pytorch data2vec-audio automatic-speech-recognition en dataset:librispeech_asr arxiv:2202.03555 transformers speech hf-asr-leaderboard license:apache-2.0 model-index
---
language: en
datasets:
- librispeech_asr
tags:
- speech
- hf-asr-leaderboard

license: apache-2.0
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
model-index:
- name: data2vec-audio-large-960h
  results:
  - task:
      name: Automatic Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: LibriSpeech (clean)
      type: librispeech_asr
      config: clean
      split: test
      args: 
        language: en
    metrics:
    - name: Test WER
      type: wer
      value: 1.89
  - task:
      name: Automatic Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: LibriSpeech (other)
      type: librispeech_asr
      config: other
      split: test
      args: 
        language: en
    metrics:
    - name: Test WER
      type: wer
      value: 4.07
---

# Data2Vec-Audio-Large-960h

[Facebook's Data2Vec](https://ai.facebook.com/research/data2vec-a-general-framework-for-self-supervised-learning-in-speech-vision-and-language/)

The large model pretrained and fine-tuned on 960 hours of Librispeech on 16kHz sampled speech audio. When using the model
make sure that your speech input is also sampled at 16Khz.

[Paper](https://arxiv.org/abs/2202.03555)

Authors: Alexei Baevski, Wei-Ning Hsu, Qiantong Xu, Arun Babu, Jiatao Gu, Michael Auli

**Abstract**

While the general idea of self-supervised learning is identical across modalities, the actual algorithms and objectives differ widely because they were developed with a single modality in mind. To get us closer to general self-supervised learning, we present data2vec, a framework that uses the same learning method for either speech, NLP or computer vision. The core idea is to predict latent representations of the full input data based on a masked view of the input in a self-distillation setup using a standard Transformer architecture. Instead of predicting modality-specific targets such as words, visual tokens or units of human speech which are local in nature, data2vec predicts contextualized latent representations that contain information from the entire input. Experiments on the major benchmarks of speech recognition, image classification, and natural language understanding demonstrate a new state of the art or competitive performance to predominant approaches.

The original model can be found under https://github.com/pytorch/fairseq/tree/main/examples/data2vec .

# Pre-Training method

![model image](https://raw.githubusercontent.com/patrickvonplaten/scientific_images/master/data2vec.png)

For more information, please take a look at the [official paper](https://arxiv.org/abs/2202.03555).

# Usage

To transcribe audio files the model can be used as a standalone acoustic model as follows:

```python
 from transformers import Wav2Vec2Processor, Data2VecAudioForCTC
 from datasets import load_dataset
 import torch
 
 # load model and processor
 processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-large-960h")
 model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-large-960h")
     
 # load dummy dataset and read soundfiles
 ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
 
 # tokenize
 input_values = processor(ds[0]["audio"]["array"],, return_tensors="pt", padding="longest").input_values  # Batch size 1
 
 # retrieve logits
 logits = model(input_values).logits
 
 # take argmax and decode
 predicted_ids = torch.argmax(logits, dim=-1)
 transcription = processor.batch_decode(predicted_ids)
 ```
 
  ## Evaluation
 
 This code snippet shows how to evaluate **facebook/data2vec-audio-large-960h** on LibriSpeech's "clean" and "other" test data.
 
```python
 from transformers import Wav2Vec2Processor, Data2VecAudioForCTC
 from datasets import load_dataset
 import torch
 from jiwer import wer
 
 # load model and processor
 processor = Wav2Vec2Processor.from_pretrained("facebook/data2vec-audio-large-960h").to("cuda")
 model = Data2VecAudioForCTC.from_pretrained("facebook/data2vec-audio-large-960h")
 

librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")

def map_to_pred(batch):
    input_values = processor(batch["audio"]["array"], return_tensors="pt", padding="longest").input_values
    with torch.no_grad():
        logits = model(input_values.to("cuda")).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=1, remove_columns=["audio"])

print("WER:", wer(result["text"], result["transcription"]))
```

*Result (WER)*:

| "clean" | "other" |
|---|---|
| 1.89 | 4.07 |