duoxinhaofenli

利用MATLAB来实现语音的分帧及语音端点检测(Using MATLAB to achieve voice segmentation and voice endpoint detection)

语音信号的特征提取，语音信号的Mel倒谱特征（MFCC）的求解方法，语音信号的线性预测原理以及LPC特征的求解方法 (Speech signal feature extraction, speech signal characteristics of Mel cepstrum (MFCC) the solution, voice signals, as well as the principle of linear prediction characteristics of the LPC method)

自相关函数的应用，进行基音检测，在MATLAB下进行仿真(Application of autocorrelation function for pitch detection, carried out in the MATLAB simulation)

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)

自适应滤波器设计的matlab源程序，包括LMS,RLS,Kalman滤波等(Adaptive filter design matlab source code, including the LMS, RLS, Kalman filtering)

观测实时模拟信号（语音）的频谱 用音频设备采集一段语音，将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的，应把它分段分析，而且实际运用中的数字信号处理的FFT的点数是有限的，一般只能达到千点。用傅里叶反变换IFFT，从频域恢复信号。画出频谱图和语音波形图。 (Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform.)

该文件为基于DTW的孤立词识别系统（特定人、小词汇量，未采用鲁棒语音识别技术）(The document DTW for isolated word recognition system based on (a specific person, small vocabulary, did not use robust speech recognition technology))

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)

源－目标说话人声音转换的学位论文：源－目标说话人声音转换则是指将源说话人的语音模式转换成某个指定的特定目标说话人的语音模式，即保持源说话人原有的语音信息内容不变，使转换后的语音具有目标说话人的声音特点。(Source- target speaker voice conversion refers to the source speaker voice patterns into a designated target speaker-specific speech patterns, that is, the source speaker to keep the original voice message unchanged, so that the converted speech with the goal of the speaker s voice characteristics.)

目前2.4kbps的混合激励线性预测 (MELP) 语音编码方法已经被确定为美国新的联邦语音编码标准.本文提出了一种改进的MELP语音编码方法,利用滤波器相似度和基于LPC系数分类的矢量量化技术,可以把MELP的码率降到1.7kbps以下,仍有较好的合成语音质量.(An Improved MELP Speech Coder)