共查询到19条相似文献,搜索用时 125 毫秒
1.
为了改善发声力度变化对说话人识别系统性能的影响.针对不同发声力度下语音信号的分析,提出了使用发声力度最大后验概率(Vocal Effort Maximum A Posteriori,VEMAP)自适应方法更新基于高斯混合模型-通用背景模型(Gaussian Mixture Model-Universal Background Model,GMM-UBM)的说话人识别系统模型.实验表明,所提出的方法使不同发声力度下系统EER%降低了88.45%与85.16%,有效解决了因发声力度变化引起的训练语音与测试语音音量失配,从而导致说话人识别性能降低的问题,改善说话人识别系统性能效果显著. 相似文献
2.
3.
听觉特性和语谱特性在说话人识别中的应用 总被引:1,自引:1,他引:0
大多数说话人识别系统当由实验室走向实际应用时,环境噪声的存在会造成其识别性能下降。为了提高噪声环境下说话人识别系统的识别性能,将基于听觉特性和语谱特性的语音增强技术作为预处理器,首先对语音信号进行降噪处理,提高输入信号的信噪比。实验证明,经过降噪处理的语音信号送入说话人识别系统,提高了系统的识别性能。 相似文献
4.
5.
基于小波变换的鲁棒型特征提取及说话人识别 总被引:4,自引:0,他引:4
说话人识别系统在实际应用中面临的主要困难之一是鲁棒性问题,干净语音环境下识别率很高的说话人识别系统,在有噪语音环境下识别性能显著降低。解决这一问题的方法之一是寻找具有鲁棒性的特征参数。本文结合具有多分辨率分析特点的小波变换技术,提出一种基于小波变换的鲁棒型特征提取算法,以提高说话人识别系统在噪声环境下的识别性能。对40个说话人的语音库SUDA2002-D2,在加性高斯白噪声环境下进行的识别实验结果表明,本文提出的特征提取算法可以有效地提高说话人识别系统在噪声环境下的识别性能。 相似文献
6.
说话人识别是信息技术和生物学的新一代身份验证方式,在说话人识别的研究中,特征参数的提取直接影响到识别系统最终的识别效率.通过对Mel频率倒谱系数特征参数进行分析研究,基于Mel频率倒谱系数改进加权函数,将体现个人语音特性的加权特征参数与反映语音帧间变化的差分Mel频率倒谱系数进行维度筛选,再进行参数混合.实验结果表明,通过改进加权函数提取得到的特征参数与差分Mel频率倒谱系数的混合参数在矢量量化的说话人识别系统中,码本容量为16和32时可以达到100%的识别率. 相似文献
7.
大多数实际应用环境中总是存在各种各样的噪声,由于训练环境与识别环境不匹配,现有的绝大多数说话人识别系统在噪声环境中的性能都不可避免的急剧下降。为了让说话人识别系统在强噪声环境中,有较好的识别效果.研究一个将语音增强器和说话人识别系统级连起来的系统,该系统中将语音增强作为前端处理来提高输入的信噪比。实验证明,该系统具有很好的抗噪声性能。 相似文献
8.
模仿者蓄意模仿说话人的语音,当相似度较高时,说话人识别系统就有可能被模仿者欺骗。语音特征参数作为说话人识别系统的关键组成部分,直接影响系统的性能。Mel系数是语音识别领域最成熟的特征参数之一,但是,MFCC特征参数在语音识别中对中、高频段的识别精度较低。为了解决上述问题,融合Mid-MFCC和IMFCC,采用增减分量法,提出了MMI-MFCC特征参数。实验结果表明,新的MMI-MFCC特征参数比传统的MFCC特征参数更有效的区分模仿语音的相似度。 相似文献
9.
噪声环境下,为了提高说话人识别系统的鲁棒性,需要对系统进行各种抗噪声处理。采用梅尔频率倒谱系数作为语音的特征参数,矢量量化方法进行模式匹配,将改进的基于听觉掩蔽效应的语音增强器作为预处理器,对语音信号首先进行降噪处理。语音增强器实验结果表明,经过降噪处理后提高了输入信号的信噪比,减少了语音失真,同时很好地抑制了背景噪声和残余音乐噪声。将经过降噪处理的语音信号送入说话人识别系统,提高了系统的识别性能。 相似文献
10.
11.
