ETSI发布的用于研究分布式语音识别的前端降噪算法

通过ACTIVEX控件与MATLAB进行双向通信实现实时语音端点检测-through ACTIVEX control and MATLAB for two-way real-time voice communication endpoint detection

hss speaker system for direct target voice transmission

程序名：Guard.c //该程序是特定人语音识别 //在程序中通过三条语句的训练演示特定人连续音识别，其中第一条语句为触发命令。 //在程序开始首先训练触发命令，然后提示训练两条命令，训练完毕开始语音识别。 //当识别出触发名称后，发布命令则会听到应答-procedures Name : Guard.c// The procedure is a specific person voice recognition// procedures adopted by the three phrases of training people for a specific demonstration of the sound recognition, the first of which triggered an order for the statement.// First in the beginning of the proceedings training trigger orders, and then suggested training two orders completion of the training started to voice recognition.// Trigger out when name recognition, will be issued an order to hear responses

Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis.

该函数可以对语音信号进行加窗和分帧,简洁有效-enframe for speech signals

我是做语音识别的，这是我做的一个关于端点检测的小程序，里面还有一个工具包-I have to do speech recognition, which is what I am doing on a small endpoint detection procedures, there is also a tool kits

将IBM TTS 和微软的SAPI封装在ACTIVEX中供其他语言调用,例子指写了用WEB网页如何调用的方式-The IBM TTS and Microsoft SAPI packaged in ACTIVEX call in for other languages, for example, refers to the use of WEB pages to write how to call the way

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

g.729协议源代码,c语言,直接可以应用(dsp定点化)

语音识别与合成系统. 里面有详细的介绍与程序-Speech Recognition and Synthesis System. There are a detailed introduction and procedures