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README.md

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+---
+language: mn
+datasets:
+- common_voice mn
+metrics:
+- wer
+tags:
+- audio
+- automatic-speech-recognition
+- speech
+- xlsr-fine-tuning-week
+license: apache-2.0
+model-index:
+- name: XLSR Wav2Vec2 Mongolian V1 by Bayartsogt
+  results:
+  - task: 
+      name: Speech Recognition
+      type: automatic-speech-recognition
+    dataset:
+      name: Common Voice mn
+      type: common_voice
+      args: mn
+    metrics:
+       - name: Test WER
+         type: wer
+         value: 35.33
+---
+
+# Wav2Vec2-Large-XLSR-53-Mongolian-v1
+
+Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Mongolian using the [Common Voice](https://huggingface.co/datasets/common_voice).
+
+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", "mn", split="test[:2%]")
+processor = Wav2Vec2Processor.from_pretrained("bayartsogt/wav2vec2-large-xlsr-mongolian-v1")
+model = Wav2Vec2ForCTC.from_pretrained("bayartsogt/wav2vec2-large-xlsr-mongolian-v1")
+resampler = torchaudio.transforms.Resample(48_000, 16_000)
+# Preprocessing the datasets.
+# We need to read the aduio 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 Mongolian 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", "mn", split="test")
+wer = load_metric("wer")
+
+processor = Wav2Vec2Processor.from_pretrained("bayartsogt/wav2vec2-large-xlsr-mongolian-v1")
+model = Wav2Vec2ForCTC.from_pretrained("bayartsogt/wav2vec2-large-xlsr-mongolian-v1")
+model.to("cuda")
+
+chars_to_ignore_regex = '[\!\"\'\,\.\«\»\?\-]'
+resampler = torchaudio.transforms.Resample(48_000, 16_000)
+
+# Preprocessing the datasets.
+# We need to read the aduio files as arrays
+def speech_file_to_array_fn(batch):
+     batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower()
+     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)
+
+# Preprocessing the datasets.
+# We need to read the aduio 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("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)
+
+print("WER: {:2f}".format(100 * wer.compute(predictions=result["pred_strings"], references=result["sentence"])))
+```
+
+**Test Result**: 35.33 % 
+
+
+## Training
+
+The Common Voice `train` dataset was used for training as well as ... and ...