IBM和微软的SAPI TTS封装在ActiveX调用其他语言，…

该程序是非特定人语音识别程序，原来在SPCE500A下运行，能识别并应答16个命令，存储空间为128K字节。 //我们将其进行了修改，使其能在SPCE061A下运行，由于存储空间为32K字，只能识别并应答9个命令。 //在这个例程当中，我们示范了如何使用语音识别引擎对8个已经经过训练的命令进行识别。 //第一个命令 "Snoopy"，可以作为使能命令，也就是说使用者只能在该命令之后发出其他命令。 //例如，用户只有在发出 "Snoopy"命令并且听到响应"Snoopy here"之后才能继续下达其他命令 //如"How are you", "I love you",等等。 //当"Snoopy"命令响应之后，程序将等待8秒钟，如果没有其他命令发出， //系统将跳回到第0组，并且再次等待触发命令"Snoopy"。当用户的命令无法被识别时， //程序将响应为"Execuse me?", "I can t hear you!"或 "Say again!"这时用户可以再一次下达语音命令。-the procedures were non-specific voice recognition procedure, the original run in SPCE500A can identify and Response 16 orders, storage space for 128K bytes.// We had to amending it so that it can run in SPCE061A, storage space for 32K words, recognize only nine responses and orders.// This routine, we have demonstrated how to use voice recognition engine has eight pairs of trained orders for identification.// The first order "Jack Champion", as it can make an order, users can only say in the order issued after other orders.// For exampl

语音文件加窗后比较去零的范围，以及它们的比较图。-After the voice file comparison window to increase the scope of zero, and their comparison.

AppWizard has created this SRTest application for you.  This application not only demonstrates the basics of using the Microsoft Foundation classes but is also a starting point for writing your application. This file contains a summary of what you will find in each of the files that make up your SRTest application. SRTest.dsp     This file (the project file) contains information at the project level and     is used to build a single project or subproject. Other users can share the

语音信号的G.711编码。 The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the quantizer scale factor, y(k), is determined by "unlocked" factor yu(k) only. The factor yl(k) and speed-contral factor al(k) are ignored.-G.711 speech signal coding. The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the Quantizer scale factor, y (k), is determined by "unlocked" factor yu (k) only. The factor yl (k) and speed- al contral factor (k) are ignored.

c++平台开发的可以进行text to speech软件源代码 内有详细的使用安装说明 包括freephone和wxmbrola两个模块-c++ platform that can be text to speech software source code used within the detailed installation instructions, including freephone and wxmbrola two modules

本源码为语音识别中lpcc（线性预测倒谱系数）的计算方法。经试验，可用。为方便大家理解，关键语句和函数后面都加上了注释。-The source code for speech recognition in lpcc (Linear Prediction Cepstrum Coefficient) is calculated. After testing, can be used. To facilitate the understanding of everyone, the key behind the statements and functions are added to the Notes.

一个基于ITU-R recommendation BS.1387的客观语音质量评价算法源码。-Based on ITU-R recommendation BS.1387 objective voice quality assessment algorithm source.

C#设计高精度语音识别程序。内有演示程序。-C# design accurate speech recognition procedures. Within a demonstration program.

这个是实时采集语音信号，以及小波消噪的运行程序，我用过的，还算可以，但是小波消噪的那个程序有点问题，大家如果运行后，发现问题出在什么地方，谢谢告诉我。-This is a real-time voice signal acquisition and wavelet denoising operating procedures, I used that figure, Wavelet denoising but that the process is flawed, and if we run, it is found that the problem is in place, Thank you to tell me.

在这几十年里，数字信号处理技术取得了飞速发展，特别是字适应信号处理技术以其计算简单、收敛速度快等许多优点而广泛被使用。它通过起内部参数的最优化来自动改变其特性。自适应滤波算法在统计信号处理的许多应用中都是非常重要的。在这几十年里，数字信号处理技术取得了飞速发展，特别是字适应信号处理技术以其计算简单、收敛速度快等许多优点而广泛被使用。它通过起内部参数的最优化来自动改变其特性。自适应滤波算法在统计信号处理的许多应用中都是非常重要的。