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　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.)

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.)

matlab程序，从语音中提取lpcc，可以修改维数，带详细中文解释，方便阅读。(matlab program, extracted from voice lpcc, you can modify the dimension, with a detailed explanation of the Chinese, and easy to read.)

语音信号的时域频域分析，从短时能量到语谱图，以及线性预测参数和梅尔倒谱系数等等(Speech signal in time domain frequency domain analysis, from the short-term energy to the spectrogram, and the linear prediction parameters and the Mel cepstral coefficients, etc.)

the document explains speech recognition using mel frequency cpstrum coeefficient and noise reduction method

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.)

这是一个语音的文件的一个FFT转换文件,是用C++写的源程序,相信对大家有用(This is a voice of a paper document conversion FFT is written in C source files, it may be useful to think)

语音压缩和回声消除，支持LINUX和其他平台(voip echo cancellation)

C# 对接 Java 3des md5WithRsa, 有需要的可以参考下(If you need C # to dock Java 3des md5WithRsa, you can refer to the following)

语音处理的实用代码，完成耳语音的声韵分割功能(Practical speech processing code, complete the ear speech sound segmentation function)