SCIENTIFIC AND TECHNICAL AEROSPACE REPORTS
A Biweekly Publication of the National Aeronautics and Space Administration
VOLUME 44, ISSUE 2 - January 27, 2006
32 COMMUNICATIONS AND RADAR
Includes radar; radio, wire, and optical communications; land and global communications; communications theory.
For related information see also 04 Aircraft Communications and Navigation; and 17 Space Communications, Spacecraft Communications, Command and Tracking; for search and rescue, see 03 Air Transportation and Safety; and 16 Space Transportation and Safety.
20060002826 Lawrence Livermore National Lab., Livermore, CA USA
Scalable Analysis Techniques for Microprocessor Performance Counter Metrics
Ahn, D. H.; Vetter, J. S.; Jul. 24, 2002; 18 pp.; In English Report No.(s): DE2005-15013340; UCRL-JC-148058; No Copyright; Avail.: Department of Energy Information Bridge
Contemporary microprocessors provide a rich set of integrated performance counters that allow application developers and system architects alike the opportunity to gather important information about workload behaviors. These counters can capture instruction, memory, and operating system behaviors. Current techniques for analyzing data produced from these counters use raw counts, ratios, and visualization techniques to help users make decisions about their application source code. While these techniques are appropriate for analyzing data from one process, they do not scale easily to new levels demanded by contemporary computing systems. Indeed, the amount of data generated by these experiments is on the order of tens of thousands of data points. Furthermore, if users execute multiple experiments, then we add yet another dimension to this already knotty picture. This flood of multidimensional data can swamp efforts to harvest important ideas from these valuable counters. Very simply, this paper addresses these concerns by evaluating several multivariate statistical techniques on these datasets. We find that several techniques, such as statistical clustering, can automatically extract important features from this data. These derived results can, in turn, be feed directly back to an application developer, or used as input to a more comprehensive performance analysis environment, such as a visualization or an expert system. NTIS
Computer Networks; Counters; Microprocessors; Statistical Analysis; Memory (Computers)
20060002930 Bureau of the Census, Washington, DC, USA
Economic Census 2002: Manufacturing, Industry Series. Switchgear and Switchboard Apparatus Manufacturing
Dec. 2004; 50 pp.; In English Report No.(s): PB2006-103382; EC02-31I-335313(RV); No Copyright; Avail.: CASI: A03, Hardcopy
The economic census is the major source of facts about the structure and functioning of the nation's economy. It provides essential information for government, business, industry, and the general public. Title 13 of the USA Code (Sections 131, 191, and 224) directs the Census Bureau to take the economic census every 5 years, covering years ending in '2' and '7.' The economic census furnishes an important part of the framework for such composite measures as the gross domestic product estimates, input/output measures, production and price indexes, and other statistical series that measure short-term changes in economic conditions. The Industry Series, containing 473 reports, covers a single NAICS industry (six-digit code). These reports include such statistics as number of establishments, employment, payroll, value added by manufacture, cost of materials consumed, value of shipments, capital expenditures, etc. The industry reports also include data for states with 100 employees or more in the industry. The data in industry reports are preliminary and subject to change. This U.S. industry comprises establishments primarily engaged in manufacturing switchgear and switchboard apparatus. NTIS
Census; Economic Analysis; Economics; Industries; Manufacturing
20060002937 Bureau of the Census, Washington, DC, USA
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Economic Census 2002: Manufacturing, Industry Series. Other Communication and Energy Wire Manufacturing
Dec. 2004; 50 pp.; In English Report No.(s): PB2006-103387; EC02-31I-335929(RV); No Copyright; Avail.: CASI: A03, Hardcopy
The economic census is the major source of facts about the structure and functioning of the nation's economy. It provides essential information for government, business, industry, and the general public. Title 13 of the USA Code (Sections 131, 191, and 224) directs the Census Bureau to take the economic census every 5 years, covering years ending in '2' and '7.' The economic census furnishes an important part of the framework for such composite measures as the gross domestic product estimates, input/output measures, production and price indexes, and other statistical series that measure short-term changes in economic conditions. The Industry Series, containing 473 reports, covers a single NAICS industry (six-digit code). These reports include such statistics as number of establishments, employment, payroll, value added by manufacture, cost of materials consumed, value of shipments, capital expenditures, etc. The industry reports also include data for states with 100 employees or more in the industry. The data in industry reports are preliminary and subject to change. This U.S. industry comprises establishments primarily engaged in manufacturing insulated wire and cable of nonferrous metals from purchased wire. NTIS
