cubica算法performes

11阶自适应滤波器的LMS算法，并且与RLS进行了比较(11 steps LMS adaptive filter algorithm and the RLS were compared)

Chapter 2 Low Bit-Rate Speech Coding In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.(In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.)

房间脉冲响应函数IMAGE,用于计算房间混响时间的(Room impulse response function IMAGE, used to calculate the room reverberation time)

这是开源语音合成espeak的源代码，对研究TTS的同学很有帮助。(This source code is open source speech synthesizer espeak, helpful studies TTS classmates.)

有趣的语音识别程序，可以利用这个程序实现语音防盗门。是根据每个人不同的声调来实现的(interesting voice identification procedures, we can use this procedure voice security doors. Everyone is different tone to achieve the)

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))

这是语音信号处理中求倒谱的程序,我自己编的,很简单,实现了一段序列求倒谱的功能(This is the voice signal processing for cepstrum procedures, I developed, is very simple. achieve a sequence for the functional cepstrum)

1.单通道语音增强算法； 2.一种能够快速跟踪噪声变化的，基于噪声统计特性的噪声估计算法，结合谱减法进行消除噪声； 3.算法复杂度适中，可以满足实时性(1. Single-channel speech enhancement algorithm 2. A fast-track noise can change, based on the noise statistical properties of the noise estimation algorithm, combined with spectral subtraction to eliminate noise 3. Algorithm moderate complexity to meet the real-time)

对语音信号基频的提取对于语音信号的分析有一定作用(Basic frequency of speech Analysis )

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.)