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基于正弦模型的语音频域参数编码 被引量:1
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作者 郑新春 柴佩琪 《计算机应用与软件》 CSCD 北大核心 2002年第7期53-56,共4页
本文提出了一种基于正弦模型的语音编码技术,通过对语音频率、幅值和相位参数的分析处理,合成高质量的语音。在编码处理过程中,我们应用了语音叠加技术和频率轨迹跟踪技术,以提高合成语音的清晰度。实验结果表明,该编码方式具有很好的... 本文提出了一种基于正弦模型的语音编码技术,通过对语音频率、幅值和相位参数的分析处理,合成高质量的语音。在编码处理过程中,我们应用了语音叠加技术和频率轨迹跟踪技术,以提高合成语音的清晰度。实验结果表明,该编码方式具有很好的顽健性,适合于不同来源的语音信号,例如带背景音乐的语音。 展开更多
关键词 正弦模型 语音频域参数编码 音信号处理 频率匹配
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简易语音频宽演示仪
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作者 张玉华 何丽桥 张振远 《吉林工业大学学报》 CSCD 1994年第4期104-106,共3页
采用电子技术和单片微型计算机技术相结合的方法设计出简易语音频宽演示仪,主要组成部分有48个相互独立的RC有源带通滤波器,利用DS-98C型单片机教学开发机实现语音频宽的显示。
关键词 语音频 RC有源带通滤波 记忆电路 滤波器
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ITU-TP系列语音质量评测标准综述 被引量:2
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作者 曾琦 李璋 +2 位作者 杨玉红 高丽 翟晴 《电声技术》 2011年第12期73-77,共5页
国际电信联盟ITU组织制定的一系列标准,已成为计算机多媒体应用最广泛的国际标准之一。其中ITU-T P系列标准是针对语音质量评测的标准,该系列标准按照测试方法不同,主要分为主观测试和客观模型两大类,分类简介各标准。
关键词 音质量评测标准 ITU-TP系列标准 语音频
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一种基于模糊聚类的矢量量化码书生成算法 被引量:1
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作者 张涛 于凤萍 +1 位作者 张海 韩笑青 《天津大学学报》 EI CAS CSCD 北大核心 2011年第2期157-161,共5页
码书设计是矢量量化中的关键技术.为此,针对经典LBG算法对初始码书敏感的缺陷提出一种基于模糊聚类的码书生成算法.为了提高收敛速度,首先设定距离门限的初始值,然后依次循环逐级递减调整以减少迭代次数.逐级调整门限的也可以降低新聚... 码书设计是矢量量化中的关键技术.为此,针对经典LBG算法对初始码书敏感的缺陷提出一种基于模糊聚类的码书生成算法.为了提高收敛速度,首先设定距离门限的初始值,然后依次循环逐级递减调整以减少迭代次数.逐级调整门限的也可以降低新聚类生成的速度,从而得到更好的更具有典型性的码书;此外,通过对胞腔中矢量按从大到小的顺序择优选取,设计出的码书性能更好,更加接近全局最优.将该算法应用于移动语音频编码标准中线谱频率矢量量化的码书训练,与LBG算法的对比实验结果表明,该算法在主客观质量评价方面都有效地提高了语音频编码算法的性能. 展开更多
关键词 矢量量化 码书生成 模糊聚类 移动语音频编码
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基于丢包率的改进前向纠错算法研究 被引量:1
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作者 王赞 张聪 《软件导刊》 2014年第5期54-57,共4页
针对移动环境下传输信道不稳定,从而造成数据丢失、语音频质量下降等问题,提出了一种基于丢包率的改进前向纠错算法。通过在接收端采用改进型归一化最小均方误差(MLMS)预测器来预测下一个前向纠错块中的丢帧个数,根据丢帧个数计算下一... 针对移动环境下传输信道不稳定,从而造成数据丢失、语音频质量下降等问题,提出了一种基于丢包率的改进前向纠错算法。通过在接收端采用改进型归一化最小均方误差(MLMS)预测器来预测下一个前向纠错块中的丢帧个数,根据丢帧个数计算下一次的丢包率。该丢包率反映了当前的网络状况,这样发送端就可以根据接收端反馈回来的丢包率和丢包个数采取最佳的前向纠错编码方案。仿真实验结果表明,与传统前向纠错算法相比,该算法具有明显优势。 展开更多
关键词 丢包率 MLMS 前向纠错 语音频
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Research on fast real-time adaptive audio mixing in multimedia conference 被引量:2
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作者 樊星 顾伟康 叶秀清 《Journal of Zhejiang University-Science A(Applied Physics & Engineering)》 SCIE EI CAS CSCD 2005年第6期507-512,共6页
In multimedia conference, the capability of audio processing is basic and requires more for real-time criteria. In this article, we categorize and analyze the schemes, and provide several multipoint speech audio mixin... In multimedia conference, the capability of audio processing is basic and requires more for real-time criteria. In this article, we categorize and analyze the schemes, and provide several multipoint speech audio mixing schemes using weighted algorithm, which meet the demand of practical needs for real-time multipoint speech mixing, for which the ASW and AEW schemes are especially recommended. Applying the adaptive algorithms, the high-performance schemes we provide do not use the saturation operation widely used in multimedia processing. Therefore, no additional noise will be added to the output. The above adaptive algorithms have relatively low computational complexity and good hearing perceptibility. The schemes are designed for parallel processing, and can be easily implemented with hardware, such as DSPs, and widely applied in multimedia conference systems. 展开更多
关键词 Multimedia conference MCU REAL-TIME ADAPTIVE Audio mixing Aligned-to-self Aligned-to-energy
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Digital libraries: A testbed for multimedia technology
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作者 庄越挺 《Journal of Zhejiang University-Science A(Applied Physics & Engineering)》 SCIE EI CAS CSCD 2005年第11期1201-1203,共3页
A distinguishing feature of a digital library is that it has Terabyte volumes of multimedia resources. One challenge for researchers in the field of multimedia is to find a testbed for showing the potentials of multim... A distinguishing feature of a digital library is that it has Terabyte volumes of multimedia resources. One challenge for researchers in the field of multimedia is to find a testbed for showing the potentials of multimedia technologies such as video summarization, semantic annotation, multimedia cross indexing and retrieval, and etc. Deeper research and wider applications of digital libraries revealed their indispensable role as testbed for multimedia technologies. This paper presents challenging issues of some key techniques used in digital libraries and their specific needs for multimedia technologies. 展开更多
关键词 Digital library Multimedia technology CADAL (China-America Digital Academic Library)
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Emotion recognition of Uyghur speech using uncertain linear discriminant analysis
