Instructions to use jonatasgrosman/wav2vec2-large-xlsr-53-japanese with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use jonatasgrosman/wav2vec2-large-xlsr-53-japanese with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("automatic-speech-recognition", model="jonatasgrosman/wav2vec2-large-xlsr-53-japanese")# Load model directly from transformers import AutoProcessor, AutoModelForCTC processor = AutoProcessor.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-japanese") model = AutoModelForCTC.from_pretrained("jonatasgrosman/wav2vec2-large-xlsr-53-japanese") - Notebooks
- Google Colab
- Kaggle
Fine-tuned XLSR-53 large model for speech recognition in Japanese
Fine-tuned facebook/wav2vec2-large-xlsr-53 on Japanese using the train and validation splits of Common Voice 6.1, CSS10 and JSUT. When using this model, make sure that your speech input is sampled at 16kHz.
This model has been fine-tuned thanks to the GPU credits generously given by the OVHcloud :)
The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
Usage
The model can be used directly (without a language model) as follows...
Using the HuggingSound library:
from huggingsound import SpeechRecognitionModel
model = SpeechRecognitionModel("jonatasgrosman/wav2vec2-large-xlsr-53-japanese")
audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
transcriptions = model.transcribe(audio_paths)
Writing your own inference script:
import torch
import librosa
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
LANG_ID = "ja"
MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-japanese"
SAMPLES = 10
test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
batch["speech"] = speech_array
batch["sentence"] = batch["sentence"].upper()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
inputs = processor(test_dataset["speech"], 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)
predicted_sentences = processor.batch_decode(predicted_ids)
for i, predicted_sentence in enumerate(predicted_sentences):
print("-" * 100)
print("Reference:", test_dataset[i]["sentence"])
print("Prediction:", predicted_sentence)
| Reference | Prediction |
|---|---|
| ็ฅๆฏใฏใใใใใญๆฉๅซใใใใตใคใณใญใใใใใใฆใใใ | ไบบๆฏใฏ้ใซใใญ่ตทใใใใใใฆใใ |
| ่ฒกๅธใใชใใใใฎใงใไบค็ชใธ่กใใพใใ | ่ฒกๅธใใชใๆ็ซฏใฎใงๅพ็ชใธ่กใใพใ |
| ้ฃฒใฟๅฑใฎใใใใๆ ้คจใฎไธปไบบใๅป่ ใใฏใใใไบค้ใฎใใไบบใซใใใฆใพใใฃใใใใฟใใชใ็งใใๅๅ ฅใๅคใใฏใใชใฎใซใ็จ้ใฏๅฎใใ | ใๅฎฎๅฑใฎใ่ฆชใๆ ้คจใฎไธปใซๅป่ ใใฏใใไบค้ใฎใขใซไบบใใซ่ใใฆๅใฃใใใฟใใช็งใใๅๅ ฅใๅคใใฏใชใใซ็จ้ใฏๅฎใ |
| ๆฐใใ้ดใใฏใใฆๅบใใใพใใ | ใ ใใใ้ดใใฏใใฆๅบใใใพใ |
| ใใฎใใใใฉใบใไธญใฎใคใชใณใ้ปๅญใฎๆใคๅนณๅ้ๅใจใใซใฎใผใๆธฉๅบฆใง่กจ็พใใใใจใใใ | ใใฎใใใใฉใบใไธญใฎใคใชใณใ้ปๅญใฎๆใคๅนณๅ้ๅใจใใซใฎใผใๆธฉๅบฆใง่กจๅผใใใใจใใใ |
| ๆพไบใใใฏใตใใซใผใใ้็ใฎใปใใไธๆใงใใ | ๆพไบใใใฏใตใใซใผใใ้็ใฎใปใใไธๆใงใ |
| ๆฐใใใ็ฟใไฝฟใใพใใ | ๆฐใใใ็ฟใไฝฟใใพใ |
| ็ตๅฉไปฅๆฅไธๅนดๅใถใใฎๆฑไบฌใใๆงๅใจใฎใ้ ใใๅค่กๅ่ปใใ้ง ใงๅฏใฆใๆใๅพ ใคใฎใไน ใใถใใ ใ | ็ตๅฉใซไบๆฅไธๅนดๅ้ใใฎๆฑไบฌใๅธใจใฎใ้ ใ้่ถ่ ใ้ง ใงๅฏใฆๆใๅพ ใคใฎไน ใใถใใ |
| ใใใพใงใๅฐๅนด้็ใใใใใใใฌใผใชใฉใๅฐๅในใใผใใๆฏใใๅธๆฐใซๅฏ็ใใฆใใใฎใฏใ็กๆฐใฎใใฉใณใใฃใขใ ใฃใใ | ใใใพใงๅฐๅนด้็ไธใใฌใผใชใฉๅฐๅในใใผใใๆฏใๅธๆฐใซๆบ็ใใฆใใใฎใฏๅจๆฐใฎใใฉใณใใฃใขใ ใฃใ |
| ้ดใ่ฑใใงใในใชใใใใฏใใพใใ | ้ดใ่ฑใใงในใคใใผใใฏใใพใ |
Evaluation
The model can be evaluated as follows on the Japanese test data of Common Voice.
