SpecSubTD_Boll79

可对一段信号进行分帧加窗，利用过零率和短时能量检测语音端点(It can divide the frame into a window and use the zero crossing rate and short time energy to detect speech endpoints.)

在平台上找到的滑动/移动DCTII算法，都是用不了的，尤其是用于LMS，根本不适用，查了国内外论文，遍了程序，供大家参考(In PUDN,most DCT programm are not fit to LMS adaptive filter. with several literatures, this programm is finished for LMS filter.)

语音信号处理，赵力，对语音信号处理有一个基本介绍(the book of speech signal processing)

采用加权网络中节点强度和权重都是幂率分布的模型，计算两个矩阵之间的欧氏距离，实现了对10个数字音的识别。( Using weighted model nodes in the network strength and weight are power law distribution, Calculation of the Euclidean distance between the two matrices, To achieve the recognition of 10 digital sound.)

数字语音处理及MATLAB仿真_张雪英编著第二版 源码(Digital audio processing and MATLAB simulation _ Zhang Xueying edited version second source code)

一个LMS自适应信号增强。输入信号是一个正弦 波将白噪声。 详细介绍了程序源代码(An LMS adaptive signal enhancement. The input signal is a sine wave with added white noise. The detailed description is in the program source code)

本文主要研究普通计算环境下语音自动识别(ASR)技术的软件实时实现，基于语音识别的基本原理和过程。在VC环境下建立了一套语音识别系统及其演示软件。实验表明，该系统对特定人情况有较高的识别率和一定的实时性(In this paper, under the computing environment on general automatic speech recognition (ASR) technology to achieve real-time software, speech recognition based on the basic principles and processes. VC environment in the establishment of a voice recognition system and its presentation software. The experiments show that the systems are in particular have a high recognition rate and a certain degree of real-time)

语音信号去噪，在MATLAB开发环境下，对含噪声的语音信号进行卡尔曼滤波，实现语音增强的功能。(The voice signal denoising, the Kalman filter, the noisy speech signal in the MATLAB development environment Wave, speech enhancement.)

characteristic parameters for speech recognition on the extraction function

块和子带自适应滤波器matlab源文件，内有详细说明及运行结果图(Block and Subband Adaptive Filters)