一种基于语音频谱的基频和共振峰提取算法

基音频率和共振峰频率的提取在语音编码、语音合成和语音识别中有着广泛的应用。通过深入分析语音信号的时域和频域性质,针对语音信号幅度谱的特征设计了一种有效的基频和共振峰提取算法。并对实际语音信号进行参数提取测试,实验结果证明了这种算法能够准确提取不同讲话者和录音条件下的语音信号的基频与共振峰频率。     

语音信号的混沌遮掩及其正定盲提取算法

利用小波变换提取语音信号的能量聚集带,将其隐藏在混沌载体信号中进行传输,设计一种盲提取算法实现不同混沌动力学系统下语音信号的有效提取。以3种不同维数的混沌动力学系统为背景,仿真实验定性和定量地分析了所提出算法的性能,验证了噪声环境下算法的可靠性,证明盲提取算法可作为对混沌保密通信系统保密性验证的有效方法。     

基于语音频谱的共振峰声码器实现

共振峰声码器因其在理论上具有最低码率而一直是参数语音编码算法研究的重点。共振峰编码器的关键算法是基频和共振峰等语音参数的提取。在高分辨率语谱图基础上,利用语音信号的频域特性设计了一种简单有效的基频和共振峰提取算法。通过评价重建语音信号的音质,证明了参数提取算法的准确性。根据语音实验确定编码参数包含基频和前4个共振峰,并在保证语音质量的前提下制定各参数的量化指标。应用实际语音信号对算法的性能进行测试,试验结果证明算法在码率为1 400 b/s时具有良好的语音质量。     

基于非负约束矩阵分解的语音增强算法研究

从噪声背景中提取尽可能纯净的语音信号，增强有用信号，抑制、降低噪声干扰的技术称为语音增强技术。语音增强有着广泛的应用，因此寻求一种有效的算法对带噪语音信号进行处理得到较纯净的原始语音信号的研究有着很大的意义。多年来很多经典的语音增强算法被提出，如谱减算法，子空间算法等。文章提出了一种新颖的语音增强方法，即基于非负低秩稀疏分解的原理在强噪声环境下实现语音增强。把语音信号和噪声信号看做是一个非负低秩稀疏分解问题并且不断的优化算法分离出语音信号和噪声信号的幅度谱。实验结果表明在强噪声环境下这种方法对比一些传统的语音增强方法效果更好，具有更少的噪声残余与较低的语音失真等优点     

基于分频带自相关函数的混叠语音基频分离提取新算法

混叠语音的基频分离提取问题是听觉场景分析系统的重要一环。以往的分频带自相关函数的混叠语音基频分离提取方法都是基于频带只受混叠信号之一支配的假设,而事实上,频带常常同时受两个信号影响,为此,本文提出了一种混叠语音基频分离提取新算法,算法在寻找可能的频带组时采用了闭环自适应频带选取模块,根据频带组的基频及其周期度确定两个潜在基频,提高了搜索潜在基频的鲁棒性;利用两个潜在基频重新判断频带的归属来分离信号提取基频,提高了提取基频的精度。实验结果证明新算法具有较高的有效基频提取精度。     

采用子带长时信号变化特征的VAD检测

为提高语音活动检测(VAD)在低信噪比下的准确率,提出了一种基于子带长时信号变化特征的VAD算法。将语音信号转换到频域,并分解为几个不重复的子频带,对这些子带信号分别提取长时信号变化特征,然后采用GMM在线建立语音和非语音模型,以模型的似然比进行VAD判决。实验结果表明,算法在较低的信噪比下能够显著地提高语音活动检测的准确率,且在多种噪声环境和信噪比条件下具有较好的稳健性。应用于语音识别系统的实验表明,该算法能有效提高噪声环境下的语音识别率。     

采用子带长时信号变化特征的稳健语音活动检测

为提高语音活动检测(VAD)在低信噪比下的准确率,提出了一种基于子带长时信号变化特征的VAD算法.将语音信号转换到频域,并分解为几个不重复的子频带,对这些子带信号分别提取长时信号变化特征,然后采用GMM在线建立语音和非语音模型,以模型的似然比进行VAD判决.实验结果表明,算法在较低的信噪比下能够显著地提高语音活动检测的准确率,且在多种噪声环境和信噪比条件下具有较好的稳健性.应用于语音识别系统的实验表明,该算法能有效提高噪声环境下的语音识别率.     

