Wav2vec2 Base 760 Turkish

🚀 Wav2Vec2-Base-760-土耳其语模型

本模型是基于Transformer架构的语音识别模型，通过在土耳其语数据集上微调预训练模型得到，可用于土耳其语的自动语音识别任务。

🚀 快速开始

本模型是预训练的土耳其语模型 ceyda/wav2vec2-base-760，在 Common Voice 土耳其语数据集上进行了微调。使用此模型时，请确保输入的语音采样率为16kHz。

✨ 主要特性

• 数据集：使用了 Common Voice 数据集进行训练和评估。
• 评估指标：使用字错率（WER）作为评估指标。
• 应用场景：适用于土耳其语的自动语音识别任务。

📦 安装指南

文档未提供具体安装步骤，跳过该章节。

💻 使用示例

基础用法

模型可以直接使用（无需语言模型），示例代码如下：

import torch
import torchaudio
from datasets import load_dataset
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor

test_dataset = load_dataset("common_voice", "tr", split="test[:2%]") 

processor = Wav2Vec2Processor.from_pretrained("ceyda/wav2vec2-base-960-turkish")
model = Wav2Vec2ForCTC.from_pretrained("ceyda/wav2vec2-base-960-turkish")

resampler = torchaudio.transforms.Resample(48_000, 16_000)

# Preprocessing the datasets.
# We need to read the aduio 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])

高级用法

在 Common Voice 土耳其语测试数据上评估模型的代码如下：

import torch
import torchaudio
from datasets import load_dataset, load_metric
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
import re

test_dataset = load_dataset("common_voice", "tr", split="test")
wer = load_metric("wer")

processor = Wav2Vec2Processor.from_pretrained("ceyda/wav2vec2-base-960-turkish")
model = Wav2Vec2ForCTC.from_pretrained("ceyda/wav2vec2-base-960-turkish")
model.to("cuda")

chars_to_ignore_regex = '[\,\?\.\!\-\;\:\"\“\‘\”\'\`…\’»«]'
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):
    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)

# Preprocessing the datasets.
# We need to read the aduio files as arrays

#Attention mask is not used because the base-model was not trained with it. reference: https://github.com/huggingface/transformers/blob/403d530eec105c0e229fc2b754afdf77a4439def/src/transformers/models/wav2vec2/tokenization_wav2vec2.py#L305
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")).logits

    pred_ids = torch.argmax(logits, dim=-1)
    batch["pred_strings"] = processor.batch_decode(pred_ids,skip_special_tokens=True)
    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

• 字错率（WER）: 22.602390
• 字符错误率（CER）: 6.054137

📚 详细文档

训练信息

使用了 Common Voice 数据集的 train 和 validation 子集进行训练。训练脚本可在 此处 找到。

📄 许可证

本模型使用 Apache-2.0 许可证。

模型信息