speech-signal-processing

（附带有文档，文档中有方案，有程序有结果，设计完后对一段语音进行处理）给定滤波器的规一化性能指标（参考指标,实际中依据每个同学所叠加噪声情况而定）通带截止频率wp=0.25*pi。 采用窗函数法设计低通、高通、带通型FIR滤波器，对叠加噪声前后的语音信号进行滤波处理，绘出滤波器的频域响应，绘出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；在相同的性能指标下比较各方法的滤波效果，并从理论上进行分析((With the document, document, program, program results, the design after a voice) to set the filter rules of performance indicators (reference, practice, each student adding noise may be) passband cutoff frequency wp = 0.25* pi. Window function method to design low pass, high pass, band-pass FIR filter, filtering the voice signal before and after adding noise, draw the filter frequency response plot the time-domain waveform and spectrum of the filtered signal, and filtering the signal before and after comparison and analysis of signal changes the filtering effect of the various methods in the same performance indicators, and the theoretical analysis)

程序功能可以基本上实现基于C语言的G.729A语音算法的解码功能，添加了部分本人的注释和理解，便于读者理解(Program features can basically C language-based decoding algorithm G.729A voice, add some of my notes and understanding, easier to understand)

基于麦克风阵列的声源定位一维music算法源代码(Based on the microphone array beamformer one-dimensional music algorithm source code)

Speech PROcessing tools. First edit the makefile to match your needs

用vq算法进行说话人识别，能达到良好的效果(Using VQ speaker recognition algorithm can achieve good results)

谱减法自适应滤波以及听觉掩蔽效应00865833(Spectral subtraction )

语音压缩感知，观测矩阵采用混沌序列，稀疏字典采用KSVD训练得到。(compressed sensing with chaotic sequence and K-SVD )

男女变声器，语言信号处理，录音，matlab，各种滤波器，有预处理，自定义时长录音并保存，自主选择文件，自主制作界面(Male and female voice changer, speech signal processing, audio, MATLAB, various filters, pretreatment, long recording and save custom, choose file, independent production interface)

这 里主要对LMS算法及一些改进的LMS算法（NLMS算法、变步长LMS算法、变换域LMS算法）之间的不同点进行了比较，，在传统的LMS算法的基础上发 展了LMS算法的应用。另一方面又从RLS算法的分析析中对其与LMS算法的不同特性进行了比较。 (Here the main difference between the LMS algorithm and improved LMS algorithm (NLMS algorithm, variable step size LMS algorithm, the transform domain LMS algorithm) comparison, the traditional LMS algorithm based on the development of the application of the LMS algorithm . On the other hand and from its different characteristics of the LMS algorithm of the analytical analysis of the RLS algorithm.)

对语音信号进行脉冲编码调制，采用A律13折线(Voice signal pulse coding modulation, using A Law 13 polyline)