Census; Communication Equipment; Economic Analysis; Economics; Industries; Manufacturing; Wire
20060002940 Swedish Defence Research Establishment, Linkoeping, Sweden
UIntelligibility of Simultaneous Radiocommunication
Carlander, O.; Kindstroem, M.; Dec. 2004; 28 pp.; In Swedish Report No.(s): PB2006-100613; FOI-R-1525-SE; No Copyright; Avail.: CASI: A03, Hardcopy
A command operator of fire and rescue units needs to pay attention to several simultaneous streams of radio voices to optimize information gathering for the coordination of simultaneous emergency missions. An experiment investigated command operator ability to discern stereo and 3D-audio call-signs presented in background noise of added voice sources. Each of 10 command operators listened to 1 to 4 call-signs combined with 2 to 4 background voices with the primary task to identify to speaker of each call-sign. A secondary visual and manual-response task was used to induce an overall heightened mental workload situation. 3D-audio presentation resulted in a slightly increased number of correctly indicated locations of call-signs. Four background voices reduced correctness compared to sets of 1 and 2 call-signs, respectively. The results are discussed in relation to the potential for improving 3D-audio presentation and intelligibility, and its possible impact on operator effectiveness. NTIS
Intelligibility; Telecommunication
20060003056 Hokkaido Univ., Sapporo, Japan
A Lattice Filter Model with Accurate Lip Impedance for Dynamic Articuratory Movement
Miki, Nobuhiro; Motoki, Kunitoshi; Nagai, Nobuo; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 22A4.1 - 22A4.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
In the vocal tract model, the area function of vocal tracts is accurately represented by a polynomial function with a few parameters, and the digital filter realization includes the loss, the accurate lip impedance and the variation of the tract length. We measured the lip impedance using a manikin and pipes set in a baffle, and obtained the reflection coefficient at the lips. From the results of these experiments, we discuss the difference between the measured characteristics and those of a sypherical baffle model, and show a method to obtain approximate digital filters for the reflection characteristics at the end of the vocal-tract; i.e. the lips. By introducing this filter, it becomes possible to express arbitrary changes of the length of the vocal-tract without changing the sampling frequency. Author
Digital Filters; Impedance; Baffles
20060003059 Bell Telephone Labs., Inc., Holmdel, NJ, USA
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An Efficient Block Matching Algorithm for Motion-Compensated Coding
Puri, A.; Hang, H.-M.; Schilling, D. L.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 25.4.1 - 25.4.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
We present an efficient search technique which minimizes the computations necessary for estimating the motion in video-sequences by the block matching method.
We also discuss the theoretical basis for conducting such a reduced search by our technique.
We then present two algorithms which employ the proposed technique for estimating the motion typical of videoconferencing environment.
Next, the results of computer simulations on a real video-sequence are included which demonstrate the effectiveness of the proposed technique.
Finally, the results of a study of statistical properties of block motion-compensated frame difference signals are also summarized, to assist in future choice of a coding strategy for such signals. Author
Algorithms; Computerized Simulation; Video Communication
20060003084 Queensland Inst. of Tech., Brisbane, Australia
Implementation of State-Space Digital Filter Structures Using Block Floating Point Arithmetic
Sridharan, S.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 908-911; In English; See also 20060003045; Copyright; Avail.: Other Sources
Block floating-point arithmetic is considered as an alternative to fixed-point and floating-point arithmetic in the implementation of recursive digital filters. Block floating-point implementation of state-space digital structures is shown to have improved signal-to-noise ratio compared to fixed-point implementation and can be designed to be free of over flow. It is shown that the filter cannot support zero input limit cycle oscillations of period higher than one. An architecture suitable for the VLSI implementation of a block floating point co-processor is described. Author
Floating Point Arithmetic; IIR Filters; Spacecraft Structures
20060003090 BBN Systems and Technologies Corp., Cambridge, MA, USA
A Single Board Multirate APC Speech Coding Terminal
Field, K.; Derr, A.; Cosell, L.; Henry, C.; Krasner, M.; Tiao, J.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 960-963; In English; See also 20060003045; Copyright; Avail.: Other Sources
A single-board 16 kb/s and 32 kb/s APC speech coding terminal is described. The board is configurable as either two receivers, two transmitters, or a single-channel full-duplex coder. The terminal consists of two identical, independent TI TMS32020- based modules. Included in each module is a TMS32020, an analog I/O section, ND/A conversion, digital I/O circuitry, static RAMs, and EPROMs. The board size is 5.6 in. x 6.0 in., and can be packaged in a standard secretarial desk phone shell. A software architecture using this hardware to implement the 16/32 kb/s Adaptive Predictive Coding with Hybrid Quantization (APCHQ) algorithm is described. Author
