* Add model card.

README.md

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+---
+language: eu
+datasets:
+- common_voice
+metrics:
+- wer
+tags:
+- audio
+- automatic-speech-recognition
+- speech
+- xlsr-fine-tuning-week
+license: apache-2.0
+model-index:
+- name: XLSR Wav2Vec2 Large 53 Basque by pcuenq 
+  results:
+  - task: 
+      name: Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: Common Voice eu
+      type: common_voice
+      args: eu
+    metrics:
+       - name: Test WER
+         type: wer
+         value: 15.34
+---
+
+# Wav2Vec2-Large-XLSR-53-EU
+
+Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Basque using the [Common Voice](https://huggingface.co/datasets/common_voice) 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", "eu", split="test[:2%]")
+
+processor = Wav2Vec2Processor.from_pretrained("pcuenq/wav2vec2-large-xlsr-53-eu")
+model = Wav2Vec2ForCTC.from_pretrained("pcuenq/wav2vec2-large-xlsr-53-eu")
+
+resampler = torchaudio.transforms.Resample(48_000, 16_000)
+
+# Preprocessing the datasets.
+# We need to read the audio 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 Basque 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", "eu", split="test")
+wer = load_metric("wer")
+
+model_name = "pcuenq/wav2vec2-large-xlsr-53-eu"
+
+processor = Wav2Vec2Processor.from_pretrained(model_name)
+model = Wav2Vec2ForCTC.from_pretrained(model_name)
+model.to("cuda")
+
+## Text pre-processing
+
+chars_to_ignore_regex = '[\,\¿\?\.\¡\!\-\;\:\"\“\%\‘\”\\…\’\ː\'\‹\›\`\´\®\—\→]'
+chars_to_ignore_pattern = re.compile(chars_to_ignore_regex)
+
+def remove_special_characters(batch):
+    batch["sentence"] = chars_to_ignore_pattern.sub('', batch["sentence"]).lower() + " "
+    return batch
+
+## Audio pre-processing
+
+import librosa
+def speech_file_to_array_fn(batch):
+    speech_array, sample_rate = torchaudio.load(batch["path"])
+    batch["speech"] = librosa.resample(speech_array.squeeze().numpy(), sample_rate, 16_000)
+    return batch
+
+# Text transformation and audio resampling
+def cv_prepare(batch):
+    batch = remove_special_characters(batch)
+    batch = speech_file_to_array_fn(batch)
+    return batch
+
+# Number of CPUs or None
+num_proc = 16
+test_dataset = test_dataset.map(cv_prepare, remove_columns=['path'], num_proc=num_proc)
+
+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)
+
+# WER Metric computation
+print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
+```
+
+**Test Result**: 15.34 %
+
+## Training
+
+The Common Voice `train` and `validation` datasets were used for training. Training was performed for 22 + 20 epochs with the following parameters:
+
+- Batch size 16, 2 gradient accumulation steps.
+- Learning rate: 2.5e-4
+- Activation dropout: 0.05
+- Attention dropout: 0.1
+- Hidden dropout: 0.05
+- Feature proj. dropout: 0.05
+- Mask time probability: 0.08
+- Layer dropout: 0.05
+