--- language: - ar license: apache-2.0 tags: - automatic-speech-recognition - hf-asr-leaderboard - robust-speech-event datasets: - mozilla-foundation/common_voice_7_0 metrics: - wer - cer base_model: facebook/wav2vec2-xls-r-300m model-index: - name: wav2vec2-xls-r-300m-arabic results: - task: type: automatic-speech-recognition name: Speech Recognition dataset: name: Common Voice ar type: mozilla-foundation/common_voice_7_0 args: ar metrics: - type: wer value: 38.83 name: Test WER With LM - type: cer value: 15.33 name: Test CER With LM - task: type: automatic-speech-recognition name: Automatic Speech Recognition dataset: name: Robust Speech Event - Dev Data type: speech-recognition-community-v2/dev_data args: ar metrics: - type: wer value: 89.8 name: Test WER - task: type: automatic-speech-recognition name: Automatic Speech Recognition dataset: name: Robust Speech Event - Test Data type: speech-recognition-community-v2/eval_data args: ar metrics: - type: wer value: 87.46 name: Test WER --- # wav2vec2-large-xlsr-300-arabic This model is a fine-tuned version of [facebook/wav2vec2-xls-r-300m](https://huggingface.co/facebook/wav2vec2-xls-r-300m) on the common_voice dataset. It achieves the following results on the evaluation set: - Loss: 0.4514 - Wer: 0.4256 - Cer: 0.1528 #### Evaluation Commands 1. To evaluate on `mozilla-foundation/common_voice_7_0` with split `test` ```bash python eval.py --model_id kingabzpro/wav2vec2-large-xlsr-300-arabic --dataset mozilla-foundation/common_voice_7_0 --config ur --split test ``` ### Inference With LM ```python import torch from datasets import load_dataset from transformers import AutoModelForCTC, AutoProcessor import torchaudio.functional as F model_id = "kingabzpro/wav2vec2-large-xlsr-300-arabic" sample_iter = iter(load_dataset("mozilla-foundation/common_voice_8_0", "ar", split="test", streaming=True, use_auth_token=True)) sample = next(sample_iter) resampled_audio = F.resample(torch.tensor(sample["audio"]["array"]), 48_000, 16_000).numpy() model = AutoModelForCTC.from_pretrained(model_id) processor = AutoProcessor.from_pretrained(model_id) input_values = processor(resampled_audio, return_tensors="pt").input_values with torch.no_grad(): logits = model(input_values).logits transcription = processor.batch_decode(logits.numpy()).text ``` ### Training hyperparameters The following hyperparameters were used during training: - learning_rate: 0.0003 - train_batch_size: 64 - eval_batch_size: 8 - seed: 42 - gradient_accumulation_steps: 2 - total_train_batch_size: 128 - optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08 - lr_scheduler_type: linear - lr_scheduler_warmup_steps: 1000 - num_epochs: 10 - mixed_precision_training: Native AMP ### Training results | Training Loss | Epoch | Step | Validation Loss | Wer | Cer | |:-------------:|:-----:|:----:|:---------------:|:------:|:------:| | 5.4375 | 1.8 | 500 | 3.3330 | 1.0 | 1.0 | | 2.2187 | 3.6 | 1000 | 0.7790 | 0.6501 | 0.2338 | | 0.9471 | 5.4 | 1500 | 0.5353 | 0.5015 | 0.1822 | | 0.7416 | 7.19 | 2000 | 0.4889 | 0.4490 | 0.1640 | | 0.6358 | 8.99 | 2500 | 0.4514 | 0.4256 | 0.1528 | ### Framework versions - Transformers 4.17.0.dev0 - Pytorch 1.10.2+cu102 - Datasets 1.18.2.dev0 - Tokenizers 0.11.0