speech recognition with DTW

本次课程设计主要介绍了语音信号的录制、语音信号的采集与分析、语音信号的采样、语音信号的频谱分析、加噪后的频谱分析、滤波器的设计以及对于语音信号的回放等知识。通过 Cool Edit录制一段语音信号，运用MATLAB对其进行时域与频域分析，然后设计滤波器加入噪声进行滤波处理，比较滤波前后始于波形或频谱的变化，最后实现对于语音的回放设计。(Digital voice processing filter design)

近场的MUSIC算法，可用于近场噪声源识别(Near-field MUSIC algorithm can be used for near-field noise source identification)

提取MFCC的matlab代码，用于提取MFCC-梅尔倒普系数(MFCC extraction of matlab code, for the extraction of MFCC-Mel Pu inverted coefficient)

语音识别中的两种特征提取方法lpcc和mfcc，还有一个是文本无关的识别算法dtw，另外还有一个是预处理消噪部分的。共享一下，这些都是我调试过的，好用。(Speech Recognition two feature extraction methods and mfcc lpcc. There is a text-independent recognition algorithm dtw, in addition to a pretreatment is part of the noise source. Sharing that those are off the debugging and ease of use.)

使用TD-PSOLA（td-psola、psola）算法，做了一个语音合成，实现语音声调的变调，时长的变化等功能。参考praat源码，提取出其合成方法，效果不错。需要提供pitchtier、durationtier、pointprocess等准换信息文本（可由praat产生）供readfile函数读入(Use TD-PSOLA (TD-PSOLA, PSOLA algorithm, made a speech synthesis, speech tone tone sandhi, long-term changes in the function. Reference Praat source code, extract its synthetic method, the effect of the same.)

做毕设是用到的gmm的matlab程序，测试过的 ，绝对可以用(Complete set is used to do the matlab program gmm, tested, can surely be)

利用DTW模版匹配算法实现0~9十个数字的识别。(use DTW template matching algorithm 0-9 10-digit identification.)

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')

文字转语音C#源码，利用windows speech.lib 实现文本转语音 可设置rate 保存为wav文件格式。(Text to voice C# source code, using windows speech. lib to achieve text to voice can be set rate saved as WAV file format.)

本文介绍了基于麦克风阵列的声源定位算法，给初学者有很大的帮助(This article describes the microphone array based sound source localization algorithm, a great help for beginners)