检索结果-内蒙古大学图书馆

Mo1503 Narrow Band Imaging and High-Magnification Endoscopy (NBI-Hme) for Gastric Lesions Differentiation and Histology Prediction

引用

Gastrointestinal Endoscopy 2014年第5期79卷 AB461-AB461页

作者： Roman Kuvaev Evgeny Nikonov Sergey V. Kashin Herbert Edelsbrunner Viktor Kapranov Michael Machin OLGA a. Dunaeva Endoscopy Yaroslavl Regional Cancer Hospital Yaroslavl Russian Federation Administration Polyclinic No.1 of the Buisness Administration for the President of the Russian Federation Moscow Russian Federation Institute of Science and Technology Austria (IST Austria) Klosterneuburg Austria Department of the University Internet Centre Demidov Yaroslavl State University Yaroslavl Russian Federation Digital Signal and Image Processing Laboratory Demidov Yaroslavl State University Yaroslavl Russian Federation

来源：评论

学校读者我要写书评

暂无评论

Real-time classification of Persian musical Dastgahs using artificial neural network

Real-time classification of Persian musical Dastgahs using a...

引用

16th CSI International Symposium on Artificial Intelligence and signal processing, AISP 2012

作者： Hajimolahoseini, H. Amirfattahi, R. Zekri, M. Digital Signal Processing Research Lab Department of Electrical and Computer Engineering Isfahan University of Technology Isfahan Iran

ISBN: (纸本)9781467314794

In this paper, a three layer artificial neural network is proposed that, independent of Root note selection, classifies the Persian musical Dastgahs into five main groups: Shur, Homayun, Segah, Chahargah and Mahur. It implies the training of the network in order to be able to classify all of 120 possible cases for the Persian musical Dastgahs based on 24 different Root notes in an octave of Persian music. The network receives a vector of 24 possible notes of an octave in which the used notes are represented by a 1 and unused notes are represented by a 0. After training, the network is able to correctly classify 100 percent of the 120 possible cases for Persian musical Dastgahs in five main groups. © 2012 IEEE.

关键词： Network layers

来源：评论

学校读者我要写书评

暂无评论

Higher-order integrated wavetable synthesis

Higher-order integrated wavetable synthesis

引用

15th International Conference on digital Audio Effects, DAFx 2012

作者： Franck, Andreas Välimäki, Vesa Acoustics Department Fraunhofer Institute for Digital Media Technology Ilmenau Germany Department of Signal Processing and Acoustics Aalto University Espoo Finland

Wavetable synthesis is a popular sound synthesis method enabling the efficient creation of musical sounds. Using sample rate conversion techniques, arbitrary musical pitches can be generated from one wavetable or from a small set of wavetables: downsampling is used for raising the pitch and upsampling for lowering it. A challenge when changing the pitch of a sampled waveform is to avoid disturbing aliasing artifacts. Besides bandlimited resampling algorithms, the use of an integrated wavetable and a differentiation of the output signal has been proposed previously by Geiger. This paper extends Geiger's method by using several integrator and differentiator stages to improve alias-reduction. The waveform is integrated multiple times before it is stored in a wavetable. During playback, a sample rate conversion method is first applied and the output signal is then differentiated as many times as the wavetable has been integrated. The computational cost of the proposed technique is independent of the pitch-shift ratio. It is shown that the higher-order integrated wavetable technique reduces aliasing more than the first-order technique with a minor increase in computational cost. Quantization effects are analyzed and are shown to become notable at high frequencies, when several integration and differentiation stages are used.

关键词： signal sampling

来源：评论

学校读者我要写书评

暂无评论

A PLL based adaptive power line interference filtering from ECG signals

A PLL based adaptive power line interference filtering from ...