基于独立分量分析的盲源分离方法研究

盲源分离是指在没有任何先验知识的前提下,从观测到的多个混合信号中提取或分离出未知源信号的过程.本文主要探讨了基于独立分量分析的盲源分离自然梯度算法及活动函数的选取,并利用该算法实现了5路混合信号和3路自然语音信号的分离,最后在Mat-lab2008下进行了仿真验证.结果表明该算法能够有效实现语音信号的分离.     

基于听觉掩蔽效应的改进MFCC特征提取算法

目前,关于语音识别的研究尚处在实验室环境中,而实际的语音总是与噪声和干扰并存。人类能够在信噪比很低甚至在有干扰声音存在的环境中正确识剐语音主要是依靠人的双耳输入作用,本文就模仿人耳的听觉掩蔽效应来掩蔽噪声信号,提出了一种MFCC（Mel频率倒谱系数）改进提取算法。该算法能更好地减少噪声信号对纯净语音信号的影响,从而提高语音信号的识别率。实验表明改进后的算法相对于传统的MFCC提取算法大约有4．43％～8．42％的相对性能提升。     

一种改进的特定人语音识别系统及算法研究

针对传统特定人语音识别过程中存在的算法复杂、所占存储空间大等问题,提出了一种改进的基于动态时间规整算法(DTW)的特定人语音识别系统.在对参数提取方法进行详细对比之后,提取美尔频率倒谱系数(MFCC)作为本系统的语音识别参数,有效的解决了人耳响应不同信号灵敏度不同的问题.利用MATLAB环境下语音工具箱Voice Box实现了对若干数字的孤立词识别,识别速度提高了约30％,识别成功率达到95％以上.仿真结果证明,该系统在算法简单,识别成功率高,是一种简单有效的语音识别方法.     

Speech recognition method using quantised LSP parameters inCELP-type coders

A low-complexity speech recognition method applicable to digital communication networks is proposed. A feature set suitable for speech recognition is obtained from quantised LSP parameters in CELP-type coders without reconstructing the speech signals. The authors present the effects of the speech coder on speaker-independent recognition performance, and show that the recognition accuracy of the proposed method is better than that of the recogniser using reconstructed speech signals     

语音识别的非线性方法

语音信号是一个复杂的非线性过程，这使得基于线性系统理论发展起来的传统语音识别技术性能难以进一步提高。近年来人们开始逐渐重视非线性在语音识别技术中的应用，本文概括地介绍了非线性理论在语音识别技术中的所取得的成果和发展方向，除了涉及较为流行的隐马尔柯夫过程和人工神经网络在语音识别中的应用外，文中着重论述了近年来发展迅猛的混沌，分形理论在语音识别中的应用，本文最后还提到不可忽视的分形理论在语音编码中的应     

基于LS-SVM的情感语音识别

提出了一种基于LS-SVM的情感语音识别方法。即先提取实验中语音信号的基频,能量,语速等参数为情感特征,然后采用LS-SVM方法对相应的情感语音信号建立模型,进行识别。实验结果表明,利用LS-SVM进行基本情感识别时,识别率较高。     

基于语音信号的嵌入式假肢控制系统研究

文中以凌阳SPCE061A单片机作为核心控制芯片,研究了一种基于语音信号的嵌入式假肢控制系统.该控制系统主要由声音采集和音频输出构成,通过语音命令就能够实现对假肢的多自由度运动控制,弥补了基于EMG和EEG控制系统的不足,并且提高了假肢控制系统的识别率,实现了语音信号对假肢的控制.     