由于环境噪声的影响,实际应用中说话人识别系统性能会出现急剧下降。提出了一种基于高斯混合模型-通用背景模型和自适应并行模型组合的鲁棒性语音身份识别方法。自适应并行模型组合是一种噪声鲁棒性的特征补偿算法,能够有效减少训练环境与测试环境之间的不匹配现象,从而提高系统识别准确率和抗噪性能。首先,算法从测试语音中估计出噪声特征,然后用一个单高斯模型对噪声特征进行拟合得到噪声均值和协方差。最后,根据得出的噪声均值和协方差,调整训练好的高斯混合模型均值向量和协方差矩阵,使其尽可能地匹配测试环境。实验结果表明,该方法可以准确地重构干净语音的高斯混合模型参数,并且能够显著提高说话人识别的准确率,特别是在低信噪比情况下。 相似文献
12.
We consider the feature recombination technique in a multiband approach to speaker identification and verification. To overcome the ineffectiveness of conventional feature recombination in broadband noisy environments, we propose a new subband feature recombination which uses subband likelihoods and a subband reliable‐feature selection technique with an adaptive noise model. In the decision step of speaker recognition, a few very low unreliable feature likelihood scores can cause a speaker recognition system to make an incorrect decision. To overcome this problem, reliable‐feature selection adjusts the likelihood scores of an unreliable feature by comparison with those of an adaptive noise model, which is estimated by the maximum a posteriori adaptation technique using noise features directly obtained from noisy test speech. To evaluate the effectiveness of the proposed methods in noisy environments, we use the TIMIT database and the NTIMIT database, which is the corresponding telephone version of TIMIT database. The proposed subband feature recombination with subband reliable‐feature selection achieves better performance than the conventional feature recombination system with reliable‐feature selection. 相似文献
13.
14.
Feature Compensation Combining SNR‐Dependent Feature Reconstruction and Class Histogram Equalization
In this letter, we propose a new histogram equalization technique for feature compensation in speech recognition under noisy environments. The proposed approach combines a signal‐to‐noise‐ratio–dependent feature reconstruction method and the class histogram equalization technique to effectively reduce the acoustic mismatch present in noisy speech features. Experimental results from the Aurora 2 task confirm the superiority of the proposed approach for acoustic feature compensation. 相似文献
15.
噪声环境下说话人识别的组合特征提取方法 总被引:1,自引:0,他引:1
针对在干净语音环境下识别率很高的说话人识别系统,在噪声环境下识别率显著降低的缺点,本文结合具有多分辨率分析特点的小波变换技术,提出一种基于小波变换的组合特征提取算法,以提高说话人识别系统在噪声环境下的识别性能。对40个说话人的语音库SUDA2002-D2,在噪声环境下进行的识别实验结果表明,本文提出的组合特征提取算法可以在噪声环境下有效地提高说话人识别系统的识别性能。 相似文献
16.
Robust speech features based on wavelet transform with application to speaker identification 总被引:2,自引:0,他引:2
Hsieh C.-T. Lai E. Wang Y.-C. 《Vision, Image and Signal Processing, IEE Proceedings -》2002,149(2):108-114
An effective and robust speech feature extraction method is presented. Based on the time-frequency multiresolution property of the wavelet transform, the input speech signal is decomposed into various frequency channels. For capturing the characteristics of an individual speaker, the linear predictive cepstral coefficients of the approximation channel and entropy value of the detail channel for each decomposition process are calculated. In addition, an adaptive thresholding technique for each lower resolution is also applied to remove the influence of noise interference. Experimental results show that using this mechanism not only effectively reduces the influence of noise interference but also improves the recognition performance. Finally, the proposed method is evaluated on the MAT telephone speech database for text-independent speaker identification using the group vector quantisation identifier. Some popular existing methods are also evaluated for comparison, and the results show that the proposed feature extraction algorithm is more effective and robust than the other existing methods. In addition, the performance of the proposed method is very satisfactory even in a low SNR environment corrupted by Gaussian white noise. 相似文献
17.
18.
19.
Ho‐Young Jung 《ETRI Journal》2004,26(3):273-276
We propose a novel feature processing technique which can provide a cepstral liftering effect in the log‐spectral domain. Cepstral liftering aims at the equalization of variance of cepstral coefficients for the distance‐based speech recognizer, and as a result, provides the robustness for additive noise and speaker variability. However, in the popular hidden Markov model based framework, cepstral liftering has no effect in recognition performance. We derive a filtering method in log‐spectral domain corresponding to the cepstral liftering. The proposed method performs a high‐pass filtering based on the decorrelation of filter‐bank energies. We show that in noisy speech recognition, the proposed method reduces the error rate by 52.7% to conventional feature. 相似文献