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Arithmetically sub-optimal floating point digital filters an architectural power perspective

Arithmetically sub-optimal floating point digital filters an...

引用

Canadian conference on Electrical and Computer Engineering (CCECE)

作者： R.V.K. Pillai D. Al-Khalili A.J. Al-Khalili Concordia University Montreal Canada Royal Military College of Canada Kingston Canada

In CMOS floating point hardware design, transition activity scaling of functional units, taking into account the limitations of machine arithmetic, offers promising results as far as architectural power optimization of these units is concerned. This work targets characterization of the architectural power implications of floating point adder cores of DSP data paths during implementation of arithmetically sub-optimal floating point digital filters. Instrumented digital filter programs that emulate a DSP multiply-accumulate unit form the core of our experimental platform. With a class of arithmetically sub-optimal band pass/stop filters having a normalized centre frequency of 0.5, the transition activity scaled triple data path floating point adder scheme offers a power reduction of better than 75%.

关键词： Digital filters Hardware Digital signal processing Band pass filters Finite impulse response filter Design optimization Adders Frequency Floating-point arithmetic Energy consumption

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Low-complexity velocity estimation in high-speed optical Doppler tomography systems

Low-complexity velocity estimation in high-speed optical Dop...

引用

IEEE International conference on Image processing

作者： M. Milosevic W. Schwartzkopf T.E. Milner B.L. Evans A.C. Bovik Department of Electrical and Computer Engineering University of Texas Austin Austin TX USA

Optical Doppler Tomography (ODT) is a noninvasive 3-D optical interferometric imaging technique that measures static and dynamic structures in a sample. To obtain the dynamic structure, e.g. blood flowing in tissue, a velocity estimation algorithm detects the Doppler shift in the received interference fringe data with respect to the carrier frequency. Previous velocity estimation algorithms use conventional Fourier magnitude techniques that do not provide sufficient frequency resolution in fast ODT systems because of the high data acquisition rates and hence short time series. In this paper, we propose a nonlinear algorithm that uses the phase shift between two successive scans of interference fringe data to give a high-resolution estimate of the Doppler shift. The algorithm detects Doppler shifts of 0.1 to 3 kHz with respect to a 1 MHz carrier. In processing 5 frames/s with 100×100 pixels/frame and 32 samples/pixel, i.e. 1.6 million samples/s, the algorithm requires 26 million multiply-accumulates/s. The algorithm works well at 4 bits/sample. The low complexity and small input data size are well-suited for real-time implementation in software. We provide a mathematical analysis of the Doppler shift resolution by modeling the interference fringe data as an AM-FM signal.

关键词： High speed optical techniques Optical interferometry Frequency estimation Doppler shift Nonlinear optics Interference Tomography Optical imaging Blood data acquisition

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Fast speaker adaptation using a priori knowledge

Fast speaker adaptation using a priori knowledge

引用

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

作者： R. Kuhn P. Nguyen J.-C. Junqua R. Boman N. Niedzielski S. Fincke K. Field M. Contolini Speech Technology Laboratory Panasonic Technologies Inc. Santa Barbara CA USA Institut Eurécom France

Previously, we presented a radically new class of fast adaptation techniques for speech recognition, based on prior knowledge of speaker variation. To obtain this prior knowledge, one applies a dimensionality reduction technique to T vectors of dimension D derived from T speaker-dependent (SD) models. This offline step yields T basis vectors, the eigenvoices. We constrain the model for new speaker S to be located in the space spanned by the first K eigenvoices. Speaker adaptation involves estimating K eigenvoice coefficients for the new speaker; typically, K is very small compared to original dimension D. Here, we review how to find the eigenvoices, give a maximum-likelihood estimator for the new speaker's eigenvoice coefficients, and summarize mean adaptation experiments carried out on the Isolet database. We present new results which assess the impact on performance of changes in training of the SD models. Finally, we interpret the first few eigenvoices obtained.

关键词： Hidden Markov models Principal component analysis Loudspeakers Speech recognition Maximum likelihood estimation Independent component analysis Training data Laboratories databases Surfaces

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Power implications of additions in floating point DSP-an architectural perspective

Power implications of additions in floating point DSP-an arc...

