheduler_warmup_steps: 500 | |
| - num_epochs: 100 | |
| - mixed_precision_training: Native AMP | |
| ### Training results | |
| ### Framework versions | |
| - Transformers 4.16.0.dev0 | |
| - Pytorch 1.10.1+cu102 | |
| - Datasets 1.17.1.dev0 | |
| - Tokenizers 0.11.0 |