Voice Data Processing; Speech Recognition; Algorithms
20060003100 Bell Telephone Labs., Inc., Morristown, NJ, USA
Statistical Features versus Word Templates for Speaker Independent Digit Recognition over Long Distance Telephone Connections
Bocchieri, Enrico L.; Doddington, George R.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 27.7.1 - 27.1.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
We extend to speaker independent recognition of isolated digits over long distance telephone connections (from the FAA telephone speech data base) the use of frame-specific statistical features that have already been tested on a studio-quality data base (Texas Instruments isolated digit data base). In addition, for both the telephone and the studio-quality data bases, we compare recognition performance of frame-specific statistical models versus pattern matching of the input speech with multiple templates of the vocabulary digits. When testing is performed on the more difficult long distance telephone data base, the performance of frame-specific distance measures (3.6% substitution rate) is superior to pattern matching with multiple templates (6.8% substitution rate). Author
Digits; Templates; Words (Language); Data Bases
20060003103 California State Univ., Long Beach, CA, USA
Adaptive Noise Cancellation for Speech with a TMS32020
Hen Geul, Yeh; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 1171-1174; In English; See also 20060003045; Copyright; Avail.: Other Sources
The noise cancellation using an adaptive filtering technique for speech signal is experimentally studied with a TMS32020 digital signal processor. The LMS algorithm is employed in the implementation of an adaptive filter due to its simplicity and ease of computation. The TMS32020 digital signal processor is suited for the implementation since it has the capability and features to implement all of the required function with full precision. The transversal filter structure is selected due to its simplicity and ease of coding. Author
Cancellation; Digital Systems; Signal Analyzers; Signal Processing
20060003107 North Carolina State Univ., Raleigh, NC, USA
Consistency of the Minimum Mean Square Error Estimate
Trussell, H. Joel; Civanlar, M. Reha; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 1213-1216; In English; See also 20060003045; Copyright; Avail.: Other Sources
The minimum mean square error estimate for the deconvolution problem of a Gaussian signal in Gaussian noise is shown to be feasible in the sense of being inside closed convex sets defined by the noise statistics. It is pointed out that there is some a priori knowledge which is not satisfied by the Wiener solution but the set formed by this information is not convex. Author
Random Noise; Mean Square Values; Error Analysis; Consistency
20060003108 State Univ. of New York, Buffalo, NY, USA
Reconstruction of Bandlimited Signals from Their Unevenly-Spaced Sampled Data
Soumekh, M.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 28.12.1 - 28.12.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
This paper addresses the problem of reconstructing a bandlimited signal from a finite number of its unevenly spaced sampled data. We present the main results for a Fourier analysis of the available data [10] done in a framework similar to Shannon's sampling theorem for evenly spaced data. The results are then utilized to determine the required constraints and algorithms for accurate reconstruction. Author
Signal Processing; Sampled Data Systems; Reconstruction
20060003110 Illinois Univ., Urbana, IL, USA
Extrapolation of Multi-Dimensional Bandlimited Sequences Using Energy Concentration Information
Potter, L. C.; Arun, K. S.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 27.17.1 - 28.17.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
This paper presents a new algorithm for the extrapolation of multidimensional bandlimited sequences. The new algorithm is capable of incorporating prior knowledge of the concentration of signal energy on a finite region and imposes no restriction on either the location of observations from the multidimensional signal to be extrapolated or the shape of its passband. Author
Extrapolation; Tomography; Sequencing
20060003111 Arkansas Univ., Fayetteville, AR, USA
A Minimum Risk Quantizer for Noisy Sources
Cook, Mark K.; Jones, Richard A.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 28.20.1 - 28.20.3; In English; See also 20060003045 Contract(s)/Grant(s): AFOSR-84-0322; Copyright; Avail.: Other Sources
The problem of quantizing noisy signals in an optimal manner is addressed. Quantizer designs relying on training sets, 'clean' probability statistics, or composite statistics of source and noise, will yield designs which are sub-optimal and possibly detrimental to system performance. The concept of the quantizer as an estimator is used in conjunction with a risk function to produce a minimum risk quantizer for noisy sources. In particular, the minimum-risk quantizer design theory for the case of independent, identically distributed source with additive (i.i.d.) noise is developed. The minimum-risk quantizer criteria for the specific problem of a source signal with gaussian statistics corrupted by additive gaussian noise and the squared-error cost function is produced. It is further shown that the Max-Lloyd optimal quantizer criteria is a subset of the minimum-risk criteria for noiseless conditions. Author
