--- language: ja datasets: - common_voice metrics: - wer tags: - audio - automatic-speech-recognition - speech - xlsr-fine-tuning-week license: apache-2.0 model-index: - name: XLSR Wav2Vec2 Japanese by Chien Vu results: - task: name: Speech Recognition type: automatic-speech-recognition dataset: name: Common Voice Japanese type: common_voice args: ja metrics: - name: Test WER type: wer value: 30.837004 --- # Wav2Vec2-Large-XLSR-53-Japanese Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Japanese using the [Common Voice](https://huggingface.co/datasets/common_voice) and Japanese speech corpus of Saruwatari-lab, University of Tokyo [JSUT](https://sites.google.com/site/shinnosuketakamichi/publication/jsut). 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 import librosa from datasets import load_dataset import MeCab import pykakasi from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor # config wakati = MeCab.Tagger("-Owakati") chars_to_ignore_regex = '[\,\、\。\.\「\」\…\?\・]' # load data, processor and model test_dataset = load_dataset("common_voice", "ja", split="test[:2%]") processor = Wav2Vec2Processor.from_pretrained("vumichien/wav2vec2-large-xlsr-japanese") model = Wav2Vec2ForCTC.from_pretrained("vumichien/wav2vec2-large-xlsr-japanese") resampler = lambda sr, y: librosa.resample(y.numpy().squeeze(), sr, 16_000) # Preprocessing the datasets. def speech_file_to_array_fn(batch): batch["sentence"] = wakati.parse(batch["sentence"]).strip() batch["sentence"] = re.sub(chars_to_ignore_regex,'', batch["sentence"]).strip() speech_array, sampling_rate = torchaudio.load(batch["path"]) batch["speech"] = resampler(sampling_rate, speech_array).squeeze() 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 Japanese test data of Common Voice. ```python import torch import librosa import torchaudio from datasets import load_dataset import MeCab import pykakasi from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor #config wakati = MeCab.Tagger("-Owakati") chars_to_ignore_regex = '[\,\、\。\.\「\」\…\?\・]' # load data, processor and model test_dataset = load_dataset("common_voice", "ja", split="test") wer = load_metric("wer") processor = Wav2Vec2Processor.from_pretrained("vumichien/wav2vec2-large-xlsr-japanese") model = Wav2Vec2ForCTC.from_pretrained("vumichien/wav2vec2-large-xlsr-japanese") model.to("cuda") resampler = lambda sr, y: librosa.resample(y.numpy().squeeze(), sr, 16_000) # Preprocessing the datasets. def speech_file_to_array_fn(batch): batch["sentence"] = kakasi.do(wakati.parse(batch["sentence"]).strip()) batch["sentence"] = re.sub(chars_to_ignore_regex,'', batch["sentence"]).strip() speech_array, sampling_rate = torchaudio.load(batch["path"]) batch["speech"] = resampler(sampling_rate, speech_array).squeeze() return batch test_dataset = test_dataset.map(speech_file_to_array_fn) # evaluate function 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**: 30.837004 ## Training The Common Voice `train`, `validation` datasets and Japanese speech corpus `basic5000` datasets were used for training. The script used for training can be found [here](https://colab.research.google.com/drive/1ZTxoYzgOotUjcyoBf0m8gZj5Kcmu2yGU)