histgram

各种各样的不同代码源打包，都是经过运行的，绝对可以使用。(A variety of different source code packages are run through, and absolutely can be used.)

simple edge color coding

本人自己做的MATLAB工程文件，实现整流逆变(I own MATLAB project file, realize rectifier inverter)

基于Markov方法的风机出力预测方法，具有较强的可转换性(The method of wind power forecast using Markov)

步态识别 matlab编程包括建立路径、生成训练库、视频帧提取、格式归一化、生成步态能量图进行识别(Gait recognition matlab programming)

用matlab做的一个模拟时钟，参数可调，可用于制作时钟界面(Use matlab to do an analog clock, adjustable parameters, can be used to make the clock interface)

The simulation FBMC with AWGN channel

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 )

对fsk 信号phase 进行仿真，提取信息 对目标速度，距离进行计算(simulation for fsk signal)

通过设计一个二阶加权系数自适应横向FIR滤波器，对一个加随机噪声的正弦信号实现滤波。 具体设计方案为： 1，生成标准正弦信号S 2，生成等长的随机信号N 3，生成加随机噪声的正弦信号X 4，X通过参数可调数字滤波器，输出Y 5，Y与参考信号作差得到误差E 6，E通过自适应算法调整权值W 7，用LMS算法处理噪声干扰的信号，最终实现滤波器功能(A two order weighted coefficient adaptive transverse FIR filter is designed to filter a sinusoidal signal with random noise. The specific design scheme is as follows: 1, generating a standard sinusoidal signal S 2, generating an equal length random signal N 3, generating a sinusoidal signal X with random noise 4, X can adjust the digital filter by parameter and output Y 5, the difference between the Y and the reference signal is E 6, E adjusts the weight W through the adaptive algorithm 7, the LMS algorithm is used to deal with the noise interference signal, and the filter function is finally realized.)