语音信号处理pdf 赵力

ITU P.563 语音质量评价，能只送入失真信信号，而不需要参考信号，便能评价失真信号的MOS分 (ITU P.563 voice quality assessment can only be sent to the distortion of the letter signals without the need for a reference signal will be able to evaluate distortion signal MOS score)

对语音信号进行脉冲编码调制，采用A律13折线(Voice signal pulse coding modulation, using A Law 13 polyline)

本软件在vs2010平台下实现了语音合成（Text to Speech）功能，实现将输入的文字信息转化为声音信息，同时能够选择系统语音库，调节朗读音量以及播放频率等详细参数，同时提供了编译过的exe测试程序，并可以选择测试文本信息。

iLBC Speech Coder ANSI-C Source Code WebRtcIlbcfix_CbSearch.h.

基于LPC的语音合成，可以实现变调变速功能，附带注释便于学习。(LPC-based speech synthesis, can achieve modulation speed function, annotated easy to learn.)

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 )

一种加权K近邻语音情感识别程序，包括部分柏林语音情感库(A Weighting K-nearest neighbor speech emotion recognition program, including some libraries Berlin Speech Emotion)

　介绍了一种针对语音增值服务开发的智能语音终端,该终端以EM78806A单片机为核心研发,借助公共电话网络,实 现了电话与局端语音信箱中心的通信与智能化控制。文中主要对智能语音终端的硬件结构、工作原理及程序设计方法进行 了论述。(Introduce a value-added voice services for the development of intelligent voice terminal, the terminal to EM78806A Singlechip as the core research and development, through the public telephone network, the realization of a telephone central office voice mail with the Center for Communications and intelligent control. The main text of the intelligent voice terminal hardware structure, working principle and procedure of design methods are discussed.)

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.)

语音信号的盲源分离算法研究及应用，本程序利用matlab实现(Blind Source Separation Algorithm Research and Application of the voice signal, the program using matlab realize)