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作者 Tashpolat Nizamidin Zhao Li +2 位作者 Zhang Mingyang Xu Xinzhou Askar Hamdulla 《Journal of Southeast University(English Edition)》 EI CAS 2017年第4期437-443,共7页
To achieve efficient a d compact low-dimensional features for speech emotion recognition,a novel featurereduction method using uncertain linear discriminant analysis is proposed.Using the same principles as for conven... To achieve efficient a d compact low-dimensional features for speech emotion recognition,a novel featurereduction method using uncertain linear discriminant analysis is proposed.Using the same principles as for conventional linear discriminant analysis(LDA),uncertainties of the noisy or distorted input data ae employed in order to estimate maximaiy discriminant directions.The effectiveness of the proposed uncertain LDA(ULDA)is demonstrated in the Uyghur speech emotion recognition task.The emotional features of Uyghur speech,especially,the fundamental fequency and formant,a e analyzed in the collected emotional data.Then,ULDA is employed in dimensionality reduction of emotional features and better performance is achieved compared with other dimensionality reduction techniques.The speech emotion recognition of Uyghur is implemented by feeding the low-dimensional data to support vector machine(SVM)based on the proposed ULDA.The experimental results show that when employing a appropriate uncertainty estimation algorithm,uncertain LDA outperforms the conveetional LDA counterpart on Uyghur speech emotion recognition. 展开更多
关键词 Uyghur language speech emotion corpus PITCH FORMANT uncertain linear discriminant analysis (ULDA)
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ARMA Modelling for Whispered Speech
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作者 栗学丽 周卫东 《Journal of Measurement Science and Instrumentation》 CAS 2010年第3期300-303,共4页
The Autoregressive Moving Average (ARMA) model for whispered speech is proposed. with normal speech, whispered speech has no fundamental frequency because of the glottis being semi-opened and turbulent flow being cr... The Autoregressive Moving Average (ARMA) model for whispered speech is proposed. with normal speech, whispered speech has no fundamental frequency because of the glottis being semi-opened and turbulent flow being created, and formant shifting exists in the lower frequency region due to the narrowing of the tract in the false vocal fold regions and weak acoustic coupling with the aubglottal system. Analysis shows that the effect of the subglottal system is to introduce additional pole-zero pairs into the vocal tract transfer function. Theoretically, the method based on an ARMA process is superior to that based on an AR process in the spectral analysis of the whispered speech. Two methods, the least squared modified Yule-Walker likelihood estimate (LSMY) algorithm and the Frequency-Domain Steiglitz-Mcbide (FDSM) algorithm, are applied to the ARMA mfldel for the whispered speech. The performance evaluation shows that the ARMA model is much more appropriate for representing the whispered speech than the AR model, and the FDSM algorithm provides a name acorate estimation of the whispered speech spectral envelope than the LSMY algorithm with higher conputational complexity. 展开更多
关键词 ARMA model AR model whispered speech LSMY
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An Empirical Study of Reading by Chinese English Majors in the Presence of Background Speech
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作者 何享 李霄翔 《Chinese Journal of Applied Linguistics》 2014年第2期213-230,265,共19页
This study investigated how background speech affected L1 and L2 reading of Chinese English major students. English, Dutch, and Mandarin Chinese were respectively set as the second language (L2), foreign language ... This study investigated how background speech affected L1 and L2 reading of Chinese English major students. English, Dutch, and Mandarin Chinese were respectively set as the second language (L2), foreign language (FL), and first language (L1) background speech conditions. Self-paced word-by-word reading paradigm was used to collect the response time (RT) of each word. The conventional analysis revealed that L1 background speech exerted the most disruptive effect on both L1 and L2 reading could be phonological and could be at the and suggested that the background speech effect stage of phonological processing of L1 and L2 reading. It also implied that L1 phonological processing could be simultaneously activated during L2 reading. Spectral analysis of ten subjects' reading data indicated that pink noise existed in each time series of word RT of L1 and L2 reading in each condition. It provided clear evidence that L1 and L2 reading processing are similar with different concurrent background speech. 展开更多
关键词 READING phonological processing background speech pink noise spectral analysis
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