8.31 kB
--- | |
language: ar | |
datasets: | |
- common_voice | |
- arabic_speech_corpus | |
metrics: | |
- wer | |
- cer | |
tags: | |
- audio | |
- automatic-speech-recognition | |
- speech | |
- xlsr-fine-tuning-week | |
license: apache-2.0 | |
model-index: | |
- name: XLSR Wav2Vec2 Arabic by Jonatas Grosman | |
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: 39.59 | |
- name: Test CER | |
type: cer | |
value: 18.18 | |
--- | |
# Fine-tuned XLSR-53 large model for speech recognition in Arabic | |
Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Arabic using the train and validation splits of [Common Voice 6.1](https://huggingface.co/datasets/common_voice) and [Arabic Speech Corpus](https://huggingface.co/datasets/arabic_speech_corpus). | |
When using this model, make sure that your speech input is sampled at 16kHz. | |
This model has been fine-tuned thanks to the GPU credits generously given by the [OVHcloud](https://www.ovhcloud.com/en/public-cloud/ai-training/) :) | |
The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint | |
## Usage | |
The model can be used directly (without a language model) as follows... | |
Using the [HuggingSound](https://github.com/jonatasgrosman/huggingsound) library: | |
```python | |
from huggingsound import SpeechRecognitionModel | |
model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-large-xlsr-53-arabic") | |
audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"] | |
transcriptions = model.transcribe(audio_paths) | |
``` | |
Writing your own inference script: | |
```python | |
import torch | |
import librosa | |
from datasets import load_dataset | |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor | |
LANG_ID = "ar" | |
MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-arabic" | |
SAMPLES = 10 | |
test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]") | |
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID) | |
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID) | |
# Preprocessing the datasets. | |
# We need to read the audio files as arrays | |
def speech_file_to_array_fn(batch): | |
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000) | |
batch["speech"] = speech_array | |
batch["sentence"] = batch["sentence"].upper() | |
return batch | |
test_dataset = test_dataset.map(speech_file_to_array_fn) | |
inputs = processor(test_dataset["speech"], 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) | |
predicted_sentences = processor.batch_decode(predicted_ids) | |
for i, predicted_sentence in enumerate(predicted_sentences): | |
print("-" * 100) | |
print("Reference:", test_dataset[i]["sentence"]) | |
print("Prediction:", predicted_sentence) | |
``` | |
| Reference | Prediction | | |
| ------------- | ------------- | | |
| ألديك قلم ؟ | ألديك قلم | | |
| ليست هناك مسافة على هذه الأرض أبعد من يوم أمس. | ليست نالك مسافة على هذه الأرض أبعد من يوم الأمس م | | |
| إنك تكبر المشكلة. | إنك تكبر المشكلة | | |
| يرغب أن يلتقي بك. | يرغب أن يلتقي بك | | |
| إنهم لا يعرفون لماذا حتى. | إنهم لا يعرفون لماذا حتى | | |
| سيسعدني مساعدتك أي وقت تحب. | سيسئدنيمساعدتك أي وقد تحب | | |
| أَحَبُّ نظريّة علمية إليّ هي أن حلقات زحل مكونة بالكامل من الأمتعة المفقودة. | أحب نظرية علمية إلي هي أن حل قتزح المكوينا بالكامل من الأمت عن المفقودة | | |
| سأشتري له قلماً. | سأشتري له قلما | | |
| أين المشكلة ؟ | أين المشكل | | |
| وَلِلَّهِ يَسْجُدُ مَا فِي السَّمَاوَاتِ وَمَا فِي الْأَرْضِ مِنْ دَابَّةٍ وَالْمَلَائِكَةُ وَهُمْ لَا يَسْتَكْبِرُونَ | ولله يسجد ما في السماوات وما في الأرض من دابة والملائكة وهم لا يستكبرون | | |
## Evaluation | |
The model can be evaluated as follows on the Arabic test data of Common Voice. | |
```python | |
import torch | |
import re | |
import librosa | |
from datasets import load_dataset, load_metric | |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor | |
LANG_ID = "ar" | |
MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-arabic" | |
DEVICE = "cuda" | |
CHARS_TO_IGNORE = [",", "?", "¿", ".", "!", "¡", ";", ";", ":", '""', "%", '"', "�", "ʿ", "·", "჻", "~", "՞", | |
"؟", "،", "।", "॥", "«", "»", "„", "“", "”", "「", "」", "‘", "’", "《", "》", "(", ")", "[", "]", | |
"{", "}", "=", "`", "_", "+", "<", ">", "…", "–", "°", "´", "ʾ", "‹", "›", "©", "®", "—", "→", "。", | |
"、", "﹂", "﹁", "‧", "~", "﹏", ",", "{", "}", "(", ")", "[", "]", "【", "】", "‥", "〽", | |
"『", "』", "〝", "〟", "⟨", "⟩", "〜", ":", "!", "?", "♪", "؛", "/", "\\", "º", "−", "^", "'", "ʻ", "ˆ"] | |
test_dataset = load_dataset("common_voice", LANG_ID, split="test") | |
wer = load_metric("wer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/wer.py | |
cer = load_metric("cer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/cer.py | |
chars_to_ignore_regex = f"[{re.escape(''.join(CHARS_TO_IGNORE))}]" | |
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID) | |
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID) | |
model.to(DEVICE) | |
# Preprocessing the datasets. | |
# We need to read the audio files as arrays | |
def speech_file_to_array_fn(batch): | |
with warnings.catch_warnings(): | |
warnings.simplefilter("ignore") | |
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000) | |
batch["speech"] = speech_array | |
batch["sentence"] = re.sub(chars_to_ignore_regex, "", batch["sentence"]).upper() | |
return batch | |
test_dataset = test_dataset.map(speech_file_to_array_fn) | |
# Preprocessing the datasets. | |
# We need to read the audio 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(DEVICE), attention_mask=inputs.attention_mask.to(DEVICE)).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) | |
predictions = [x.upper() for x in result["pred_strings"]] | |
references = [x.upper() for x in result["sentence"]] | |
print(f"WER: {wer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}") | |
print(f"CER: {cer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}") | |
``` | |
**Test Result**: | |
In the table below I report the Word Error Rate (WER) and the Character Error Rate (CER) of the model. I ran the evaluation script described above on other models as well (on 2021-05-14). Note that the table below may show different results from those already reported, this may have been caused due to some specificity of the other evaluation scripts used. | |
| Model | WER | CER | | |
| ------------- | ------------- | ------------- | | |
| jonatasgrosman/wav2vec2-large-xlsr-53-arabic | **39.59%** | **18.18%** | | |
| bakrianoo/sinai-voice-ar-stt | 45.30% | 21.84% | | |
| othrif/wav2vec2-large-xlsr-arabic | 45.93% | 20.51% | | |
| kmfoda/wav2vec2-large-xlsr-arabic | 54.14% | 26.07% | | |
| mohammed/wav2vec2-large-xlsr-arabic | 56.11% | 26.79% | | |
| anas/wav2vec2-large-xlsr-arabic | 62.02% | 27.09% | | |
| elgeish/wav2vec2-large-xlsr-53-arabic | 100.00% | 100.56% | | |
## Citation | |
If you want to cite this model you can use this: | |
```bibtex | |
@misc{grosman2021xlsr53-large-arabic, | |
title={Fine-tuned {XLSR}-53 large model for speech recognition in {A}rabic}, | |
author={Grosman, Jonatas}, | |
howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-arabic}}, | |
year={2021} | |
} | |
``` | |