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精通MATLAB综合辅导与指南例程,适合初学者使用(Proficient in MATLAB Guide Comprehensive Counseling and routines, suitable for beginners to use)

说明：  地统计学中基于空间变异原理的差值分析，做空间分析的好材料。matlab实现(To statistical theory based on spatial variability of the difference analysis, a good material to do spatial analysis. matlab implementation)

moteur asynchrone simulation mli

文中介绍了传统的PID控制原理，PID分为位置式和增量式，计算机控制是数字PID控制。一些 改进的PID控制算法是针对实际应用中的不足提出的，如积分分离式PID控制算法，抗 积分饱和算法，变速积分算法等，实际应用中，控制又分为简单PID控制和串级PID控 制，对一些典型的控制算法，选用典型控制对象用MArLAB软件进行了对应仿真。对仿 真结果进行了对比分析，说明了改进算法的作用。(This paper introduces the principle of the traditional PID control, PID into position and incremental, computer-controlled digital PID control. Some improvements in the PID control algorithm is less than the practical application of the proposed, such as discrete integral PID control algorithm, anti-integral saturation algorithm, changing integration algorithm and so on, practical application, the control is divided into a simple cascade PID control and PID control , for some typical control algorithm, selection of a typical control object MArLAB software with the corresponding simulation. The simulation results of comparative analysis carried out to illustrate the role of the improved algorithm.)

number of Bayesian networks

书籍，介绍matlab与dsp，阐述了数字信号处理原理，例程丰富(Books, introduction matlab and dsp, digital signal processing principles expounded rich routines)

dising fir filter in the fdatool matlab

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 )

a digital compression algoritm in matlab

it is the code for wiener filter