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- TODO
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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+ ---
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+ language: mn
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+ datasets:
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+ - common_voice
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+ metrics:
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+ - wer
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+ tags:
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+ - audio
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+ - automatic-speech-recognition
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+ - speech
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+ - xlsr-fine-tuning-week
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+ license: apache-2.0
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+ model-index:
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+ - name: Mongolian XLSR Wav2Vec2 Large 53 by Anton Lozhkov
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+ results:
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+ - task:
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+ name: Speech Recognition
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+ type: automatic-speech-recognition
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+ dataset:
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+ name: Common Voice mn
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+ type: common_voice
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+ args: mn
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+ metrics:
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+ - name: Test WER
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+ type: wer
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+ value: 38.53
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+ ---
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+
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+ # Wav2Vec2-Large-XLSR-53-Mongolian
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+
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+ 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) dataset.
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+ When using this model, make sure that your speech input is sampled at 16kHz.
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+
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+ ## Usage
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+
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+ The model can be used directly (without a language model) as follows:
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+
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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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+
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+ test_dataset = load_dataset("common_voice", "mn", split="test[:2%]")
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+
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+ processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
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+ model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
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+
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+ resampler = torchaudio.transforms.Resample(48_000, 16_000)
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+
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+ # Preprocessing the datasets.
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+ # We need to read the audio 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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+
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+ test_dataset = test_dataset.map(speech_file_to_array_fn)
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+ inputs = processor(test_dataset["speech"][:2], sampling_rate=16_000, return_tensors="pt", padding=True)
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+
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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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+
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+ predicted_ids = torch.argmax(logits, dim=-1)
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+
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+ print("Prediction:", processor.batch_decode(predicted_ids))
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+ print("Reference:", test_dataset["sentence"][:2])
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+ ```
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+
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+
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+ ## Evaluation
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+
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+ The model can be evaluated as follows on the Mongolian test data of Common Voice.
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+
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+ ```python
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+ import torch
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+ import torchaudio
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+ import urllib.request
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+ import tarfile
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+ import pandas as pd
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+ from tqdm.auto import tqdm
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+ from datasets import load_metric
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+ from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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+
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+ # Download the raw data instead of using HF datasets to save disk space
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+ data_url = "https://voice-prod-bundler-ee1969a6ce8178826482b88e843c335139bd3fb4.s3.amazonaws.com/cv-corpus-6.1-2020-12-11/mn.tar.gz"
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+ filestream = urllib.request.urlopen(data_url)
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+ data_file = tarfile.open(fileobj=filestream, mode="r|gz")
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+ data_file.extractall()
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+
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+ wer = load_metric("wer")
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+
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+ processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
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+ model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-mongolian")
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+ model.to("cuda")
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+
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+ cv_test = pd.read_csv("cv-corpus-6.1-2020-12-11/mn/test.tsv", sep='\t')
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+ clips_path = "cv-corpus-6.1-2020-12-11/mn/clips/"
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+
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+ def clean_sentence(sent):
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+ sent = sent.lower()
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+ # replace non-alpha characters with space
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+ sent = "".join(ch if ch.isalpha() else " " for ch in sent)
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+ # remove repeated spaces
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+ sent = " ".join(sent.split())
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+ return sent
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+
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+ targets = []
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+ preds = []
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+
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+ for i, row in tqdm(cv_test.iterrows(), total=cv_test.shape[0]):
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+ row["sentence"] = clean_sentence(row["sentence"])
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+ speech_array, sampling_rate = torchaudio.load(clips_path + row["path"])
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+ resampler = torchaudio.transforms.Resample(sampling_rate, 16_000)
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+ row["speech"] = resampler(speech_array).squeeze().numpy()
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+
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+ inputs = processor(row["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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+
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+ with torch.no_grad():
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+ logits = model(inputs.input_values.to("cuda"), attention_mask=inputs.attention_mask.to("cuda")).logits
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+
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+ pred_ids = torch.argmax(logits, dim=-1)
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+
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+ targets.append(row["sentence"])
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+ preds.append(processor.batch_decode(pred_ids)[0])
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+
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+ print("WER: {:2f}".format(100 * wer.compute(predictions=preds, references=targets)))
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+ ```
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+
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+ **Test Result**: 38.53 %
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+
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+
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+ ## Training
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+
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+ The Common Voice `train` and `validation` datasets were used for training.
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+