Speech-enhancement

这是一个基于DTW的语音识别的代码，里面包含端点检测和MFCC的提取。(This is a DTW-based speech recognition code, which contains end point detection and the extraction of MFCC.)

实现已知训练序列的矢量量化器（LBG）算法。初始码书从训练序列每隔五个样本选取一组(Achieve a known training sequence of the vector quantizer (LBG) algorithm. The initial codebook from the training sequence every five samples to select a group of)

利用自适应滤波器对心音信号进行预测，包括了LMS算法，归一化LMS，变步长LMS和RLS算法，其中RLS的预测效果最好.(Heart sound signals using adaptive filter to predict, including the LMS algorithm, the normalized LMS, variable step size LMS and RLS algorithm, in which the best forecast of RLS.)

将wav格式文件转化为txt格式的程序,用于读入语音文件转换TXT格式，便于程序分析(Wav format files into txt format program)

声源定位程序，语音信号与房间脉冲响应进行卷积得到混响语音信号，再通过互相关时延估计得到定位(Sound source localization procedure, the speech signal and the room impulse response are convoluted to get the reverberation speech signal, and then through the correlation time delay estimation to obtain the localization)

C#的WinForm项目的dataGridView控件的CheckBox(The CheckBox of the dataGridView control for the WinForm project of the C #)

基于HMM的语音识别算法和毕设例程，包括数据。(HMM-based speech recognition algorithm and complete routine, including data.)

语音信号处理课件：语音基础知识、短时时域分析、频域分析、倒谱域分析、线性预测技术、语音增强、语音识别(Speech signal processing courseware: basic knowledge of speech, short-term time domain analysis, frequency domain analysis, cepstrum domain analysis, linear prediction technology, speech enhancement, speech recognition)

在evmdm642平台下的一个语音采集-回放任务，用48K/秒的采样率采集立体声音频输入，用给定的FIR 滤波参数处理数据后再播放出来。注：内附个人整理的详细操作文档。(evmdm642 platform in a speech to the gathering-intervals task with 48 K/s sampling rate Acquisition stereo audio input, used to the FIR filter parameters for data processing later. Note : with a detailed personal finishing operations documentation.)

语音信号分帧后，计算每帧信号的短时平均能量(After the speech signal is framed, the short-term average energy of each frame signal is calculated)