引用

AFRICON conference

作者： R.V.K. Pillai D. Al-Khalili A.J. Al-Khalili Concordia University Montreal Canada Royál Military College Kingston Canada

This work targets experimental characterization of the architectural power implications of floating point adders that form part of DSP cores. High level models that characterize the power behavior of floating point adders are developed. Instrumented digital filter programs that emulate a DSP multiply-accumulate unit form the core of our experimental platform. The experiments substantiate the validity of our transition activity scaling based approach for the design of low power floating point adders. The worst case power reduction offered by the proposed transition activity scaled triple data path floating point adder (TDPFADD) is consistently above 50%. The corresponding reduction in power delay product is better than 70%.

关键词： Digital signal processing Adders Statistics Energy consumption Educational institutions Instruments Digital filters Delay Hardware Very large scale integration

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Reusable binary-paired partitioned neural networks for text-independent speaker identification

Reusable binary-paired partitioned neural networks for text-...

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

作者： S.A. Zahorian Department of Electrical and Computer Engineering Old Dominion University Norfolk VA USA

A neural network algorithm for speaker identification with large groups of speakers is described. This technique is derived from a technique in which an N-way speaker identification task is partitioned into N*(N-1)/2 two-way classification tasks. Each two-way classification task is performed using a small neural network which is a two-way, or pair-wise, network. The decisions of these two-way networks are then combined to make the N-way speaker identification decision (Rudasi and Zahorian, 1991 and 1992). Although very accurate, this method has the drawback of requiring a very large number of pair-wise networks. In the new approach, two-way neural network classifiers, each of which is trained only to separate two speakers, are also used to separate other pairs of speakers. This method is able to greatly reduce the number of pair-wise classifiers required for making an N-way classification decision, especially when the number of speakers is very large. For 100 speakers extracted from the TIMIT database, the number of pair-wise classifiers can be reduced by approximately a factor of 5, with only minor degradation in performance when 3 seconds or more of speech is used for identification. Using all 630 speakers from the TIMIT database, this method can be used to obtain over 99.7% accuracy. With the telephone version of the same database, an accuracy of 40.2% can be obtained.

关键词： Neural networks databases Speech Training data Partitioning algorithms Degradation Telephony Vector quantization Pattern recognition

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The use of iterative channel estimation (ICE) to improve link margin in wideband CDMA systems

The use of iterative channel estimation (ICE) to improve lin...

引用

IEEE conference on Vehicular Technology (VTC)

作者： T.M. Schmidl A.G. Dabak S. Hosur DSPS R&D Center Texas Instruments Inc. Dallas TX USA

Wideband CDMA (WCDMA) systems employ coherent detection by periodically inserting pilot symbols in the data stream. Because of the small number of the pilot symbols available, the channel estimates are not perfect, and there is a loss in link margin. Iterative channel estimation (ICE) is a novel method that uses both pilot and data symbols for channel estimation. Link margin gains of 0.3 to 1.3 dB are obtained for data rates of 32 to 64 KSPS, with larger gains at higher Doppler frequencies. The extra complexity to implement ICE is very modest at 2.56 and 5.12 MIPS for 32 and 64 KSPS channels, respectively, which is easily accommodated in today's DSP technology and represents a small fraction of the total power consumed by the mobile handset. Reducing the required E/sub b//N/sub o/ by even 0.4 dB will allow the system to support 10% more voice users. This paper gives the theoretical analysis of ICE, and Monte Carlo simulations are used to compare the performance of ICE against channel estimation methods which only use pilot symbols.

关键词： Channel estimation Ice Wideband Multiaccess communication Digital signal processing Iterative methods Frequency Standardization Downlink Research and development

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A comparison of part of speech taggers in the task of changing to a new domain

A comparison of part of speech taggers in the task of changi...