Pitch-synchronous wavelet representations of speech and musicsignals

A new wavelet representation is explored. The transform is based on a pitch-synchronous vector representation and it adapts to the oscillatory or aperiodic characteristics of signals. Pseudo-periodic signals are represented in terms of an asymptotically periodic trend and aperiodic fluctuations at several scales. The transform reverts to the ordinary wavelet transform over totally aperiodic signal segments. The pitch-synchronous wavelet transform is particularly suitable to the analysis, rate-reduction coding and synthesis of speech signals and it may serve as a preprocessing block in automatic speech recognition systems. Feature extraction such as separation of voice from noise in voiced consonants is easily performed by means of partial wavelet expansions. A stochastic model of aperiodic fluctuations is proposed     

A new algorithm on multiple unknown source signals estimation and separation in a reverberant space

In many voice-related applications, the presence of echoes and overlapping speech signals can degrade the quality or intelligibility of a desired speech signal to be processed. It is, therefore, important to cancel the echoes and to separate overlapping speech signals from a mixture of these components, so that a specific function of the system, for instance, transmission, speech identification, or recognition, can be accomplished with better performance. However, in many cases we do not know the properties of the communication channel, and sometimes even the number of speech sources is unknown. In this paper, we propose to use a reference signal to determine the channel characteristics. When the estimated channel parameter matrices are obtained, a recurrence formula can then be used to separate various speech signals including their reverberant counterparts. As a finite impulse response (FIR) model is used to describe the observation model of the sources in the reverberant environment, it is not necessary for the processing speech signals to be uncorrelated. Because it involves only simple computation, our approach can be used in online applications. In this paper, we will investigate the validity of our algorithm and compare it with extended fourth-order blind identification (EFOBI). It is found that our method preserves both signal waveforms and their amplitudes even in a noisy environment, whereas EFOBI has not been able to achieve similar performance.     

Improved perceptually non-uniform spectral compression for robust speech recognition

According to the decline of recognition rate of speech recognition system in the noise environments, an improved perceptually non-uniform spectral compression feature extraction algorithm is put forward in this paper. This method can realize an effective compression of the speech signals and make the training and recognition environments more matching, so the recognition rate can be improved in the noise environments. By experimenting on the intelligent wheelchair platform, the result shows that the algorithm can effectively enhance the robustness of speech recognition, and ensure the recognition rate in the noise environments.     

Efficient Speech Enhancement Using Recurrent Convolution Encoder and Decoder

The accuracy of voice or speech recognition is affected due to the presence of various background noises present in the surroundings. Automatic Speech Recognition communication systems are utilized for enhancing the speech by either reducing or eliminating the surrounding noises. The corrupted speech signals are enhanced by using different techniques. In this paper, Recurrent Convolutional Encoder-Decoder (R-CED) network is proposed for enhancing the speech by the elimination of noise signals. The efficiency of the proposed work is determined by comparing the performance metrics like PESQ, STOI and CER with various existing techniques. From the results obtained, it can be confirmed that the efficiency of proposed R-CED is higher and optimal when compared to the existing techniques.

     

Integration of acoustic and visual speech signals using neuralnetworks

Results from a series of experiments that use neural networks to process the visual speech signals of a male talker are presented. In these preliminary experiments, the results are limited to static images of vowels. It is demonstrated that these networks are able to extract speech information from the visual images and that this information can be used to improve automatic vowel recognition. The structure of speech and its corresponding acoustic and visual signals are reviewed. The specific data that was used in the experiments along with the network architectures and algorithms are described. The results of integrating the visual and auditory signals for vowel recognition in the presence of acoustic noise are presented     

一种改进的语音信号特征参数提取算法研究

为了提高语音信号的识别率,提出了一种改进的语音信号特征提取算法。该算法在MFCC参数的基础上,增加每帧信号的短时能量和短时过零率,使得新参数能够更为准确地表征语音信号。通过仿真实验。说明了新特征参数取得了较高的识别率。