语言: 芬兰语
数据集:
评估指标:
标签:
许可协议: Apache-2.0
模型索引:
- 名称: 芬兰语XLSR Wav2Vec2大模型53版
结果:
- 任务:
名称: 语音识别
类型: 自动语音识别
数据集:
名称: 通用语音库芬兰语版
类型: common_voice
参数: fi
指标:
- 名称: 测试WER
类型: wer
值: 35.43
Wav2Vec2大模型-XLSR-53-芬兰语版
基于facebook/wav2vec2-large-xlsr-53模型,使用通用语音库、CSS10和芬兰议会会议记录第二期数据集对芬兰语进行微调。
使用该模型时,请确保语音输入采样率为16kHz。
使用方法
该模型可直接使用(无需语言模型),如下所示:
import numpy as np
import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
test_dataset = load_dataset("common_voice", "fi", split="test[:2%]")
processor = Wav2Vec2Processor.from_pretrained("Tommi/wav2vec2-large-xlsr-53-finnish")
model = Wav2Vec2ForCTC.from_pretrained("Tommi/wav2vec2-large-xlsr-53-finnish")
resampler = lambda sr, y: librosa.resample(y.squeeze(), sr, 16_000)
def speech_file_to_array_fn(batch):
speech_array, sampling_rate = torchaudio.load(batch["path"])
batch["speech"] = resampler(sampling_rate, speech_array.numpy()).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("预测结果:", processor.batch_decode(predicted_ids))
print("参考文本:", test_dataset["sentence"][:2])
评估
可在通用语音库芬兰语测试数据上按如下方式评估模型:
import librosa
import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re
test_dataset = load_dataset("common_voice", "fi", split="test")
wer = load_metric("wer")
processor = Wav2Vec2Processor.from_pretrained("Tommi/wav2vec2-large-xlsr-53-finnish")
model = Wav2Vec2ForCTC.from_pretrained("Tommi/wav2vec2-large-xlsr-53-finnish")
model.to("cuda")
chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\"\%\'\"\�\'\...\…\–\é]'
resampler = lambda sr, y: librosa.resample(y.numpy().squeeze(), sr, 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(sampling_rate, speech_array).squeeze()
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: 35.43%
训练
训练使用了通用语音库的train
、validation
和other
数据集,以及CSS10和芬兰议会会议记录第二期。
训练脚本链接在此 # 待补充:请在此处填写训练脚本链接。如果在colab中训练模型,直接在此处填写链接。如果在本地训练模型,建议将训练脚本上传至github并在此处粘贴链接。