fp_enhancement

基于单通道的在线次级通道建模的主动 噪声控制FXLMS算法的MATLAB实现( a single channel feed-forward active noise control system based on On-line secondary path modellingthe FxLMS alogrithms You can find many information about this simulation in "Use of random noise for online transducer estimate in an adaptive attenuation system " written by L,J Eriksson and M.A Allie in 1989.)

语音唤醒就是在设备待机状态下，用户说出特定的语音指令（唤醒词）使设备进入工作状态或完成某一操作。设备为了实现语音唤醒功能，就需要设备实时监听，也就是实时的录音并分析有没有唤醒词来唤醒设备。评价一个语音唤醒系统的标准有三个：唤醒正确率、误报率、功耗。一般情况下，唤醒正确率越高 ，误报率也越高。好的系统就需要唤醒率高，误报率低，功耗低。(Speech wakeup is when the device is in standby mode, the user speaks a particular voice command (wake up word), so that the device enters the work state or completes an operation. In order to realize the function of speech wakeup, the device needs real-time monitoring, that is, real-time recording and analyzing whether or not wake words are used to wake up the equipment. There are three criteria for evaluating a voice wakeup system: wakeup accuracy, false positive rate, and power consumption. Generally, the higher the correct rate of arousal, the higher the false positive rate. A good system requires high wake-up rate, low false positive rate and low power consumption.)

预白化是对语音信号在处理之前的一中预处理操作，对其后的操作有很重要的作用()

基于svm分类器的汉语声调识别，实现了对声调四声的识别，纯净语音识别率100 ，噪声环境下识别率90 (The svm classifier based Chinese tone recognition, the four tones of tone recognition, the pure voice recognition rate of 100 , 90 recognition rate in noisy environments)

实现语音识别，使用HTK工具。README描述了具体的训练过程。(Voice recognition, using HTK tools. README describes the specific training process.)

数字语音处理及MATLAB仿真_张雪英编著第二版 源码(Digital audio processing and MATLAB simulation _ Zhang Xueying edited version second source code)

C# 对接 Java 3des md5WithRsa, 有需要的可以参考下(If you need C # to dock Java 3des md5WithRsa, you can refer to the following)

实验示例是基于语音中的mfcc，语音倒谱特征来进行聚类，先利用训练样本来计算训练样本聚类中心（用到了lbg算法），之后再进行分类。 注意：使用代码时需要自己更改文件路径。(This example is based on the MFCC in speech and the feature of speech Cepstrum to cluster. First, the training sample is used to calculate the training sample clustering center (using the LBG algorithm), then the classification is then carried out.)

鲁棒数字音频盲水印水印置乱算法实现基于阿诺德变换的离散小波变换，离散余弦变换(Robust Blind Digital Audio Watermarking Implementation Based on Watermark Scrambling Algorithm-Arnold Transform Discrete Wavelet Transform,Discrete Cosine Transform,Erro Correcting Code)

提取语音信号的lpc参数并进行时间归整，需要将wav语音文件放在指定目录下‘e:yyzl’(voice signal from the lpc parameters and time consolidation, need to wav sound files on the specified directory 'e : yyzl')