MIT License

项目介绍

gionanide/Speech_Signal_Processing_and_Classification 是一个托管在 GitHub 上的开源项目,主要使用 Python 编写,采用 MIT License 许可证,目前获得 257 个 Star、63 次 Fork。

仓库全名
gionanide/Speech_Signal_Processing_and_Classification
开源许可证
MIT License
Star 数
257
Fork 数
63
未关闭 Issue
3

README 摘要

Speech-Signal-Processing-and-Classification Aristotle University of Thessaloniki - University of Groningen Abstract of my thesis conducted during the 7-8th semester. " Two-class classification problems by analyzing the speech signal. " Front-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…

完整内容见 GitHub 原始 README 或本页 。

常见问题

Speech_Signal_Processing_and_Classification 用什么语言开发?

主要使用 Python 编写,采用 MIT License 许可证,目前有 257 个 Star。

Speech_Signal_Processing_and_Classification 可以免费商用吗?

仓库采用 MIT License 许可证,可免费使用与修改,商业使用请遵守许可证条款(例如保留版权声明与许可声明)。

如何获取 Speech_Signal_Processing_and_Classification 的源码?

可以直接克隆仓库:git clone https://github.com/gionanide/Speech_Signal_Processing_and_Classification.git,默认分支为 master;安装与运行步骤见仓库 README。

Speech_Signal_Processing_and_Classification 最近还在维护吗?

仓库最近一次更新为 2023年3月4日,当前有 3 个未关闭 Issue;Star 数为 257,可作为社区活跃度的参考。

关于

GitHub 原始描述(英文)

Front-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].

描述取自 GitHub 官方 API,保持仓库作者书写的原文。

语言占比

Python

README

Speech-Signal-Processing-and-Classification

Aristotle University of Thessaloniki - University of Groningen

Abstract of my thesis conducted during the 7-8th semester. " Two-class classification problems by analyzing the speech signal. "

Front-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 traditional features will be tested against agnostic-features extracted by convolutive neural networks (CNNs) (e.g., auto-encoders) [4]. Additionally as concerns SVM algortihm the dimensionality reduction step took place using algorithms like Principal Component Analysis (PCA) and Linear Discriminant Analysis (LDA) also the kernel form of PCA - KernelPCA. In the multiclass implementation the use of (KPCA) following with LDA was essential, in the binary classification we compare the use of PCA and KernelPCA. For experimental purposes Graph Spectral Analysis(IsoMap, LLE) was used for dimensionality reduction followed with Spectral Clustering in order to investigate subsets. 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. At the application level, a library for feature extraction and classification in Python will be developed. Credible publicly available resources will be used toward achieving our goal, such as KALDI. Comparisons will be made against [5-7].

[1]X. Huang, A. Acero, and H.-W. Hon, Spoken Language Processing. Up- per Saddle River, N.J.: Pearson Education-Prentice Hall, 2001.

[2] J. R. Deller, J. H. L. Hansen, and J. G. Proakis, Discrete-Time Pro- cessing of Speech Signals. New York, N.Y.: Wiley-IEEE, 1999.

[3] L. R. Rabiner and R. W. Schafer, Theory and Applications of Digital Speech Processing. Upper Saddle River, N.J.: Pearson Education- Prentice Hall, 2011.

[4] Wei-Ning Hsu, Yu Zhang, and James R. Glass, Unsupervised Domain Adaptation for Robust Speech Recognition via Variational Autoencoder- Based Data Augmentation,2017, http://arxiv.org/abs/1707.06265.

[5] C. Kotropoulos and G.R. Arce, ”Linear discriminant classifier with re- ject option for the detection of vocal fold paralysis and vocal fold edema”, EURASIP Advances in Signal Processing, vol. 2009, article ID 203790, 13 pages, 2009 (DOI:10.1155/2009/203790).

[6] E.Ziogas and C.Kotropoulos, ”Detection of vocal fold paralysis and edema using linear discriminant classifiers” in Proc. 4th Panhellenic Ar- tificial Intelligence Conf. (SETN-06), Heraklion, Greece, vol. LNAI 3966, pp. 454-464, May 19-20, 2006.

[7] M.Marinaki, C.Kotropoulos, I.Pitas, and N.Maglaveras, ”Automatic de- tection of vocal fold paralysis and edema” in Proc. 8th Int. Conf. Spoken Language Processing (INTERSPEECH 2004), Jeju, Korea, pp. 537-540, Oc- tober, 2004.

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