引用

16th CSI International Symposium on Artificial Intelligence and signal processing, AISP 2012

作者： Zeinali Zadeh, M.M. Niketeghad, S. Amirfattahi, R. Digital Signal Processing Research Lab. Department of Electrical and Computer Engineering Isfahan University of Technology 84156-83111 Iran

ISBN: (纸本)9781467314794

One of the challenges for electrocardiogram (ECG) signals extraction is additive power line interference (PLI) which contains 50Hz or 60Hz base frequency and several harmonics. Variety of fixed-frequency and adaptive notch filter structures are proposed for power line interference suppression. Because of frequency varying behavior of power line interference, fixed frequency filters suffer from the problem of low noise attenuation rate when the interference frequencies drift from their nominal values, and also signal amplitude and phase destruction around interference frequencies due to relatively high notching bandwidth. On the other hand adaptive filters are given high calculation costs. In this paper we propose a real-time, low calculation cost adaptive notch filter, using phase-locked loop (PLL). Experimental results show that in extremely noisy environments where input SNR is around 20dB, average SNR enhancement of proposed method is approximately 40dB. This value is also as good as 35dB for 10dB input SNR. © 2012 IEEE.

关键词： Phase locked loops

来源：评论

学校读者我要写书评

暂无评论

Multichannel signal enhancement using a remote wireless microphone

Multichannel signal enhancement using a remote wireless micr...

引用

International Workshop on Acoustic signal Enhancement, IWAENC 2012

作者： Bloemendal, Brian Van De Laar, Jakob Sommen, Piet Eindhoven University of Technology Department of Electrical Engineering Eindhoven Netherlands Philips Research Laboratories Digital Signal Processing Group Eindhoven Netherlands

ISBN: (纸本)9783800734511

A novel approach to multichannel signal enhancement is presented that exploits data from a remote wireless microphone (RWM). This RWM is placed near an interfering source and transmits only autocorrelation data of its observations to a host, i.e., not the entire signal. The host has access to the autocorrelation data from the RWM and measurements from a local microphone array. A blind source extraction algorithm exploits this correlation data in order to identify the desired extraction filter. This novel approach has several advantages compared to transmitting the entire signal of the RWM. By transmitting only correlation data the required transmission power reduces significantly and synchronization at sample level is not required. Furthermore, leakage of the desired source into the RWM does not degrade performance;and finally, both the desired and interfering sources may be active simultaneously. © 2012, Institute of Electrical and Electronics Engineers Inc. All rights reserved.

关键词： Autocorrelation

来源：评论

学校读者我要写书评

暂无评论

ICL ONVIF Middleware: Additional services on TI Davinci platform

ICL ONVIF Middleware: Additional services on TI Davinci plat...

引用

International Conference on Fuzzy Systems and Knowledge Discovery (FSKD)

作者： Yi-Hsing Tsai Digital Signal Processing Technology Department Industrial Technology and Research Institute Hsinchu Taiwan

This paper describes the efficiency distributed video content management in surveillance system and ICL ONVIF Middleware which is followed onvif Interface Specification. The system is designed on TI DaVinci DM36x architecture and passed ONVIF test tool. The main project was to develop an IP-based physical security follow ONVIF standard. This ICL 1 ONVIF middleware was using iSCSI distributed network to establish distributed surveillance system. In this paper, we aim at the combination of multimedia content processing by load balance control and building a distributed network storage space surveillance system.

关键词： Streaming media Middleware Multimedia communication Surveillance Cameras Computer architecture Standards

来源：评论

学校读者我要写书评

暂无评论

Real-time classification of Persian musical Dastgahs using artificial neural network

Real-time classification of Persian musical Dastgahs using a...

引用

International Symposium on Artificial Intelligence and signal processing (AISP)

作者： H. Hajimolahoseini R. Amirfattahi M. Zekri Digital Signal Processing Research Lab Department of Electrical and Computer Engineering Isfahan University of Technology Isfahan Iran

In this paper, a three layer artificial neural network is proposed that, independent of Root note selection, classifies the Persian musical Dastgahs into five main groups: Shur, Homayun, Segah, Chahargah and Mahur. It implies the training of the network in order to be able to classify all of 120 possible cases for the Persian musical Dastgahs based on 24 different Root notes in an octave of Persian music. The network receives a vector of 24 possible notes of an octave in which the used notes are represented by a “1” and unused notes are represented by a “0”. After training, the network is able to correctly classify 100 percent of the 120 possible cases for Persian musical Dastgahs in five main groups.

