pytorch wav2vec2 ar transformers audio automatic-speech-recognition speech xlsr-fine-tuning-week license:apache-2.0
.gitattributes README.md config.json preprocessor_config.json pytorch_model.bin special_tokens_map.json tokenizer_config.json vocab.json .ipynb_checkpoints/README-checkpoint.md .ipynb_checkpoints/preprocessor_config-checkpoint.json .ipynb_checkpoints/special_tokens_map-checkpoint.json .ipynb_checkpoints/tokenizer_config-checkpoint.json .ipynb_checkpoints/vocab-checkpoint.json
---
language: ar
datasets:
- common_voice: Common Voice Corpus 4
metrics:
- wer
tags:
- audio
- automatic-speech-recognition
- speech
- xlsr-fine-tuning-week
license: apache-2.0
model-index:
- name: Hasni XLSR Wav2Vec2 Large 53
  results:
  - task: 
      name: Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: Common Voice ar 
      type: common_voice
      args: ar
    metrics:
       - name: Test WER
         type: wer
         value: 52.18
---

# Wav2Vec2-Large-XLSR-53-Arabic

Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Arabic using the [Common Voice Corpus 4](https://commonvoice.mozilla.org/en/datasets) dataset.
When using this model, make sure that your speech input is sampled at 16kHz.

## Usage

The model can be used directly (without a language model) as follows:

```python
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor

test_dataset = load_dataset("common_voice", "ar", split="test[:2%]")

processor = Wav2Vec2Processor.from_pretrained("anas/wav2vec2-large-xlsr-arabic")
model = Wav2Vec2ForCTC.from_pretrained("anas/wav2vec2-large-xlsr-arabic")

resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = resampler(speech_array).squeeze().numpy()
    return batch

test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)

with torch.no_grad():
     logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits

predicted_ids = torch.argmax(logits, dim=-1)

print("Prediction:", processor.batch_decode(predicted_ids))
print("Reference:", test_dataset["sentence"][:2])
```


## Evaluation

The model can be evaluated as follows on the Arabic test data of Common Voice.


```python
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re

test_dataset = load_dataset("common_voice", "ar", split="test")

processor = Wav2Vec2Processor.from_pretrained("anas/wav2vec2-large-xlsr-arabic")
model = Wav2Vec2ForCTC.from_pretrained("anas/wav2vec2-large-xlsr-arabic/")
model.to("cuda")

chars_to_ignore_regex = '[\\\\,\\\\؟\\\\.\\\\!\\\\-\\\\;\\\\\\\\:\\\\'\\\\"\\\\☭\\\\«\\\\»\\\\؛\\\\—\\\\ـ\\\\_\\\\،\\\\“\\\\%\\\\‘\\\\”\\\\�]'

resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def speech_file_to_array_fn(batch):
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
    batch["sentence"] = re.sub('[a-z]','',batch["sentence"])
    batch["sentence"] = re.sub("[إأٱآا]", "ا", batch["sentence"])
    noise = re.compile(""" ّ    | # Tashdid
                             َ    | # Fatha
                             ً    | # Tanwin Fath
                             ُ    | # Damma
                             ٌ    | # Tanwin Damm
                             ِ    | # Kasra
                             ٍ    | # Tanwin Kasr
                             ْ    | # Sukun
                             ـ     # Tatwil/Kashida
                         """, re.VERBOSE)
    batch["sentence"] = re.sub(noise, '', batch["sentence"])
    speech_array, sampling_rate = torchaudio.load(batch["path"])
    batch["speech"] = resampler(speech_array).squeeze().numpy()
    return batch

test_dataset = test_dataset.map(speech_file_to_array_fn)

# Preprocessing the datasets.
# We need to read the aduio files as arrays
def evaluate(batch):
    inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)

    with torch.no_grad():
         logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits

         pred_ids = torch.argmax(logits, dim=-1)
         batch["pred_strings"] = processor.batch_decode(pred_ids)
    return batch

result = test_dataset.map(evaluate, batched=True, batch_size=8)

print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
```

**Test Result**: 52.18 %


## Training

The Common Voice Corpus 4 `train`, `validation`, datasets were used for training

The script used for training can be found [here](...)

Twitter: [here](https://twitter.com/hasnii_anas)

Email: anashasni146@gmail.com