Description: 自己编写GUI界面实现语音增强，可在main.c中点击菜单debug中的run便可以运行程序，可分别实现谱相减、最小均方和维纳滤波语音增强-GUI interface to prepare themselves to achieve enhanced voice, in main.c which debug menu click the run will be operating procedures, can be achieved spectral subtraction, both the smallest and the Wiener filter speech enhancement Platform: |
Size: 97280 |
Author:Richard |
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Description: This paper deals with the problem of speech enhancement when a
corrupted speech signal with an additive colored noise is the only
information available for processing. Kalman filtering is known as
an effective speech enhancement technique, in which speech signal
is usually modeled as autoregressive (AR) process and represented
in the state-space domain.-This paper deals with the problem of speech enhancement when a corrupted speech signal wit h an additive colored noise is the only informat ion available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) process and represented in th e state-space domain. Platform: |
Size: 102400 |
Author:rifer |
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Description: 一段语音，分别在白噪声、粉色噪声、工厂噪声、冲击噪声和信噪比变化很大的语音，用于检验语音端点、语音增强的方法的对与错。-a voice, in white noise, pink noise, factory noise, the impact of noise and signal-to-noise ratio of the great changes of voice, used to test the voice endpoint, speech enhancement of the right and wrong. Platform: |
Size: 545792 |
Author:李金宝 |
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Description: 这是一个频减法语音增强算法的matlab代码，在matlab中运行成功的-This is a frequency subtraction speech enhancement algorithms Matlab code in Matlab run successful Platform: |
Size: 1024 |
Author:怀坏 |
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Description: mmse改进的算法，关于语音增强的更进一步的算法-MMSE improved algorithm, with regard to the further speech enhancement algorithm Platform: |
Size: 3489792 |
Author:rch |
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Description: 这里是几篇关于语音增强中有声无声估计的VAD算法的文献，希望和大家一起讨论，做出与之相关的matlab源代码。-Here are a few on the speech enhancement in a silent voice has estimated VAD algorithm literature, hope and everyone together to discuss and make related matlab source code. Platform: |
Size: 1731584 |
Author:rube |
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Description: 这是本人编写的一个利用维纳滤波实现语音增强的程序.在输入语音信噪比不是很低的情况下,效果不错.-This is one I prepared realize the use of Wiener filter speech enhancement procedure. In the input voice signal to noise ratio is not very low circumstances, good results. Platform: |
Size: 1024 |
Author:李茉 |
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Description: 基于自适应子带频谱熵的稳健性语音端点检测
可用于语音增强及端点检测-Based on adaptive sub-band spectrum of the robustness of Entropy voice activity detection can be used for speech enhancement and endpoint detection Platform: |
Size: 1024 |
Author:翟景瞳 |
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Description: 该程序是卡曼滤波法在语音处理上的应用,能有效的去除噪声,达到语音增强的目的!-The program is Kaman filtering method in the voice processing application, can effectively remove the noise, to achieve the purpose of speech enhancement! Platform: |
Size: 1024 |
Author:zhjuna |
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Description: 争对移动声源采用波束形成进行语音增强，提出一种约束子带波束形成算法。其波束形成器基于一个软约束，其目的是要使波束指向特定的区域即声源方向。而其核心在于首先要进行声源定位，获得尽量准确的方位信息，然后构造软约束条件，用于波束形成。且在此过程中不断跟踪声源的移动情况。在构造的约束条件中，需要知道声源的二维信息，即与麦克风阵列的距离和方向角，“软”体现在对距离和方向角的确定都是在一定范围内的，有待进一步更正。-Between the mobile sound source using beamforming for speech enhancement, a constrained subband beamforming algorithm. Its beamformer based on a soft constraints, the aim is to make the beam point to a specific region that is the direction of the sound source. And its core is the first to sound source location, access to location information as accurately as possible, and then construct the conditions of soft constraints for beamforming. And in the process continuously tracking the movement of a sound source. Structure at the binding conditions, need to know the two-dimensional sound source information, that is, with the microphone array the distance and direction angle, the " soft" embodied in the distance and direction angle of the determination are within the scope must be further corrected . Platform: |
Size: 128000 |
Author:chen |
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Description: 语音增强是语音信号处理的重要分支，本程序用小波变换实现-Speech enhancement speech signal processing is an important branch, the procedures for implementation of Wavelet Transform Platform: |
Size: 1024 |
Author:bromheaven |
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Description: 本文介绍了一种基于自适应滤波算法的语音信号增强处理的方法。文中在扼要介绍了目前常用的语音增强方法的基础上，重点介绍了采用LMS算法的自适应语音增强系统。对三种LMS算法（基本LMS算法,符号误差LMS算法及NLMS算法）进行仔细研究，并用Matlab仿真实现，验证比较了三种算法的语音增强效果。-In this paper, a system of speech enhancement is discussed based on self-adaptive algorithm. On the basis of presenting an overview of a number of speech enhancement is described in detail based LMS algorithm.This paper researches sorts of LMS algorithms（the basic LMS algorithm,the sign error LMS algorithm and NLMS algorithm） in detail, and compares the three algorithm of speech enhancement effect with Matlab simulation validation. Platform: |
Size: 384000 |
Author:雷妮 |
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Description: The speech signal for the particular isolated word can be viewed as the one generated using the sequential generating probabilistic model known as hidden Markov
model (HMM). Consider there are n states in the HMM. The particular isolated
speech signal is divided into finite number of frames. Every frame of the speech
signal is assumed to be generated from any one of the n states. Each state is modeled as the multivariate Gaussian density function with the specified mean vector
and the covariance matrix. Let the speech segment for the particular isolated word
is represented as vector S. The vector S is divided into finite number of frames
(say M). The i th frame is represented as Si . Every frame is generated by any of the n
states with the specified probability computed using the corresponding multivariate
Gaussian density model. Platform: |
Size: 787456 |
Author:Khan17
|
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