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检索条件"任意字段=IEEE Workshop on Signal Processing Systems Design and Implementation"

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A novel portable device for pathological voice analysis

A novel portable device for pathological voice analysis

引用

ieee International workshop on Medical Measurement and Applications (MEMEA)

作者： A. Palumbo F. Amato B. Calabrese M. Cannataro P. Veltri A. Garozzo N. Lombardo School of Computer and Biomedical Engineering University Magna Græcia of Catanzaro Catanzaro Italy Department of Electronics Computer Science and Systems University of Calabria Rende Italy School of of Otorhinolaryngology University Magna Græcia of Catanzaro Catanzaro Italy

This paper presents a novel implementation of a DSP based acquisition and elaboration system for vocal signals. The proposed system performs real time spectro-acustic analysis of acquired voice samples and gives a visual feedback to patients alerting them about the eventual presence of anomalies. The prototype will be used for rehabilitation purpose (after medical treatment or surgery) or prevention purpose (early diseases detection).

关键词： Pathology Speech analysis Digital signal processing Real time systems Performance analysis Feedback Prototypes Medical treatment Surgery Diseases

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Distributed detection over adaptive networks based on diffusion estimation schemes

Distributed detection over adaptive networks based on diffus...

引用

ieee workshop on signal processing Advances in Wireless Communications (SPAWC)

作者： Federico S. Cattivelli Ali H. Sayed Department of Electrical Engineering University of California Los Angeles CA USA

We study the problem of distributed detection, where a set of nodes are required to decide between two hypotheses based on their measurements. We seek fully distributed implementations, where all nodes make individual decisions by communicating with their immediate neighbors, and no fusion center is necessary. This scheme provides the network with more flexibility, saves energy for communication and networking resources. Our distributed detection algorithm is based on a previously proposed distributed estimation algorithm. We establish the connection between the detection and estimation problems, propose a distributed detection algorithm, and analyze the performance of the algorithm in terms of its probabilities of detection and false alarm. We also provide simulation results comparing with other cooperation schemes.

关键词： Adaptive systems Detection algorithms Time measurement Iterative algorithms Performance analysis Resonance light scattering Electric variables measurement Algorithm design and analysis Context Adaptive filters

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Robust audio precompensation with probabilistic modeling of transfer function variability

Robust audio precompensation with probabilistic modeling of ...

引用

ieee workshop on Applications of signal processing to Audio and Acoustics

作者： Lars-Johan Brännmark Department of Engineering Sciences Signals and Systems University of Uppsala Uppsala Sweden

A new approach to the single-channel loudspeaker equalization problem is presented. A scalar discrete-time mixed-phase precompensation filter is designed to be spatially robust, meaning that equalization performance should be insensitive to listener movements within a predefined spatial region. The problem is posed in a single-input multiple-output (SIMO) feedforward control framework and a polynomial solution is derived, based on a set of room transfer functions (RTFs) measured at a number of control points in the region, and a multipoint mean-square error (MSE) criterion. Spatial robustness is obtained by the introduction of two novel strategies. Firstly, a probabilistic model is used to describe the RTF variability around each control point, and the MSE criterion is averaged with respect to this variability. Secondly, the pre-response errors, normally associated with mixed-phase equalizer design, are alleviated by restricting the compensator to have a certain structure. The proposed method is shown to produce filters with excellent time-and frequency-domain performance.

关键词： Robustness Transfer functions Polynomials Loudspeakers Equalizers Robust control Information filtering Information filters Conferences Acoustic applications

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Carrier Frequency Offset and I/Q imbalance estimation circuit RTL design for OFDM direct conversion receivers

Carrier Frequency Offset and I/Q imbalance estimation circui...

引用

ieee International Symposium on Intelligent signal processing and Communication systems (ISPACS)

作者： Franz A. de Leon Leonardo A. Lanante Masayuki Kurosaki Hiroshi Ochi University of Philippines Philippines Kyushu Institute of Technology Japan

In this paper, we present a model-based register transfer level (RTL) design of a simple joint parameter method for carrier frequency offset and I/Q imbalance estimation. The estimation scheme is based on a proposed algorithm that uses only two short training symbols for faster estimation. The simple yet robust algorithm is then optimized for field programmable gate array (FPGA) implementation. Simulation results show that the performance of the 16-bit fixed-point system approximates that of the ideal performance of the estimation algorithm.

