Wav2Vec2-Large-XLSR-53-Arabic

Fine-tuned facebook/wav2vec2-large-xlsr-53 on Arabic using the train splits of Common Voice and Arabic Speech Corpus. 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:

import torch
import torchaudio
from datasets import load_dataset
from lang_trans.arabic import buckwalter
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor

dataset = load_dataset("common_voice", "ar", split="test[:10]")
resamplers = {  # all three sampling rates exist in test split
    48000: torchaudio.transforms.Resample(48000, 16000),
    44100: torchaudio.transforms.Resample(44100, 16000),
    32000: torchaudio.transforms.Resample(32000, 16000),
}

def prepare_example(example):
    speech, sampling_rate = torchaudio.load(example["path"])
    example["speech"] = resamplers[sampling_rate](speech).squeeze().numpy()
    return example

dataset = dataset.map(prepare_example)
processor = Wav2Vec2Processor.from_pretrained("elgeish/wav2vec2-large-xlsr-53-arabic")
model = Wav2Vec2ForCTC.from_pretrained("elgeish/wav2vec2-large-xlsr-53-arabic").eval()

def predict(batch):
    inputs = processor(batch["speech"], sampling_rate=16000, return_tensors="pt", padding=True)
    with torch.no_grad():
        predicted = torch.argmax(model(inputs.input_values).logits, dim=-1)
    predicted[predicted == -100] = processor.tokenizer.pad_token_id  # see fine-tuning script
    batch["predicted"] = processor.tokenizer.batch_decode(predicted)
    return batch

dataset = dataset.map(predict, batched=True, batch_size=1, remove_columns=["speech"])

for reference, predicted in zip(dataset["sentence"], dataset["predicted"]):
    print("reference:", reference)
    print("predicted:", buckwalter.untrans(predicted))
    print("--")

Here's the output:

reference: ألديك قلم ؟
predicted: هلديك قالر
--
reference: ليست هناك مسافة على هذه الأرض أبعد من يوم أمس.
predicted: ليست نالك مسافة على هذه الأرض أبعد من يوم أمس
--
reference: إنك تكبر المشكلة.
predicted: إنك تكبر المشكلة
--
reference: يرغب أن يلتقي بك.
predicted: يرغب أن يلتقي بك
--
reference: إنهم لا يعرفون لماذا حتى.
predicted: إنهم لا يعرفون لماذا حتى
--
reference: سيسعدني مساعدتك أي وقت تحب.
predicted: سيسئدني مساعد سكرأي وقت تحب
--
reference: أَحَبُّ نظريّة علمية إليّ هي أن حلقات زحل مكونة بالكامل من الأمتعة المفقودة.
predicted: أحب ناضريةً علمية إلي  هي أنحل قتزح المكونا بالكامل من الأمت عن المفقودة
--
reference: سأشتري له قلماً.
predicted: سأشتري له قلما
--
reference: أين المشكلة ؟
predicted: أين المشكل
--
reference: وَلِلَّهِ يَسْجُدُ مَا فِي السَّمَاوَاتِ وَمَا فِي الْأَرْضِ مِنْ دَابَّةٍ وَالْمَلَائِكَةُ وَهُمْ لَا يَسْتَكْبِرُونَ
predicted: 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:

import jiwer
import torch
import torchaudio
from datasets import load_dataset
from lang_trans.arabic import buckwalter
from transformers import set_seed, Wav2Vec2ForCTC, Wav2Vec2Processor

set_seed(42)
test_split = load_dataset("common_voice", "ar", split="test")
resamplers = {  # all three sampling rates exist in test split
    48000: torchaudio.transforms.Resample(48000, 16000),
    44100: torchaudio.transforms.Resample(44100, 16000),
    32000: torchaudio.transforms.Resample(32000, 16000),
}

def prepare_example(example):
    speech, sampling_rate = torchaudio.load(example["path"])
    example["speech"] = resamplers[sampling_rate](speech).squeeze().numpy()
    return example

test_split = test_split.map(prepare_example)
processor = Wav2Vec2Processor.from_pretrained("elgeish/wav2vec2-large-xlsr-53-arabic")
model = Wav2Vec2ForCTC.from_pretrained("elgeish/wav2vec2-large-xlsr-53-arabic").to("cuda").eval()

def predict(batch):
    inputs = processor(batch["speech"], sampling_rate=16000, return_tensors="pt", padding=True)
    with torch.no_grad():
        predicted = torch.argmax(model(inputs.input_values.to("cuda")).logits, dim=-1)
    predicted[predicted == -100] = processor.tokenizer.pad_token_id  # see fine-tuning script
    batch["predicted"] = processor.batch_decode(predicted)
    return batch

test_split = test_split.map(predict, batched=True, batch_size=16, remove_columns=["speech"])
transformation = jiwer.Compose([
    # normalize some diacritics, remove punctuation, and replace Persian letters with Arabic ones
    jiwer.SubstituteRegexes({
        r'[auiFNKo\~_،؟»\?;:\-,\.؛«!"]': "", "\u06D6": "",
        r"[\|\{]": "A", "p": "h", "ک": "k", "ی": "y"}),
    # default transformation below
    jiwer.RemoveMultipleSpaces(),
    jiwer.Strip(),
    jiwer.SentencesToListOfWords(),
    jiwer.RemoveEmptyStrings(),
])
metrics = jiwer.compute_measures(
    truth=[buckwalter.trans(s) for s in test_split["sentence"]],  # Buckwalter transliteration
    hypothesis=test_split["predicted"],
    truth_transform=transformation,
    hypothesis_transform=transformation,
)
print(f"WER: {metrics['wer']:.2%}")

Test Result: 26.55%

Training

For more details, see Fine-Tuning with Arabic Speech Corpus.

This model represents Arabic in a format called Buckwalter transliteration. The Buckwalter format only includes ASCII characters, some of which are non-alpha (e.g., ">" maps to "أ"). The lang-trans package is used to convert (transliterate) Arabic abjad.

This script was used to first fine-tune facebook/wav2vec2-large-xlsr-53 on the train split of the Arabic Speech Corpus dataset; the test split was used for model selection; the resulting model at this point is saved as elgeish/wav2vec2-large-xlsr-53-levantine-arabic.

Training was then resumed using the train split of the Common Voice dataset; the validation split was used for model selection; training was stopped to meet the deadline of Fine-Tune-XLSR Week: this model is the checkpoint at 100k steps and a validation WER of 23.39%.

Validation WER

It's worth noting that validation WER is trending down, indicating the potential of further training (resuming the decaying learning rate at 7e-6).

Future Work

One area to explore is using attention_mask in model input, which is recommended here. Also, exploring data augmentation using datasets used to train models listed here.

Downloads last month
938
Hosted inference API
Automatic Speech Recognition
Record from browser