lmsfilter

一个用matlab来实现的最小均方误差的信噪比和信干噪比(Using matlab to achieve the minimum mean square error, signal to noise ratio and signal to interference noise ratio )

lms alghoiritm method adaptive system

基于倒谱短时部分反映了语音的声道特性，先用汉明窗取一帧语音，然后经变换得到语音倒谱，将倒谱短时部分取出，进行正交反变换后将得到声道的对数谱，即得到语音频谱的包络。将频谱包络和频谱画在一张图上，有很好的对比效果。获取的包络效果十分好。(Based Cepstral partly reflects the short channel characteristics of the speech, first take a Hamming window with a frame of speech, and speech cepstrum obtained by converting the cepstrum short segment out inverse orthogonal transform to obtain channel will of the spectrum, i.e. to obtain the envelope of the speech spectrum. The spectral envelope and spectral painted on a chart, there is a good contrast. Get the envelope effect is very good.)

matlab代码，语音的压缩和编码，主要利用了心理声学模型，得到低比特编码 和高质量语音(matlab code, voice compression and coding, psychoacoustic model used by low-bit encoding and high-quality voice)

互联网著名国产软件小沁语音信使源代码。使用TAPI接口及语音Modem，实现语音电话自动拨打，语音录制及播放，广泛适用于企事业广告营销。(The use of TAPI interface and voice Modem, realize voice telephone automatic dial, voice recording and playback, widely applicable to the business of advertising and marketing. )

Speaker Identification using MFCC feature and VQ training, very clear MATLAB code

验证码小识别源码程序，结合易语言位图支持库，调用易语言模块实现验证码识别。(Little recognition source code program , combined with easy language bitmap support library , call easy language module verification code recognition .)

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.)

通常语音信号在增强时会出现混响现象，演讲者为了消除背景混响，不得不频繁地偏转头部的方向，这样会造成脉冲响应的不断改变。我们结合盲解卷法和频谱消去法来提高逆滤波器的滤波效果。我们利用输入语音信号间的相关系数矩阵计算出稳定、精确的室内脉冲响应的逆滤波器，而这些输入信号无需测量室内的脉冲响应就能被观测到。逆滤波能够消除早期的反射，这些反射包含混响中的绝大部分能量。之后，用频谱消去法来抑制逆滤波后的信号的尾部混响。本方法在实际适应性方面的表现通过具体的实验进行了验证，结果表明盲解卷法和频谱消去法的结合相较于单独使用一种方法，能够提供一个更优越的演说环境。(Usually voice signal will be enhanced when the reverberation phenomenon, the speaker in order to eliminate the background reverberation, and had to frequently deflect the direction of the head, this will cause the impulse response of changing. We combine the blind deconvolution and spectral elimination method to improve the filtering effect of the inverse filter. We use the correlation coefficient matrix between the input speech signal to calculate the stable, accurate inverse filter of the indoor impulse response, and these input signals do not need to measure the room impulse response can be observed. Inverse filtering to eliminate early reflections, these reflections contains most of the energy in the reverberation. After the spectrum elimination method to suppress the tail reverberation of the signal after the inverse filtering. Performance in the actual adaptation of the method by specific experimental validation results show that the blind deconvolution and spectrum to eliminate )

LD3320 是一颗基于非特定人语音识别（SI-ASR：Speaker-Independent Automatic Speech Recognition）技术的语音识别/声控芯片。提供了真正的单 芯片语音识别解决方案。 LD3320 芯片上集成了高精度的A/D 和D/A 接口，不再需要外接辅助的 Flash 和RAM，即可以实现语音识别/声控/人机对话功能。并且，识别的关键词 语列表是可以动态编辑的。(Voice Recognition ；Speech Recognition ；Speech recognition technology )