关键词： Frequency estimation Frequency conversion Circuits OFDM Low pass filters signal processing algorithms Field programmable gate arrays signal design Least squares approximation Wireless communication

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Low-power linear-phase delay filters for neural signal processing: Comparison and synthesis

Low-power linear-phase delay filters for neural signal proce...

引用

ieee International Symposium on Circuits and systems (ISCAS)

作者： Benoit Gosselin Adeline Zbrzeski Mohamad Sawan Eric Kerherve Polystim Neurotechnologies Laboratory École Polytechnique de Montréal Montreal Canada IMS Laboratory University of Bordeaux 1 Bordeaux France

ISBN: (纸本)9781424438273

We present the design and implementation of linear-phase delay filters for ultra-low power neural signal processing. The filters are intended to implement a low-distortion delay element for automatic biopotential detection in neural recording implants. Continuous-time OTA-C filters are used to realize a 9 th -order equiripple transfer function presenting a constant group delay. This analog delay allows to process neural waveforms with reduced overhead compared with digital delays. An allpass transfer function is used to implement such analog delay because it achieves wider constant-delay bandwidth than all-pole does. Two filters realizations are compared for implementing it: the cascaded structure and the inverse follow-the-leader feedback filter. Their respective strengths and drawbacks are assessed by modeling parasitics and non-idealities of OTAs, and using transistor-level simulations. A power budget of 200 nA is used in both filters. Experimental measurements with the chosen topology are presented and discussed.

关键词： Delay lines Filters signal processing signal synthesis Biomedical signal processing Transfer functions Propagation delay signal design Implants Bandwidth

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Robust leakage-based transmit beamforming with probabilistic constraint for downlink multi-user MIMO system

Robust leakage-based transmit beamforming with probabilistic...

引用

ieee/SP workshop on Statistical signal processing (SSP)

作者： Huiqin Du Pei-Jung Chung School of Engineering University of Edinburgh UK

Multi-user multiple-input and multiple-output (MU-MIMO) wireless systems have the potential to provide a substantial gain by using transmit beamforming to allow multi-user communication in the same frequency and time slots. The main challenge for transmit beamforming design is to suppress the co-channel interference (CCI) from other users. In order to completely cancel the CCI at each user, perfect channel state information (CSI) is required at base station, which is generally not available in practice. To overcome the performance degradation caused by the imperfections, the most common approach is the worst-case method, which leads to conservative result as the extreme (but rare) conditions may occur at a very low probability. In this work, we propose a probabilistic-constrained beamforming based on signal-to-leakage ratio (SLR) criterion under consideration of inaccurate channel information. The simulation results show that the proposed beamformer achieves the lowest bit error rate (BER) and leaks the least transmit power from the desired user to all other users among the state-of-art transmit beamformers.

关键词： Robustness Array signal processing Downlink MIMO Radiofrequency interference Bit error rate Interference constraints Frequency Interchannel interference Channel state information

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Distributed implementation design of full software real-time encoder applicable for super high definition video

Distributed implementation design of full software real-time...

引用

ieee International Symposium on Intelligent signal processing and Communication systems (ISPACS)

作者： Sei Naito Shuichi Matsumoto KDD Research and Development Laboratories Inc. Fujimino Saitama Japan

Super high definition video such as 4 K digital cinema is expected as the quite attractive video application. With regard to the video transmission, recent compression coding standards are applicable for 4 K digital cinema. However, the typical implementation approach of a commercial video encoder is based on hardware architecture. As the limitation of hardware base implementation, once the equipment has been completed, further improvement such as supporting higher resolution video or adding unsupported coding tools would be pretty difficult. On the other hand, distributed computing by utilizing several PC platforms has become more typical as the approach for developing a high performance computer system. With regard to the real time application where precise timing control is required, the practical example of distributed computing system has not been existing yet. From this perspective, this paper describes the key technology necessary to realize full software implementation of real time video encoder by utilizing distributed computing technology.

关键词： Software design High definition video Video compression Distributed computing Motion pictures Application software Hardware Computer architecture High performance computing Real time systems

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Analysis of temporal gene expression profiles using time-dependent MUSIC algorithm

Analysis of temporal gene expression profiles using time-dep...

