语音信号处理与分类项目教程

语音信号处理与分类项目教程

Speech_Signal_Processing_and_ClassificationFront-end speech processing aims at extracting proper features from short- term segments of a speech utterance, known as frames. It is a pre-requisite step toward any pattern recognition problem employing speech or audio (e.g., music). Here, we are interesting in voice disorder classification. That is, to develop two-class classifiers, which can discriminate between utterances of a subject suffering from say vocal fold paralysis and utterances of a healthy subject.The mathematical modeling of the speech production system in humans suggests that an all-pole system function is justified [1-3]. As a consequence, linear prediction coefficients (LPCs) constitute a first choice for modeling the magnitute of the short-term spectrum of speech. LPC-derived cepstral coefficients are guaranteed to discriminate between the system (e.g., vocal tract) contribution and that of the excitation. Taking into account the characteristics of the human ear, the mel-frequency cepstral coefficients (MFCCs) emerged as descriptive features of the speech spectral envelope. Similarly to MFCCs, the perceptual linear prediction coefficients (PLPs) could also be derived. The aforementioned sort of speaking tradi- tional features will be tested against agnostic-features extracted by convolu- tive neural networks (CNNs) (e.g., auto-encoders) [4]. The pattern recognition step will be based on Gaussian Mixture Model based classifiers,K-nearest neighbor classifiers, Bayes classifiers, as well as Deep Neural Networks. The Massachussets Eye and Ear Infirmary Dataset (MEEI-Dataset) [5] will be exploited. At the application level, a library for feature extraction and classification in Python will be developed. Credible publicly available resources will be 1used toward achieving our goal, such as KALDI. Comparisons will be made against [6-8].项目地址:https://gitcode.com/gh_mirrors/sp/Speech_Signal_Processing_and_Classification

1. 项目的目录结构及介绍

Speech_Signal_Processing_and_Classification/
├── CONTRIBUTING.md
├── LICENSE
├── README.md
├── feature_extraction_techniques
│   └── speech_features
├── speech_signal_classification
│   └── convolutional_neural_networks
└── two_class_classification
    └── voice_disorder_classification

• CONTRIBUTING.md: 贡献指南文件，指导如何为项目做出贡献。
• LICENSE: 项目许可证文件，本项目使用MIT许可证。
• README.md: 项目说明文件，包含项目的基本信息和使用指南。
• feature_extraction_techniques/speech_features: 特征提取技术目录，包含用于语音信号处理的特征提取方法。
• speech_signal_classification/convolutional_neural_networks: 语音信号分类目录，包含基于卷积神经网络的分类方法。
• two_class_classification/voice_disorder_classification: 两类分类问题目录，专注于语音障碍分类。

2. 项目的启动文件介绍

项目的启动文件通常位于speech_signal_classification/convolutional_neural_networks目录下，具体文件名为main.py。该文件负责启动整个语音信号分类流程，包括数据加载、模型训练和测试。

# main.py
import tensorflow as tf
from models import CNNModel
from data_loader import load_data

def main():
    # 加载数据
    train_data, test_data = load_data()
    
    # 创建模型
    model = CNNModel()
    
    # 训练模型
    model.train(train_data)
    
    # 测试模型
    model.test(test_data)

if __name__ == "__main__":
    main()

3. 项目的配置文件介绍

项目的配置文件通常位于项目根目录下，文件名为config.yaml。该文件包含项目运行所需的各种配置参数，如数据路径、模型参数、训练参数等。

# config.yaml
data_path: "path/to/data"
model_params:
  learning_rate: 0.001
  batch_size: 32
  epochs: 10
training_params:
  validation_split: 0.2
  save_path: "path/to/save/model"

以上是语音信号处理与分类项目的目录结构、启动文件和配置文件的介绍。希望这份教程能帮助你更好地理解和使用该项目。

Speech_Signal_Processing_and_ClassificationFront-end speech processing aims at extracting proper features from short- term segments of a speech utterance, known as frames. It is a pre-requisite step toward any pattern recognition problem employing speech or audio (e.g., music). Here, we are interesting in voice disorder classification. That is, to develop two-class classifiers, which can discriminate between utterances of a subject suffering from say vocal fold paralysis and utterances of a healthy subject.The mathematical modeling of the speech production system in humans suggests that an all-pole system function is justified [1-3]. As a consequence, linear prediction coefficients (LPCs) constitute a first choice for modeling the magnitute of the short-term spectrum of speech. LPC-derived cepstral coefficients are guaranteed to discriminate between the system (e.g., vocal tract) contribution and that of the excitation. Taking into account the characteristics of the human ear, the mel-frequency cepstral coefficients (MFCCs) emerged as descriptive features of the speech spectral envelope. Similarly to MFCCs, the perceptual linear prediction coefficients (PLPs) could also be derived. The aforementioned sort of speaking tradi- tional features will be tested against agnostic-features extracted by convolu- tive neural networks (CNNs) (e.g., auto-encoders) [4]. The pattern recognition step will be based on Gaussian Mixture Model based classifiers,K-nearest neighbor classifiers, Bayes classifiers, as well as Deep Neural Networks. The Massachussets Eye and Ear Infirmary Dataset (MEEI-Dataset) [5] will be exploited. At the application level, a library for feature extraction and classification in Python will be developed. Credible publicly available resources will be 1used toward achieving our goal, such as KALDI. Comparisons will be made against [6-8].项目地址:https://gitcode.com/gh_mirrors/sp/Speech_Signal_Processing_and_Classification