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International conference on Information Intelligence and Systems

作者： L. Boggess J.S. Hamaker R. Duncan L. Klimek Yufeng Wu Yu Zeng KUDZU Laboratory for Applied Intelligent Systems Research Department of Computer Science Mississippi State University USA ISIP (Institute for Signal and Information Processing) College of Engineering Mississippi State University USA Laboratory for Applied Intelligent Systems Research Department of Computer Science Mississippi State University USA

Part-of-speech tagging in real-world applications is performed on text in domains which are different from the publicly available large training data sets. The two most successful part-of-speech taggers are trained on the Wall Street Journal corpus, a corpus of millions of words. We compare their performance on a test set from a different domain-astronomy-from documents that are available on the World Wide Web. The Maximum Entropy Part of Speech Tagger (MXPOST) and the Transformation-Based Learning Tagger are well-known and widely used in language research and development systems. The two taggers were tested in several modes: (1) after training on the Wall Street Journal corpus only, (2) after training on only a small body of text from our astronomy domain, (3) with and without an auxiliary lexicon derived from many astronomy-related Web documents, and (4) after incremental training-that is, having been trained on the Wall Street Journal, with additional training from the specific domain. One conclusion from the experiment is that different taggers exhibit different biases when trained on the same data.

关键词： Speech Tagging Laboratories Testing Natural languages Electrical capacitance tomography Entropy Natural language processing data mining Research and development

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Linear receivers for spread spectrum communication over multipath dispersive channels

Linear receivers for spread spectrum communication over mult...

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IEEE conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： M.V. Clark AT and T Research Laboratories Red Bank NJ USA

This paper studies single-signature and orthogonal multiple-signature (multicode) spread spectrum communication over frequency-selective Rayleigh fading channels. We analyze the influence of the spreading factor, which is defined here as the ratio of transmission bandwidth to user symbol rate. More specifically, we focus on the impact of small spreading factors and highly-dispersive channels which, together, introduce potentially detrimental intersymbol or inter-signature interference in the detection process. Such links are important for high-speed data transmission in future wideband code-division multiple access (WCDMA) systems. We present theoretical results for the output signal-to-distortion ratio of a range of important linear receiver types: matched filter (MF), minimum mean-square error (MMSE), and pre-equalizing (PE). We show that, at any specified spreading factor, single-signature transmission, orthogonal multiple-signature transmission, and simple frequency diversity all yield about the same link performance for these receivers. Moreover, we demonstrate the significant gain of MMSE processing over MF processing (equivalent to RAKE processing) in links with small spreading factors.

关键词： Spread spectrum communication Multiaccess communication Fading Bandwidth Intersymbol interference data communication Wideband Matched filters Frequency diversity RAKE receivers

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Source number estimators using Gerschgorin radii

Source number estimators using Gerschgorin radii

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IEEE Region 10 International conference TENCON

作者： Hsien-Tsai Wu Department of Electronic Engineering Southern Taiwan University of Technology Tainan County Taiwan

We introduce the effective uses of the Gerschgorin radii of the unitary transformed covariance matrix for source number detection. The proposed log-likelihood function for developing the detection criteria combines the Gerschgorin radii to the AIC and minimum description length (MDL) and improves their detection performances for Gaussian and white noise processes. It is verified that the Gerschgorin AIC (GAIC) criterion yields a consistent estimate of source number and the Gerschgorin MDL (GMDL) criterion does not tend to under-estimate the source number at small or moderate data samples. Furthermore, the detection performances of the criteria can be further improved through the suggested rotation and averaging.

关键词： Covariance matrix Radar detection signal resolution Direction of arrival estimation Cities and towns White noise Yield estimation Underwater tracking Radar signal processing Multiple signal classification

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Adaptive beamforming in CDPD mobile end systems

Adaptive beamforming in CDPD mobile end systems

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IEEE International conference on Electronics, Circuits and Systems (ICECS)

作者： A.K. Salkintzis P.T. Mathiopoulos Department of Electrical and Computer Engineering University of British Columbia Vancouver BC Canada

This paper addresses the application of antenna diversity in the Cellular Digital Packet data (CDPD) Mobile End Systems (M-ESs) as a means to improve the performance of the CDPD forward channel and meet the tight requirements for mobile Internet access. The focal point is the construction of a maximal-ratio digital beamformer, which effectively exploits a periodic signal, transmitted in every CDPD cell. This signal serves as a reference temporal signal and it is used in the adaptive calculation of the complex weights of the beamformer. The paper presents the design and the operation of the adaptive beamformer, it demonstrates its performance through simulation results and additionally it discusses its significance and its applications. It is argued that even for relatively small spacing between the antenna elements the considered beamforming scheme has the potential to provide significant improvements in the performance of the CDPD forward channel.

关键词： Array signal processing Mobile computing Internet Cellular networks Land mobile radio cellular systems Automatic repeat request Radio frequency Fading Base stations Application software

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