Automatic Speech Recognition
Transformers
PyTorch
JAX
Russian
wav2vec2
audio
speech
xlsr-fine-tuning-week
Eval Results (legacy)
Instructions to use anton-l/wav2vec2-large-xlsr-53-russian with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use anton-l/wav2vec2-large-xlsr-53-russian with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="anton-l/wav2vec2-large-xlsr-53-russian")# Load model directly from transformers import AutoProcessor, AutoModelForCTC processor = AutoProcessor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") model = AutoModelForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") - Notebooks
- Google Colab
- Kaggle
| language: ru | |
| datasets: | |
| - common_voice | |
| metrics: | |
| - wer | |
| tags: | |
| - audio | |
| - automatic-speech-recognition | |
| - speech | |
| - xlsr-fine-tuning-week | |
| license: apache-2.0 | |
| model-index: | |
| - name: Russian XLSR Wav2Vec2 Large 53 by Anton Lozhkov | |
| results: | |
| - task: | |
| name: Speech Recognition | |
| type: automatic-speech-recognition | |
| dataset: | |
| name: Common Voice ru | |
| type: common_voice | |
| args: ru | |
| metrics: | |
| - name: Test WER | |
| type: wer | |
| value: 17.39 | |
| # Wav2Vec2-Large-XLSR-53-Russian | |
| Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Russian 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", "ru", split="test[:2%]") | |
| processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") | |
| model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") | |
| 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 Russian test data of Common Voice. | |
| ```python | |
| import torch | |
| import torchaudio | |
| import urllib.request | |
| import tarfile | |
| import pandas as pd | |
| from tqdm.auto import tqdm | |
| from datasets import load_metric | |
| from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor | |
| # Download the raw data instead of using HF datasets to save disk space | |
| data_url = "https://voice-prod-bundler-ee1969a6ce8178826482b88e843c335139bd3fb4.s3.amazonaws.com/cv-corpus-6.1-2020-12-11/ru.tar.gz" | |
| filestream = urllib.request.urlopen(data_url) | |
| data_file = tarfile.open(fileobj=filestream, mode="r|gz") | |
| data_file.extractall() | |
| wer = load_metric("wer") | |
| processor = Wav2Vec2Processor.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") | |
| model = Wav2Vec2ForCTC.from_pretrained("anton-l/wav2vec2-large-xlsr-53-russian") | |
| model.to("cuda") | |
| cv_test = pd.read_csv("cv-corpus-6.1-2020-12-11/ru/test.tsv", sep='\t') | |
| clips_path = "cv-corpus-6.1-2020-12-11/ru/clips/" | |
| def clean_sentence(sent): | |
| sent = sent.lower() | |
| # these letters are considered equivalent in written Russian | |
| sent = sent.replace('ё', 'е') | |
| # replace non-alpha characters with space | |
| sent = "".join(ch if ch.isalpha() else " " for ch in sent) | |
| # remove repeated spaces | |
| sent = " ".join(sent.split()) | |
| return sent | |
| targets = [] | |
| preds = [] | |
| for i, row in tqdm(cv_test.iterrows(), total=cv_test.shape[0]): | |
| row["sentence"] = clean_sentence(row["sentence"]) | |
| speech_array, sampling_rate = torchaudio.load(clips_path + row["path"]) | |
| resampler = torchaudio.transforms.Resample(sampling_rate, 16_000) | |
| row["speech"] = resampler(speech_array).squeeze().numpy() | |
| inputs = processor(row["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) | |
| targets.append(row["sentence"]) | |
| preds.append(processor.batch_decode(pred_ids)[0]) | |
| # free up some memory | |
| del model | |
| del processor | |
| del cv_test | |
| print("WER: {:2f}".format(100 * wer.compute(predictions=preds, references=targets))) | |
| ``` | |
| **Test Result**: 17.39 % | |
| ## Training | |
| The Common Voice `train` and `validation` datasets were used for training. | |