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WiMax Matlab Simulation

卡尔曼滤波和扩展卡尔曼滤波Matlab源程序，广泛用于参数估计，滤波去噪，惯性导航等应用方面。(Kalman filter and extended Kalman filtering Matlab source, widely used in parameter estimation, filtering noise, inertial navigation and other applications.)

constraint nonlinear optimization

these code implement convolutional coder and decoder used in LTE mobiles. this code used in 4G mobiles release 8.

noise cancelation using matlab, select your audio file

此程序实现图象增强的功能，采用直方图均衡化的方法来实现图像增强。(This procedure realize the function of image enhancement using histogram equalization method to achieve image enhancement.)

详细的unscented particle filter 程序(Unscented particle filter detailed procedures)

Wigner-Ville 分布(简称 WVD)是典型的二次型变换，它定义为信号瞬时相关函数的傅立 叶变换。 由于在计算中不加窗操作，它避免了时域分辨率和频域分辨率之间的相互牵制。 已证明，对于单分量线性调频信号，WVD 的时频表示能量集中性最优。但也正因为不涉 及窗函数，在分析多分量信号时 WVD 会受到交叉项的干扰。(The Wigner-Ville distribution (WVD) is a typical quadratic transformation, which is defined as the Fourier of the instantaneous correlation function of a signal. Ye Bianhuan. Because of no window operation in the calculation, it avoids the mutual restraint between time domain resolution and frequency domain resolution. It has been proved that for the single component LFM signal, the WVD time frequency representation has the best energy concentration. But it is also because it does not involve. And window functions, when WVD is used to analyze multi-component signals, it will be interfered by cross terms.)

基于matlab的ofdm系统仿真，通过仿真实现了ls 和LMMSE准则的信道估计，并仿真比较了两种方法的性能(Matlab ofdm systems based on the simulation, the simulation criteria to achieve the ls and LMMSE channel estimation, and simulation performance of the two methods compared)

Abstract— We propose a simple speech compression algorithm using subband division and ADPCM (Adaptive Differential Pulse Code Modulation) algorithm. Although speech data are stored in a semiconductor memory device, its capacity and the available network capacity are limited. Therefore, it is necessary to compress the data as much as possible. The first is that the reconstructed datum must be understood correctly. Speech signal is compressed with a ratio of 1/4 using the proposed subband division and ADPCM (Adaptive Differential Pulse Code Modulation) algorithm.