基于TMS320VC5402的语音检测器

本文介绍的语音检测器以DSP芯片TMS320VC5402为核心,对短波电台接收到的信号进行分析和处理。数字语音信号采用串行输入／输出方式,语音检测算法则采用对语音信号进行降噪处理后,再进行短时平均幅度差和短时能量计算的方法。该语音检测器的电路简洁小巧,语音检测准确度高。     

数字语音品质的一种客观评价方法

介绍了一种利用LPC对数倒频谱包络偏离量和人工合成语音信号对数字语音质量进行客观评价的方法;描述了人工合成语音信号的产生方法;简要介绍了ITU—T在语音客观评价方面的有关建议。     

语音端点检测在实时语音截取中的应用

语音端点检测在语音识别系统中占有重要地位。针对在噪声多变的环境中实时截取完整语音信号存在困难,文章提出一种实时语音端点检测方法。该方法首先提取每帧信号的短时平均过零率与Mel频率倒谱系数;然后利用前N帧背景噪声的Mel频率倒谱系数对当前帧进行归一化,并以该特征矢量的L2范数作为另一特征;最后根据多特征分析对有效语音信号进行截取。实验结果表明,该方法在多变的噪声环境中,截取完整语音信号具有较高准确率。     

一种改进的语音端点检测方法研究

为了提高语音信号端点检测的准确率,提出了改进的端点检测方法。该方法在传统基于能量和过零率的端点检测方法基础上,加入第三道门限——近似熵,对信号进行三级门限检测。仿真实验表明,该方法比传统方法更有效、更优越．能够比较准确的检测语音信号。     

基于HHT和OSF的复杂环境语音端点检测

希尔伯特-黄变换是一种全数据驱动的自适应非平稳信号时频分析方法,但是在强噪声环境下语音信号的希尔伯特能量谱曲线波动较大,对语音端点检测造成很大的影响,该文提出了一种基于希尔伯特-黄变换和顺序统计滤波的检测方法。该方法将含噪语音信号进行经验模态分解,通过对固有模态函数进行自适应权重选取获得信号的希尔伯特能量谱,利用顺序统计滤波器对每帧的能量谱进行平滑处理作为语音/非语音的鉴别特征。实验结果表明,该方法适用于复杂噪声环境的端点检测,在低信噪比情况下仍然能够有效地检测出语音信号,降低信号误检率。     

语音信号数字时基压扩

一、语音信号数字特技与时基压扩的概念近年来,随着计算机及大规模数字集成电路的迅速发展,语音数字信号处理得到了相应的发展。语音信号分析模拟、语音合成、语音识别等的研究已较成熟。各种声码器、声控器、语声识别系统、语声合成器等已逐渐有商品出现。语音数字特技处理是从语音信号数字处理中发展出来的一个新的分支,近年来开始逐渐为人们所重视。语音数字特技处理是一种用数字信号处理方法对语音信号进行某种变换、组合、压缩、扩展、合成、仿真的处理技术,以求得特殊的语音效果,如:语音变速重放、音调变换、语声伪造、仿真合成等等。语音数字特技处理作为一种新技术在国外广泛地被应用于广播、教育、文艺、公安等各个领域。     

改进的基于长时谱能量差异和基音比例的语音检测方法

语音检测是语音信号处理的前端,利用长时谱能量差异特征的语音检测无法区分突发噪声和语音,掺杂着突发噪声的语音信号会对语音处理系统带来不良影响。提出了一种基于长时谱能量差异特征和基音比例特征相结合的语音检测方法,该方法的优点是,在利用长时谱能量差异特征基础上引入基音比例特征,从而有效减少了将信号中突发噪声误判为语音的错误。实验显示,该算法能够在多种信噪比环境下取得很好的检测结果。     

基于自适应滤波和小波变换的基音检测

针对语音信号在小信噪比条件下检测其基音周期。考虑自适应滤波和小波变换的优点对小信噪比条件下的语音信号进行基音周期检测,实验证明此方法能有效检测-20dB下的基音周期。     

英文语音发音标准化的模式识别对比方法改进分析

针对英文语音发音标准化评价准确性不高的问题,提出一种基于发音特征倒谱系数感知的英文语音发音标准化的模式识别对比方法。首先构建英文语音发音的语音信号采集模型,对采集的英文语音信号进行发音器官的动作属性配对描述。然后提取英文语音发音信号的倒谱特性,采用梅尔频率倒谱系数感知方法进行英文语音发音特征建模和发音位置及方式的模式识别,为语音发音提供标准化对比模式。最后进行实验分析,测试结果表明,采用该方法进行英文语音发音特征检测和模式识别的准确度较高,对发音特征的声学建模有效可靠。     

基于广义维数距离的语音端点检测方法

为能够准确有效地对含噪声语音信号进行起止位置的端点检测,该文提出了一种基于广义维数距离的端点检测方法。首先利用覆盖法求取广义维数得到该语音信号的三维特征向量,包括容量维数、信息维数、关联维数;然后计算信号的维数特征距离;最后根据特征距离对语音信号类别进行决策分类。实验结果表明,与仅使用单一维数特征检测语音起止端点相比,该文所提出的方法具有较好的鲁棒性,对混杂有不同噪声、不同信噪比的语音信号都能有较好的检测结果,尤其适用于低信噪比的语音端点检测。     