关键词： Training Neurons Vectors Multiple signal classification Artificial neural networks Support vector machine classification Biological neural networks

来源：评论

学校读者我要写书评

暂无评论

A PLL based adaptive power line interference filtering from ECG signals

A PLL based adaptive power line interference filtering from ...

引用

International Symposium on Artificial Intelligence and signal processing (AISP)

作者： M. M. Zeinali Zadeh S. Niketeghad R. Amirfattahi Digital Signal Processing Research Laboratory Department of Electrical and Computer Engineering Isfahan University of Technology Isfahan Iran

One of the challenges for electrocardiogram (ECG) signals extraction is additive power line interference (PLI) which contains 50Hz or 60Hz base frequency and several harmonics. Variety of fixed-frequency and adaptive notch filter structures are proposed for power line interference suppression. Because of frequency varying behavior of power line interference, fixed frequency filters suffer from the problem of low noise attenuation rate when the interference frequencies drift from their nominal values, and also signal amplitude and phase destruction around interference frequencies due to relatively high notching bandwidth. On the other hand adaptive filters are given high calculation costs. In this paper we propose a real-time, low calculation cost adaptive notch filter, using phase-locked loop (PLL). Experimental results show that in extremely noisy environments where input SNR is around - 20dB, average SNR enhancement of proposed method is approximately 40dB. This value is also as good as 35dB for -10dB input SNR.

关键词： IIR filters Power harmonic filters Adaptive filters Electrocardiography Interference signal to noise ratio Bandwidth

来源：评论

学校读者我要写书评

暂无评论

Efficient folded VLSI architectures for linear prediction error filters 12

Efficient folded VLSI architectures for linear prediction er...

引用

22nd Great Lakes Symposium on VLSI, GLSVLSI'2012

作者： Salehi, Sayed Ahmad Amirfattahi, Rasoul Parhi, Keshab K. Department of Electrical and Computer Engineering Digital Signal Processing Research Lab Isfahan University of Technology P.O.Box 84156-83111 Isfahan Iran Department of Electrical and Computer Engineering University of Minnesota 200 Union Street SE Minneapolis MN 55455 United States

ISBN: (纸本)9781450312448

In this paper we propose two efficient low-area, low-power folded VLSI architectures for linear prediction error filter. One of them is based on the split-Levinson-Durbin and requires half computational complexity of an architecture based on the Levinson-Durbin algorithm. The other one is based on the Schur algorithm. Using folding method, the number of multipliers and adders is minimized. In addition, by modifications in data scheduling, the number of required multiplexers are also decreased. Comparison with previous architectures demonstrates the efficiency of the proposed architectures with respect to hardware and computational complexity. Copyright 2012 ACM.

关键词： Computational complexity

来源：评论

学校读者我要写书评

暂无评论

Beamformer design exploiting blind source extraction techniques

Beamformer design exploiting blind source extraction techniq...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： Brian Bloemendal Jakob van de Laar Piet Sommen Department of Electrical Engineering Eindhoven University of Technology Eindhoven Netherlands Digital Signal Processing Group Philips Research Laboratories Eindhoven Netherlands

Recently we have shown how a blind source extraction (BSE) algorithm can be equipped with some prior information about mixing parameters of the desired source in order to extract this source. The prior information, which may contain errors, is used to construct a matrix from linear combinations of correlation matrices. The extraction filter is easily obtained from the specific eigenstructure of this matrix. Here we project the beamformer design problem onto the above mentioned BSE algorithm by parameterizing the mixing system. We show in three ways that the proposed method is efficient and flexible. First, with one procedure an LCMV and MVDR beamformer can be obtained. Second, by taking only two appropriate linear combinations of correlation matrices, which may be interpreted as selection beamformers, the desired source can be selected. Third, selection beamformers can be designed for a subset of sensors while the final beamformer exploits data from all sensors.

关键词： Vectors Sensors Correlation Data mining Arrays Eigenvalues and eigenfunctions Algorithm design and analysis

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：