Probability Theory; Counters; Signal Processing
20060003169 Bell Telephone Labs., Inc., Murray Hill, NJ, USA
A Family of ADPCM Coders Implemented on Real-Time Hardware
Cox, Richard V.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 22B2.1- 22B2.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
Recently, work in ADPCM speech coding has focused on using the adaptive predictors,already included in the coder algorithm, as the basis for noise shaping and post-filtering. This technique is based on improving the perceived quality of the signal -- no actual improvement in signal-to-noise ratio is possible. The original work was performed with an intended application of improving the performance of 16 and 24 kbps ADPCM. We have made extensions of this work for lower bit rates. Specifically we have looked at reducing the sampling rate by use of digital interpolation and at extending the technique to quantization rates as low as one bit per sample. The algorithms we have implemented run at rates in the range of 6 to 16 kbps. The coders below 9 kbps are probably not useful because their quality is too poor, but they do demonstrate the power of the noise shaping and post-filtering concept. The 9, 12 and 16 kbps coders would be useful in applications requiring a low complexity coder with only a single encoding. This paper describes the algorithms,theirimplementation using the WE (R) DSP32 signal processor,and their performance. Author
Algorithms; Voice Data Processing; Signal to Noise Ratios; Real Time Operation
20060003170 NEC Corp., Chiba, Japan, Japan
Implementation of a Multi-Pulse Speech Codec with Pitch Prediction on a Single Chip Floating-Point Signal Processor
Fukui, A.; Shibagaki, K.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 22B3.1 - 22B3.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
The multi-pulse speech coding with pitch prediction has been known as an efficient speech coding technique for coding speech at a bit rate of 8 to 9.6 kbps. To implement this coding method on a signal processor for real time applications, problems exist concerning the amount of computations, speech quality, and the data RAM size on a signal processor. As countermeasures against these problems, we propose: l) Correlation computation amount reduction method by utilizing pitch periodicity of the impulse response, 2) Pulse search method to modify pulse amplitude by only a small amount of computations in order to improve speech quality, 3) Efficient use of the memory space of a signal processor. These methods made it possible to implement a 8 to 9.6 kbps multi-pulse speech codec with pitch prediction with an optimum analysis frame length of 20 ms on a single chip 32-bit floating point signal processor (#PD77230). Author
Signal Processing; Voice Data Processing; Signal Analyzers; Coding
20060003187 Pennsylvania State Univ., PA, USA
A Transform Based Covariance Differencing Approach Bearing Estimation
Prasad, S.; Williams, R. T.; Mahalanabis, A. K.; Sibul, L., II; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 26.10.1 - 26.10.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
In recent years a new, and very powerful technique for parameter estimation--the eigenstructure, or signal subspace method--has been developed. Eigenstructure algorithms are closely related to Pisarenko's method for estimating the frequencies of sinusoids in white Gaussian noise. In theory they yield asymptotically unbiased estimates of arbitrarily close parameters, independent of the signal-to-noise ratio. Although signal subspace methods have proven to be powerful tools, they are not without drawbacks. An important weakness of all signal subspace algorithms is their need to know the noise covariance explicitly. The important problem of developing signal subspace based procedures for signals in noise fields with unknown covariance has not been satisfactorily addressed. It is our intent to propose a solution to the problem of direction-of-arrival estimation for a broad class of unknown noise fields. We will then briefly discuss other important estimation problems for which modified versions of this procedure can be applied. Author
Algorithms; Parameter Identification; Random Noise; Signal to Noise Ratios; White Noise
20060003190 Schlumberger Palo Alto Research, CA, USA
Experiments in Isolated Digit Recognition with a Cochlear Model
Loeb, Eric P.; Lyon, Richard F.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 27.3.1 - 27.3.4; In English; See also 20060003045 Contract(s)/Grant(s): N00039-84-C-0585; Copyright; Avail.: Other Sources
We have conducted speaker-independent isolated digit recognition experiments using vector quantized cochleagrams. Without the use of time order information, we were able to achieve a recognition rate of 97.24%. With a modified Viterbi algorithm we achieved a rate of 98.38%. Since we achieved a 98.05% recognition rate with a scheme that did static pattern matching on the first and second time-halves of our utterances, we must call into question the effectiveness with which the Viterbi algorithm uses time order information. Our results also lead us to conclude that future progress may depend on our ability to construct more sophisticated vector quantizers. Author
Cochlea; Pattern Recognition; Digits
