🚀 格鲁吉亚语Wav2Vec2-Large-XLSR-53模型
本项目基于格鲁吉亚语对 facebook/wav2vec2-large-xlsr-53 模型进行了微调,使用的数据集为 Common Voice。使用此模型时,请确保输入的语音采样率为 16kHz。
数据集
评估指标
标签
- audio
- automatic-speech-recognition
- speech
- xlsr-fine-tuning-week
许可证
apache-2.0
模型信息
模型名称 |
任务 |
数据集 |
评估指标 |
值 |
Georgian WAV2VEC2 Daytona |
语音识别(automatic-speech-recognition) |
Common Voice ka |
Test WER |
48.34 |
🚀 快速开始
本模型是基于Transformer架构的语音识别模型,通过在格鲁吉亚语数据集上微调预训练模型,实现了对格鲁吉亚语语音的准确识别。
✨ 主要特性
💻 使用示例
基础用法
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "ka", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("Temur/wav2vec2-Georgian-Daytona")
model = Wav2Vec2ForCTC.from_pretrained("Temur/wav2vec2-Georgian-Daytona")
resampler = torchaudio.transforms.Resample(48_000, 16_000)
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])
📚 详细文档
评估
可以按照以下方式在 Common Voice 的格鲁吉亚语测试数据上对模型进行评估:
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
test_dataset = load_dataset("common_voice", "ka", split="test")
wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained("Temur/wav2vec2-Georgian-Daytona")
model = Wav2Vec2ForCTC.from_pretrained("Temur/wav2vec2-Georgian-Daytona")
model.to("cuda")
chars_to_ignore_regex = '[\\\\,\\\\?\\\\.\\\\!\\\\-\\\\;\\\\:\\\\"\\\\“]'
resampler = torchaudio.transforms.Resample(48_000, 16_000)
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)
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"])))
测试结果:48.34 %
训练
Common Voice 的 train
、validation
等数据集用于模型训练。训练脚本可参考 此处。
📄 许可证
本项目采用 apache-2.0 许可证。