đ Wav2Vec2-Large-960h-Lv60
A large model pretrained and fine - tuned on 960 hours of Libri - Light and Librispeech for speech recognition.
This large model is pretrained and fine - tuned on 960 hours of Libri - Light and Librispeech using 16kHz sampled speech audio. When using the model, ensure that your speech input is also sampled at 16kHz.
Facebook's Wav2Vec2
Paper
Authors: Alexei Baevski, Henry Zhou, Abdelrahman Mohamed, Michael Auli
Abstract
We show for the first time that learning powerful representations from speech audio alone followed by fine - tuning on transcribed speech can outperform the best semi - supervised methods while being conceptually simpler. wav2vec 2.0 masks the speech input in the latent space and solves a contrastive task defined over a quantization of the latent representations which are jointly learned. Experiments using all labeled data of Librispeech achieve 1.8/3.3 WER on the clean/other test sets. When lowering the amount of labeled data to one hour, wav2vec 2.0 outperforms the previous state of the art on the 100 hour subset while using 100 times less labeled data. Using just ten minutes of labeled data and pre - training on 53k hours of unlabeled data still achieves 4.8/8.2 WER. This demonstrates the feasibility of speech recognition with limited amounts of labeled data.
The original model can be found under https://github.com/pytorch/fairseq/tree/master/examples/wav2vec#wav2vec-20.
đ Quick Start
To get started with this model, you can follow the usage instructions below.
⨠Features
- Trained on 960 hours of Libri - Light and Librispeech data.
- Can be used as a standalone acoustic model for audio transcription.
- Demonstrates high performance in speech recognition tasks with limited labeled data.
đĻ Installation
No specific installation steps are provided in the original document.
đģ Usage Examples
Basic Usage
To transcribe audio files, the model can be used as a standalone acoustic model as follows:
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
from datasets import load_dataset
import torch
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60")
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60")
ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values
logits = model(input_values).logits
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)
Advanced Usage
This code snippet shows how to evaluate facebook/wav2vec2-large-960h-lv60 on LibriSpeech's "clean" and "other" test data.
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import torch
from jiwer import wer
librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60").to("cuda")
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-large-960h-lv60")
def map_to_pred(batch):
inputs = processor(batch["audio"]["array"], return_tensors="pt", padding="longest")
input_values = inputs.input_values.to("cuda")
attention_mask = inputs.attention_mask.to("cuda")
with torch.no_grad():
logits = model(input_values, attention_mask=attention_mask).logits
predicted_ids = torch.argmax(logits, dim=-1)
transcription = processor.batch_decode(predicted_ids)
batch["transcription"] = transcription
return batch
result = librispeech_eval.map(map_to_pred, batched=True, batch_size=16, remove_columns=["speech"])
print("WER:", wer(result["text"], result["transcription"]))
Result (WER):
đ Documentation
The abstract section provides a detailed explanation of the model's approach and performance.
đ License
This model is licensed under the apache - 2.0 license.
đ Information Table
Property |
Details |
Model Type |
Wav2Vec2 - Large - 960h - Lv60 |
Training Data |
960 hours of Libri - Light and Librispeech on 16kHz sampled speech audio |
Datasets |
librispeech_asr |
Tags |
speech |
License |
apache - 2.0 |