20060003191 Hong Kong Univ., Hong Kong
Speaker-Independent IsolatedWord Recognition UsingWord-Based Vector Quantization and Hidden Markov Models
Cheung, Y. S.; Leung, S. T.; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 1135-1138; In English; See also 20060003045; Copyright; Avail.: Other Sources
In this paper, we investigate the possibility of using word-based vector quantization with hidden Markov models for speaker-independent isolated word recognition. Two word-based algorithms were proposed and studied. Experiments were carried out on Chinese (Cantonese) digits spoken by 110 speakers (55 males and 55 females) in two databases. An improves of about 3% in recognition rate was obtained in one of the word-based algorithms. The results and implications are discussed. Author
Speech; Vector Quantization; Words (Language); Digits
20060003199 Tsinghua Univ., Bejing, China
A Large-Vocabulary Chinese Speech Recognition System
Huang, Xue-Dong; Cai, Lian-Hong; Fang, Di-Tang; Ci, Bian-Jin; Zhou, Li; Jian, Li; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 27.12.1 - 27.12.4; In English; See also 20060003045; Copyright; Avail.: Other Sources
In this paper, we introduce a speaker dependent isolated word recognition system which is dedicated for Chinese character input. The method presented here offers an effective solution to the large-vocabulary isolated word recognition. The classification is performed hierarchically. In the preselection stage, the candidates are selected by KNN rule using the utterance length and coarse templates.Aranked list of candidates is obtained by the following final classifier which performs a nonlinear time alignment uncompressed templates using DTW algorithm. The recognition rate is more than 90% when monosyllables take up 1/3 of the vocabulary (1000). Author
China; Speech Recognition; Templates
20060003200 Matsushita Research Inst. Tokyo, Inc., Kawasaki, Japan
A Telephone Speech Recognition System Using Word-Spotting Technique Based on Statistical Measure
Kimura, Tatsuya; Niyada, Katsuyuki; Hiraoka, Shoji; Morii, Shuji; Watanabe, Taisuke; IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP '87); Volume 2; 1987, pp. 1175-1178; In English; See also 20060003045; Copyright; Avail.: Other Sources
A telephone speech recognition system for an isolated word which has sufficient performance in practical use is reported here. Word spotting technique is applied to the system so as to keep it relatively immune from noise. Word spotting is performed by a new time normalization algorithm based on linear time distortion pattern matching method named COLM (Continuous Linear Matching). The method includes only simple operations such as look up tables and additions. The entire system is implemented in a small board including a single digital signal processor. An experiment proved 96.4% average recognition rate. The experiment was carried out using 10 Japanese numerals pronounced by 240 males and females through telephone lines. Author
Digital Systems; Pattern Recognition; Speech Recognition; Words (Language)
20060003680 Helsinki Univ. of Technology, Espoo, Finland
Simulation Of A Digital Signal Processing Architecture Based On The Data Flow Principle
1982 International Symposium on Circuits And Systems, Volume 3; [1982], pp. 1053-1056; In English; See also 20060003631; Copyright; Avail.: Other Sources
The data flow principle has been proposed to be used for digital signal processing applications. To be able to analyze the properties and performance of this kind of an architecture computer simulations have been used. This paper discusses the simulation model used and gives some practical results based on the simulation experiments. The simulations show that the data flow principle can be efficiently used for digital signal processing in environments demanding high througput and flexibility. Author
Computerized Simulation; Signal Processing; Data Flow Analysis
20060003719 Centro Studi e Laboratori Telecomunicazioni, Turin, Italy
Multiplication-Free Filters For Subband Coding Of Speech
Pirani, Giancarlo; Rusina, Fulvio; Zingarelli, Valerio; 1982 International Symposium on Circuits And Systems, Volume 3; [1982], pp. 848-851; In English; See also 20060003631; Copyright; Avail.: Other Sources
Recently, the subband coding scheme for transmission of speech has been shown to be a promising technique for achieving high quality at medium bit rates (in the range between 16 and 32 kbit/s). The complexity of this coder depends partly on the banks of filters which are used in the transmitter to split the speech signal into several bands and, in the receiver, to reconstruct the signal. In this paper a way of reducing this complexity is investigated by analyzing some methods of designing multiplier-free filters, the coefficients of these filters are powers of two, thereby substituting each multiplication with a shift, it it is considered that each filter of the banks may have up to 32 coefficients and that the two banks may include 3D filters the number of avoided multiplications is quite challenging. A key point of this design is to find directly the power-of-two coefficients through a nonlinear optimization technique. The performance degradations of the overall system is also investigated when the power-of-two coefficients are used instead of the infinite-precision ones. Author
Coding; Multipliers; Voice Data Processing
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