引用

ieee International workshop on Genomic signal processing and Statistics (GENSIPS)

作者： Nidhal Bouaynaya Dan Schonfeld Radhakrishnan Nagarajan Department of Systems Engineering University of Arkansas Little Rock USA Department of Electrical and Computer Engineering University of Illinois Chicago USA Department of Biostatistics University of Arkansas for Medical Sciences USA

Identifying periodically expressed genes and their subsequent transcriptional circuitry can shed new lights in studying the molecular basis of many diseases including cancer; and subsequently provide potential drug targets to treat them. Classical approaches for detecting periodically expressed transcripts in paradigms such as cell-cycle implicitly assume the given data to be stationary. However, it has been experimentally shown that modulation in the magnitude of gene expression is ubiquitous and defy stationary assumptions. In this paper, we formulate the problem of estimating the frequencies of multicomponent amplitude modulated (AM) signals as a hypothesis testing problem based on a time-dependent extension of the MUSIC algorithm. We subsequently propose a test statistic to detect periodic components in AM time-series. The power of the proposed algorithm is assessed in synthetic test signals and in real cell-cycle gene profiles extracted from microarray data.

关键词： Algorithm design and analysis Gene expression Multiple signal classification Testing Frequency estimation Circuits Diseases Cancer Drugs Amplitude estimation

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The Simulation of Smart Antennas in Network Simulator-2 Using MATLAB

The Simulation of Smart Antennas in Network Simulator-2 Usin...

引用

ieee International workshop on Computer-Aided Modeling, Analysis, and design of Communication Links and Networks, CAMAD

作者： Fulvio Babich Massimiliano Comisso Aljosa Dorni Flavio Barisi Alessandro Mania Dipartimento di Elettrotecnica Elettronica ed Informatica University of Trieste Trieste Italy

This paper presents an extension for network simulator-2 (ns-2) that enables the simulation of adaptive antennas in realistic channel conditions using MATLAB. A hybrid discrete-event discrete-time approach is adopted to exploit the scalability properties of ns-2 for the network simulation, and the modeling capabilities of MATLAB for the physical layer implementation. The developed package takes into account the physical antenna system and the signal processing algorithm to model in detail the antenna radiation pattern generated by each network node. The wireless propagation model of ns-2, which typically accounts only for path-loss attenuation and shadowing, is modified to include the Power Azimuth Spectrum (PAS), thus improving the accuracy of the estimated signal to Interference plus Noise Ratio (SINR) at each node in presence of multipath. The proposed numerical model, which can be interfaced with any wireless network supported by ns-2, is used to evaluate the throughput of a Wireless Mesh Network (WMN).

关键词： MATLAB Mathematical model Antennas and propagation Power system modeling signal to noise ratio Wireless mesh networks Adaptive arrays Scalability Physical layer Packaging

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Structured spatio-temporal sample covariance matrix enhancement with application to blind channel estimation in cyclic prefix systems

Structured spatio-temporal sample covariance matrix enhancem...

引用

ieee workshop on signal processing Advances in Wireless Communications (SPAWC)

作者： Samir-Mohamad Omar Dirk T.M. Slock Mobile Communications Department Eurecom Institute Sophia-Antipolis France

Multichannel aspect allows the introduction of blind channel estimation techniques. Most existing such techniques for frequency-selective channels are quite complex. In this paper, we consider the blind channel estimation problem for Single Input Multi Output (SIMO) cyclic prefix (CP) systems. We have shown before that blind channel estimation becomes computationally much more attractive and more straight forward to analyze in terms of performance in CP systems. Inspired by the iterative sample covariance matrix (SCM) structure enhancement techniques of Cadzow and others, we propose here an algorithm to structure the sample block circulant covariance matrix by enforcing two essential properties: rank and FIR structure. These two properties are exhibited by the true covariance matrix in the case of FIR SIMO channels with spatially white noise and CP transmission. The proposed enhancement procedure leads to an interesting enhanced SCM, even for the single CP symbol case. A simulation study for some classical channel estimators that depend on the SCM (with and without structuring) is presented, indicating that structuring allows for considerable performance gain in terms of the channel normalized mean square error (NMSE) over a wide SNR range.

关键词： Covariance matrix Blind equalizers Performance analysis Frequency domain analysis Mobile communication Finite impulse response filter White noise Convolution Algorithm design and analysis Iterative algorithms

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