fft2hps

hmm的工具箱，里面有很全面的hmm的matlab程序包，适用于各种环境使用(hmm the toolbox, which has a very comprehensive hmm the matlab package, applicable to a variety of environmental use)

信息融合方面得例题，对已知要求的目标，利用动态矩阵控制（DMC）使其达到期望值。所有程序均已通过调试。希望对大家有所帮助(Example of information integration have on the known requirements of the target, the use of Dynamic Matrix Control (DMC) to reach the expectations. All have been through the debugging process. I hope all of you to help)

在平台上找到的滑动/移动DCTII算法，都是用不了的，尤其是用于LMS，根本不适用，查了国内外论文，遍了程序，供大家参考(In PUDN,most DCT programm are not fit to LMS adaptive filter. with several literatures, this programm is finished for LMS filter.)

绘制语音信号的 WVD 分布，对时频工具箱中的 tfrwv 及 tfrspwv 函数的使用进行了详细说明(The way of drawing speech signals WVD was shown in the m file, and corresponding functions tfrwv and tfrspwv were discussed thoroughly.)

（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)

计算语音、音频信号的信噪比程序（包含音频样本文件）(Calculation of voice, audio program signal to noise ratio (including the audio sample file))

Active Feedforward Narrowband Noise Cancellation using Matlab and Simulink

根据房间配置信息生成房间传函，进而生成模拟语音数据(Depending on the room configuration generates a room transfer function, thereby generating an analog voice data)

关于如何使用fpga实现语音识别 最新版本(err)

语音编码中对LSF系数进行VQ处理，LBG方法生成VQ码书，且采用了新的失真测度(Speech Coding for VQ processing LSF coefficients, LBG method of VQ codebook generation, and adopted a new measure of distortion)