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--- |
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language: |
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- sk |
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license: apache-2.0 |
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tags: |
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- automatic-speech-recognition |
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- mozilla-foundation/common_voice_8_0 |
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- robust-speech-event |
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- xlsr-fine-tuning-week |
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- hf-asr-leaderboard |
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datasets: |
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- common_voice |
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model-index: |
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- name: Slovak comodoro Wav2Vec2 XLSR 300M CV8 |
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results: |
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- task: |
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name: Automatic Speech Recognition |
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type: automatic-speech-recognition |
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dataset: |
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name: Common Voice 8 |
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type: mozilla-foundation/common_voice_8_0 |
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args: sk |
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metrics: |
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- name: Test WER |
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type: wer |
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value: 49.6 |
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- name: Test CER |
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type: cer |
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value: 13.3 |
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- task: |
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name: Automatic Speech Recognition |
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type: automatic-speech-recognition |
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dataset: |
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name: Robust Speech Event - Dev Data |
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type: speech-recognition-community-v2/dev_data |
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args: sk |
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metrics: |
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- name: Test WER |
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type: wer |
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value: 81.7 |
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- task: |
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name: Automatic Speech Recognition |
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type: automatic-speech-recognition |
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dataset: |
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name: Robust Speech Event - Test Data |
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type: speech-recognition-community-v2/eval_data |
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args: sk |
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metrics: |
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- name: Test WER |
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type: wer |
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value: 80.26 |
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--- |
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# wav2vec2-xls-r-300m-cs-cv8
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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 8.0 dataset.
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It achieves the following results on the evaluation set:
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- WER: 0.49575384615384616
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- CER: 0.13333333333333333
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## Usage
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The model can be used directly (without a language model) as follows:
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```python
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import torch
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import torchaudio
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from datasets import load_dataset
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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test_dataset = load_dataset("mozilla-foundation/common_voice_8_0", "sk", split="test[:2%]")
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processor = Wav2Vec2Processor.from_pretrained("comodoro/wav2vec2-xls-r-300m-sk-cv8")
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model = Wav2Vec2ForCTC.from_pretrained("comodoro/wav2vec2-xls-r-300m-sk-cv8")
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resampler = torchaudio.transforms.Resample(48_000, 16_000)
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# Preprocessing the datasets.
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# We need to read the aduio files as arrays
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def speech_file_to_array_fn(batch):
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speech_array, sampling_rate = torchaudio.load(batch["path"])
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batch["speech"] = resampler(speech_array).squeeze().numpy()
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return batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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inputs = processor(test_dataset[:2]["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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print("Prediction:", processor.batch_decode(predicted_ids))
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print("Reference:", test_dataset[:2]["sentence"])
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```
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## Evaluation
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The model can be evaluated using the attached `eval.py` script:
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```
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python eval.py --model_id comodoro/wav2vec2-xls-r-300m-sk-cv8 --dataset mozilla-foundation/common_voice_8_0 --split test --config sk
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```
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## Training and evaluation data
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The Common Voice 8.0 `train` and `validation` datasets were used for training
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### Training hyperparameters
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The following hyperparameters were used during training:
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- learning_rate: 7e-4
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- train_batch_size: 32
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- eval_batch_size: 8
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- seed: 42
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- gradient_accumulation_steps: 20
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- total_train_batch_size: 640
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- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
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- lr_scheduler_type: linear
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- lr_scheduler_warmup_steps: 500
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- num_epochs: 50
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- mixed_precision_training: Native AMP
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### Framework versions
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- Transformers 4.16.0.dev0
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- Pytorch 1.10.1+cu102
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- Datasets 1.17.1.dev0
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- Tokenizers 0.11.0
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