import torch
import re
import librosa
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
LANG_ID = "ja"
MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-japanese"
DEVICE = "cuda"
CHARS_TO_IGNORE = [",", "?", "ยฟ", ".", "!", "ยก", ";", "๏ผ", ":", '""', "%", '"', "๏ฟฝ", "สฟ", "ยท", "แป", "~", "ี",
"ุ", "ุ", "เฅค", "เฅฅ", "ยซ", "ยป", "โ", "โ", "โ", "ใ", "ใ", "โ", "โ", "ใ", "ใ", "(", ")", "[", "]",
"{", "}", "=", "`", "_", "+", "<", ">", "โฆ", "โ", "ยฐ", "ยด", "สพ", "โน", "โบ", "ยฉ", "ยฎ", "โ", "โ", "ใ",
"ใ", "๏น", "๏น", "โง", "๏ฝ", "๏น", "๏ผ", "๏ฝ", "๏ฝ", "๏ผ", "๏ผ", "๏ผป", "๏ผฝ", "ใ", "ใ", "โฅ", "ใฝ",
"ใ", "ใ", "ใ", "ใ", "โจ", "โฉ", "ใ", "๏ผ", "๏ผ", "๏ผ", "โช", "ุ", "/", "\\", "ยบ", "โ", "^", "'", "สป", "ห"]
test_dataset = load_dataset("common_voice", LANG_ID, split="test")
wer = load_metric("wer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/wer.py
cer = load_metric("cer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/cer.py
chars_to_ignore_regex = f"[{re.escape(''.join(CHARS_TO_IGNORE))}]"
processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
model.to(DEVICE)
# Preprocessing the datasets.
# We need to read the audio files as arrays
def speech_file_to_array_fn(batch):
with warnings.catch_warnings():
warnings.simplefilter("ignore")
speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
batch["speech"] = speech_array
batch["sentence"] = re.sub(chars_to_ignore_regex, "", batch["sentence"]).upper()
return batch
test_dataset = test_dataset.map(speech_file_to_array_fn)
# Preprocessing the datasets.
# We need to read the audio 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(DEVICE), attention_mask=inputs.attention_mask.to(DEVICE)).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)
predictions = [x.upper() for x in result["pred_strings"]]
references = [x.upper() for x in result["sentence"]]
print(f"WER: {wer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")
print(f"CER: {cer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")
Test Result:
In the table below I report the Word Error Rate (WER) and the Character Error Rate (CER) of the model. I ran the evaluation script described above on other models as well (on 2021-05-10). Note that the table below may show different results from those already reported, this may have been caused due to some specificity of the other evaluation scripts used.
| Model | WER | CER |
|---|---|---|
| jonatasgrosman/wav2vec2-large-xlsr-53-japanese | 81.80% | 20.16% |
| vumichien/wav2vec2-large-xlsr-japanese | 1108.86% | 23.40% |
| qqhann/w2v_hf_jsut_xlsr53 | 1012.18% | 70.77% |
Citation
If you want to cite this model you can use this:
@misc{grosman2021xlsr53-large-japanese,
title={Fine-tuned {XLSR}-53 large model for speech recognition in {J}apanese},
author={Grosman, Jonatas},
howpublished={\url{https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-japanese}},
year={2021}
}
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Evaluation results
- Test WER on Common Voice jaself-reported81.800
- Test CER on Common Voice jaself-reported20.160