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提出了一种基于双帧动态时间规整(Double Frame Dynamic Time Wrapping)的识别方法,通过缩小帧匹配距离矩阵(Frame Match Distance Matrix)的规模从而减少计算量,提高识别速度.实验表明,基于双帧动态时间规整的识别方法在识别率有所提高的条件下,缩短了识别时间,有很好的实用价值. 相似文献
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提出一种基于动态规划的矢量量化方法DPVQ(DynamicProgramBasedVectorQuantiza-tion。新方法既能有效压缩语音特征,又能充分保持时间序列信息,同时又兼顾码语音不同段的统计特性。 相似文献
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《电子技术与软件工程》2017,(24)
针对噪音环境下,语音识别率严重下降,根据清浊音发音原理的不同,提出一种清浊音分离抗噪的语音识别算法。实验结果表明,该方法能在噪音环境下明显提高语音识别率。 相似文献
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主要研究用于分布式语音识别(DSR)的语音参数的提取方法以及参数性能分析。以前所用到的语音参数大部分是LPC例谱参数,但其抗噪声性能较差。文中主要讨论了MEL倒谱参数。并在移动通信环境下,比较了两者的性能。 相似文献
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将小波变换的多分辨率特性用于改进Mel频率倒谱系数MFCC的前端处理中,给出了一种新的语音特征参数——小波MFCC。其特点在于采用小波变换、分层FFT和频率合成代替原来MFCC中的FFT部分,使频谱分辨率提高了一倍。试验证明,小波MFCC特征参数在噪声环境和较大词汇量情况下,其抗噪性和识别率均优于MFCC特征参数的结果。 相似文献
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为解决深度学习模型在语音识别芯片上内存占用过大的问题,训练时间长,训练深度不够等问题,本文在Transformer语音识别模型的基础上,提出一种减少Transformer计算复杂度的模型,新的模型参数仅有原模型的1/10,模型训练速度大大提升。此外本文在对模型进行参数调整的过程中还对Transformer模型结构中add&norm这一结构结合残差神经网络进行修改,针对模型在训练过程中因为模型深度太深(容易产生梯度消失和梯度爆炸)导致的模型收敛速度过慢或者模型不收敛等问题,在保证词错率下降的同时,加快模型的收敛速度和训练速度,修改结构之后的模型在词错率上相比原有模型词错率更低,并且收敛速度也比未修改结构的模型收敛速度更快。 相似文献
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噪声下差分复合子带语音识别方法 总被引:4,自引:0,他引:4
本文根据子带特征反映语音信号局部特性和全带特征反映语音信号整体特性的事实,提出了 一种差分复合子带语音识别新方法。先用频谱差分减少噪声的干扰,再将多子带特征识别概率与全带特征识别概率相结合进行综合判决,以得到最终识别结果。将新方法应用于TIMIT数据包0-9十个英文数字和E-Set在NoiseX92的白噪声和F16战机噪声下的识别实验。实验结果表明新方法比传统方法识别性能有很大提高。 相似文献
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语音情感识别的研究进展 总被引:11,自引:0,他引:11
情感在人类的感知、决策等过程扮演着重要角色.长期以来情感智能研究只存在于心理学和认知科学领域,近年来随着人工智能的发展,情感智能跟计算机技术结合产生了情感计算这一研究课题,这将大大的促进计算机技术的发展.情感自动识别是通向情感计算的第一步.语音作为人类最重要的交流媒介,携带着丰富的情感信息.如何从语音中自动识别说话者的情感状态近年来受到各领域研究者的广泛关注.本文从语音情感识别所涉及的几个重要问题出发,包括情感理论及情感分类、情感语音数据库、语音中的情感特征和语音情感识别算法等,介绍了当前的研究进展,并讨论了今后研究的几个关键问题. 相似文献
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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. 相似文献
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端到端语音识别模型由于结构简单且容易训练,已成为目前最流行的语音识别模型.然而端到端语音识别模型通常需要大量的语音-文本对进行训练,才能取得较好的识别性能.而在实际应用中收集大量配对数据既费力又昂贵,因此其无法在实际应用中被广泛使用.本文提出一种将RNN-T(Recurrent Neural Network Trans... 相似文献
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实际的研究表明,语音情感识别方法有多种.介绍了一种基于GMM的语音情感识别方法,包括该方法的优点、存在的问题或不足等,并对此进行了思考,给出了一些处理办法. 相似文献
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To utilize the supra-segmental nature of Mandarin tones, this article proposes a feature extraction method for hidden markov model (HMM) based tone modeling. The method uses linear transforms to project F0 (fundamental frequency) features of neighboring syllables as compensations, and adds them to the original F0 features of the current syllable. The transforms are discriminatively trained by using an objective function termed as "minimum tone error", which is a smooth approximation of tone recognition accuracy. Experiments show that the new tonal features achieve 3.82% tone recognition rate improvement, compared with the baseline, using maximum likelihood trained HMM on the normal F0 features. Further experiments show that discriminative HMM training on the new features is 8.78% better than the baseline. 相似文献
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Lee L.-M. Chen J.-K. Wang H.-C. 《Vision, Image and Signal Processing, IEE Proceedings -》1994,141(6):397-402
The authors deal with the problem of automatic speech recognition in the presence of additive white noise. The effect of noise is modelled as an additive term to the power spectrum of the original clean speech. The cepstral coefficients of the noisy speech are then derived from this model. The reference cepstral vectors trained from clean speech are adapted to their appropriate noisy version to best fit the testing speech cepstral vector. The LPC coefficients, LPC derived cepstral coefficients, and the distance between test and reference, are all regarded as functions of the noise ratio (the spectral power ratio of noise to noisy speech). A gradient based algorithm is proposed to find the optimal noise ratio as well as the minimum distance between the test cepstral vector and the noise adapted reference. A recursive algorithm based on Levinson-Durbin recursion is proposed to simultaneously calculate the LPC coefficients and the derivatives of the LPC coefficients with respect to the noise ratio. The stability of the proposed adaptation algorithm is also addressed. Experiments on multispeaker (50 males and 50 females) isolated Mandarin digits recognition demonstrate remarkable performance improvements over noncompensated method under noisy environment. The results are also compared to the projection based approach, and experiments show that the proposed method is superior to the projection approach under a severe noisy environment 相似文献
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为了识别简单语音,设计了一个基于LPCC参数的语音识别系统。该系统其主要功能有语音信号的录制、播放、预处理、分段滤波、特征提取以及识别语音。最后通过仿真实验验证了本系统能够达到识别简单语音的要求,但仍有需改进的地方,如:能否在复杂环境下识别比较复杂的语音。 相似文献