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University of Edinburgh voice processing system that can find most of voice proc...

于 2022-10-13 发布 文件大小:1.89 MB
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英国爱丁堡大学的语音处理系统,可以找到大部分语音处理/识别/合成所需要的算法-University of Edinburgh voice processing system that can find most of voice processing/recognition/synthesis algorithms needed

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    简单的模式识别的分类算法(K_均值算法)程序的使用方法简单就不介绍了!-The classification of simple pattern recognition algorithm (K_ means algorithm) to use the procedure does not introduce a simple!
    2022-01-25 16:55:37下载
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  • A simple voice
    一个简单的基于微软语音开发包设计的语音识别程序,有注释-A simple voice-based Microsoft development kit designed for speech recognition procedures have Notes
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    lpc10语音压缩算法源码,应用于voip系统-Source code of lpc10 speech compress algorithm,used in voip system.
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    积分:1
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    积分:1
  • 本软件需要东进电话卡的支持 代码完全可靠
    本软件需要东进电话卡的支持 代码完全可靠-East into phone cards are supported entirely reliable code
    2022-02-28 19:24:45下载
    积分:1
  • LMS algorithm is extremely classic example of the hope that someone will allow m...
    经典LMS算法极其示例,希望有人能让我分享个多输入的LMS算法-LMS algorithm is extremely classic example of the hope that someone will allow me to share more than the LMS algorithm input
    2022-01-27 12:48:36下载
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    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.
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    积分:1
  • Indefinite long
    不定长信号合并,%执行信号相加: y(n) = x1(n)+x2(n) %调用方式:[y,n] = sigadd(x1,n1,x2,n2) % y:叠加后的信号序列的序列号,包括n1和n2 % x1: 第一个信号序列,序列号为n1 % x2:第二个信号序列,序列号为n2(n2可以与n1不同)-Indefinite long-Signal merger, add up to the implementation of signal: y (n) = x1 (n)+ X2 (n) call the way: [y, n] = sigadd (x1, n1, x2, n2) y: Superposition signal sequence after the serial number, including the n1 and n2 x1: the first signal sequence, the serial number for n1 x2: the second signal sequence, the serial number for n2 (n2 with n1 different)
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  • 高斯混合模型(GMM)是一种经典的说话人算法,这..
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