PROGRAMS-USING-MATLAB

]LaTeX Tutorials - A Primer - TUG latex最具权威的辅导资料(] LaTeX Tutorials- A Primer- TUG latex most authoritative Tutorials)

神经网络中用矩阵分块的方法处理数据量过大的情况，效果很好(Neural network is used in the method of matrix blocking deal with the situation of the large amount of data, the effect is very good )

说明：  提取滤波器耦合矩阵:利用特征值和T向量构造M矩阵，并进行相似变换求得不同结构的滤波器耦合矩阵(Extract filter coupling matrix: use eigenvalues and T vectors to construct M matrix, and perform similar transformations to obtain filter coupling matrices with different structures)

matlab stateflow 入门教程，里面有详细的例子介绍每一个功能如何使用，非常适合初学者。(matlab stateflow Getting Started Guide, which describes detailed examples of how to use each function, very suitable for beginners.)

compressive sensing ALGORITHIM CODE FOR MATLAB

将MATLAB源文件编译成可执行程序的步骤(MATLAB source files will be compiled into an executable program steps)

学习ekf的一个源码，对初学者非常有用，随便下载并交流(Extended Kalman Filter Demo )

说明：  异步电机混合坐标系模型异步电机混合坐标系模型异步电机混合坐标系模型异步电机混合坐标系模型异步电机混合坐标系模型(Hybrid coordinate model for induction motor)

说明：  小波去噪指标-信噪比，用这个程序可以实现(Wavelet Denoising indicator- signal to noise ratio can be achieved with this procedure)

实现功能 1) 在用户界面设置录音时长并实时录音 2) 调节放音幅度、播放原声并进行时域、频域绘图 3) 对收集的声音进行倒放处理并播放 4) 对收集声音进行回声处理并播放，比对时域波形 5) 对收集声音进行回响处理并播放，比对时域波形 6) 对经过回声处理的声音进行回音消除并比对时域波形 7) 采用自适应滤波器消除采集声音中的环境噪音 8) 设计相应的滤波器组成均衡器对收集的声音进行处理并播放 9) 男女声识别 10) 设计参数可调的滤波器对采集的声音进行处理 (1) in the user interface to set the recording length and real-time recording 2) to adjust sound, played acoustic amplitude and time domain, frequency domain mapping 3) to upend the collection of sound and play 4) to collect sound echo processing and playback, than in the time domain waveform 5) to collect sound echoes, processing and playback, than in the time domain waveform 6) to echo and treated with echo s voice than in the time domain waveform 7) using adaptive filter to eliminate ambient noise in acquisition voice 8) to design the corresponding filter to process the voice of equalizer to collect and display 9) NanNvSheng recognition 10) tunable filter design parameters for voice processing )