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     

利用韵律信息的CHMM连续数字语音识别

提出了一种结合韵律信息的高性能汉语连续数字语音识别算法,该识别算法基于CHMM(连续隐马尔可夫模型),采用MFCC(MEL频率倒谱系数)为主要语音特征参数,结合韵律信息进行连续数字精确分割,能够有效区分易混数字。算法采用两级识别框架来提高语音识别率,其中,第1级对连续数字分割,在此基础上进行数字语音识别,输出各候选结果,第2级在候选结果中确定易混数字对,并运用韵律信息进一步选择正确结果。实验表明,最终汉语连续数字语音识别率有很大提高。     

混沌同步新方法研究

从数字通信技术角度提出了混沌同步定义，指出混沌数字保密通信的应用研究重点是混沌严格同步技术。从网络通信协议的思路出发，在混沌脉冲同步的启发下，提出了一种新的混沌同步方法——混沌协议同步技术．并加以改进，从而解决了混沌数字保密通信的关键技术难题。最后构建了一个基于协议同步的网络混沌语音保密通信系统．验证了此方法的可行性。     

A nonlinear operator-based speech feature analysis method withapplication to vocal fold pathology assessment

Traditional speech processing methods for laryngeal pathology assessment assume linear speech production with measures derived from an estimated glottal flow waveform. They normally require the speaker to achieve complete glottal closure, which for many vocal fold pathologies cannot be accomplished. To address this issue, a nonlinear signal processing approach is proposed which does not require direct glottal flow waveform estimation. This technique is motivated by earlier studies of airflow characterization for human speech production. The proposed nonlinear approach employs a differential Teager energy operator and the energy separation algorithm to obtain formant AM and FM modulations from filtered speech recordings. A new speech measure is proposed based on parameterization of the autocorrelation envelope of the AM response. This approach is shown to achieve impressive detection performance for a set of muscular tension dysphonias. Unlike flow characterization using numerical solutions of Navier-Stokes equations, this method is extremely computationally attractive, requiring only a small time window of speech samples. The new noninvasive method shows that a fast, effective digital speech processing technique can be developed for vocal fold pathology assessment without the need for direct glottal flow estimation or complete glottal closure by the speaker. The proposed method also confirms that alternative nonlinear methods can begin to address the limitations of previous linear approaches for speech pathology assessment     

多频编码的短波线路压扩技术

本文以传统短波线路压缩扩张器的思想为基础，提出了一种以现代数字语音信号处理技术为核心的语音分段压扩方法，因为该方法的主要处理过程都集中于数字信号处理部分，并采用了调制编码的方法，所以与传统的压扩器思想相比较更强的可实现性，同时也获得了满意的效果。     

Time diversity by pitch-synchronized interpolation

In mobile communications such as automobile telephone systems, an instantaneous interruption is caused by rapid fading, and speech quality is markedly deteriorated. A pitch-synchronized interpolation method is proposed to improve this quality. This method utilizes the fact that most speech signal regions have a pitch period. In this method, when an instantaneous interruption is detected at the receiver, a speech signal, which includes instantaneous interruption noise, is deleted and interpolated by repetition of the speech signal received one pitch interval wave before the instantaneous interruption. This method can be applied to receivers used in analog and digital transmission systems. A receiver using this method has been constructed, and it was shown that a 10-dB carrier-to-noise ratio (CNR) gain can be obtained by this receiver.     

PWM方式输出合成语音

针对采用波形编码方式语音合成集成电路设计出的新型D/A转换方式,即利用脉冲宽度调制（PWM）技术,将数字语音信息直接转化成脉冲宽度调制波,通过低通滤波器可以恢复出模拟语音信号。本文所设计的数字脉冲宽度调制器采用CMOS工艺,适合于不同的语音合成集成电路。     

用数字信号处理方法研究汉语普通话音素的区别特征

本文运用语言信号数字处理方法，研究了汉语普通话音素的区别特征，研究结果进一步完善了汉语普通话音素的区别特征矩阵表，将为基于音素的计算机汉语普通话语音分析、合成和识别提供了一种有效的参考方法。     

Pseudo-analog speech transmission in mobile radio communicationsystems

A low-complexity pseudo-analog speech transmission scheme is proposed for portable communications. It uses a speech coder based on adaptive differential pulse code modulation (ADPCM) in combination with a multilevel digital modulation technique such as M-ary DPSK or M-ary FSK and features low quantization noise, bandwidth efficiency, and robustness to transmission errors. A nonsymmetric M -ary DPSK scheme called skewed M-ary DPSK is proposed to enhance the noisy channel performance. Comparison to conventional analog FM and a digital speech transmission scheme using adaptive predictive coding and forward error correction (FEC) based on convolutional coding shows that the pseudo-analog system has the best objective signal-to-noise ratio performance under most channel conditions. Informal subjective evaluations rate the digital system superior to the pseudo-analog scheme for bad channels and conversely for good channels. It is concluded that the pseudo-analog system can be designed with low delay and high speech quality for good channels with high spectral efficiency     

抗PCM A/μ律压缩及其转换的语音密写

该文提出一种在数字语音信号中嵌入隐蔽信息的方法.根据PCM A/μ律压缩及其转换特性,采用对某些宿主语音样本预修改和动态选择嵌入比特位的方法,使密写信号不仅没有感知失真,而且经A/μ律压缩及其相互转换处理后仍可无误提取嵌入的隐蔽数据,因此含密语音信号不仅可在Internet上传输,还可在跨国PSTN中传输.实验结果表明了该方法的有效性.